--- /dev/null
+2012-10-08 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.18.0-rc1 Released.
+
+2012-10-05 20:20 +0000 [r374570-374581] dlee <dlee@localhost>:
+
+ * main/manager.c: I've committed too much. Reverting part of
+ r374570.
+
+ * main/manager.c: Improve AMI long line error handling In AMI's
+ parser, when it receives a long line (> 1024 characters), it
+ discards that line, but continues to process the message
+ normally. Typically, this is not a problem because a) who has
+ lines that long and b) usually a discarded line results in an
+ invalid message. But if that line is specifying an optional
+ field, then the message will be processed, you get a 'Response:
+ Success', but things don't work the way you expected them to.
+ This patch changes the behavior when a line-too-long parse error
+ occurs. * Changes the log message to avoid way-too-long (and
+ truncated anyways) log messages * Adds a 'parsing' status flag to
+ Response: Success * Sets parsing = MESSAGE_LINE_TOO_LONG if,
+ well, a line is too long * Responds with an appropriate error if
+ parsing != MESSAGE_OKAY (closes issue AST-961) Reported by: John
+ Bigelow Review: https://reviewboard.asterisk.org/r/2142/
+
+2012-10-05 18:20 +0000 [r374536] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c,
+ channels/misdn/isdn_lib.h, channels/chan_misdn.c: Merged
+ revisions 374515-374535 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500
+ (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode *
+ Made setup_bc() static. Patches: patch1_unused-code.diff (license
+ #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882
+ ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500
+ (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan
+ states Patches: patch2_unused-states.diff (license #6372) patch
+ uploaded by Guenther Kelleter JIRA ABE-2882 ................
+ r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012)
+ | 16 lines chan_misdn: Remove unnecessary null pointer checks and
+ checks for stack->nt * cleanup_bc() is always called with valid
+ bc (or it would've crashed before). * Value of stack->nt is known
+ in advance at some places. * Rename handle_event() to
+ handle_event_te(), handle_frm() to handle_frm_te(). Patches:
+ patch3_checks.diff (license #6372) patch uploaded by Guenther
+ Kelleter Modified JIRA ABE-2882 ................ r374518 |
+ rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines
+ chan_misdn: Fix spelling in log messages Patches:
+ patch4_spelling.diff (license #6372) patch uploaded by Guenther
+ Kelleter JIRA ABE-2882 ................ r374519 | rmudgett |
+ 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines
+ chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after
+ calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is
+ emptied, cleaned and set not in use, although
+ misdn_lib_send_event() already did the same. This is bad. When
+ it's not in use we are not allowed to touch it. * Moved log
+ message in front of the resulting actions and fixed it to match
+ the case. Patches: patch5_bccleanup.diff (license #6372) patch
+ uploaded by Guenther Kelleter JIRA ABE-2882 ................
+ r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012)
+ | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up
+ etc., really bad stuff. * Fix return codes of cb_events() for
+ EVENT_SETUP to use caller's cleanup mechanisms. * Move
+ cl_queue_chan() call after bearer check. Patches:
+ patch6_leaks.diff (license #6372) patch uploaded by Guenther
+ Kelleter JIRA ABE-2882 ................ r374521 | rmudgett |
+ 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines
+ chan_misdn: We must initialize cause on sending a DISCONNECT. We
+ must initialize cause on sending a DISCONNECT, so it is later
+ correctly indicated to ast_channel in case the answer
+ (RELEASE/RELEASE_COMPLETE) does not include one. Patches:
+ patch7_hangupcause.diff (license #6372) patch uploaded by
+ Guenther Kelleter JIRA ABE-2882 ................ r374522 |
+ rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines
+ chan_misdn: Remove unused code for upqueue Patches:
+ patch8_unused-upqueue.diff (license #6372) patch uploaded by
+ Guenther Kelleter JIRA ABE-2882 ................ r374523 |
+ rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines
+ chan_misdn: Improve debugging (port number, messages fixed, dups
+ removed) Patches: patch9_debug.diff (license #6372) patch
+ uploaded by Guenther Kelleter JIRA ABE-2882 ................
+ r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012)
+ | 8 lines chan_misdn: Better debug: we can print_bc_info even if
+ there's no ast leg. Patches: patch10_debug-bc-2.diff (license
+ #6372) patch uploaded by Guenther Kelleter Modified. JIRA
+ ABE-2882 ................ r374534 | rmudgett | 2012-10-05
+ 12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn:
+ setup_bc() is called too early for an incoming SETUP on TE. This
+ prevents the B channel from being setup for HDLC mode when
+ requested by the bearer capability and config option hdlc=yes. It
+ violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not
+ connect to the channel until a CONNECT ACKNOWLEDGE message has
+ been received." * Call setup_bc() on receipt of
+ CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for
+ PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by
+ Guenther Kelleter Modified. JIRA ABE-2881 ................
+ r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012)
+ | 2 lines chan_misdn: Remove some more deadcode. ................
+
+2012-10-04 20:15 +0000 [r374475-374479] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * CHANGES, main/dsp.c, configs/dsp.conf.sample: dsp.c User
+ Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END Instead of
+ a recompile, allow values to be adjusted in dsp.conf For binary
+ distributions allows easy adjustment for wobbly GSM calls, and
+ other reasons. Defaults to DTMF_HITS_TO_BEGIN=2 and
+ DTMF_MISSES_TO_END=3 (closes issue ASTERISK-17493) Tested by:
+ alecdavis alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2144/
+
+ * main/dsp.c: dsp.c fix incorrect DTMF Digit_Duration. it's always
+ short by 'hits_to_begin*DTMF_GSIZE', or 25.5ms if hitstobegin=2
+ (issue ASTERISK-16003) Tested by: alecdavis alecdavis (license
+ 585) Review https://reviewboard.asterisk.org/r/2145/
+
+2012-10-04 17:39 +0000 [r374456] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix a regression from direct media ACLs
+ where the directrtpsetup option no longer works. A check was
+ added for direct media ACLs that immediately forbid remote
+ bridging if there was no bridged channel. This caused
+ directrtpsetup to no longer function as it needs this information
+ before bridging actually occurs. Logic has now been adjusted so
+ if there is no bridged channel a remote bridge will still be
+ attempted. (closes issue ASTERISK-20511) Reported by: kristoff
+ Review: https://reviewboard.asterisk.org/r/2146/
+
+2012-10-04 15:25 +0000 [r374426] dlee <dlee@localhost>:
+
+ * main/db.c, res/res_agi.c: Fix DBDelTree error codes for AMI, CLI
+ and AGI The AMI DBDelTree command will return Success/Key tree
+ deleted successfully even if the given key does not exist. The
+ CLI command 'database deltree' had a similar problem, but was
+ saved because it actually responded with '0 database entries
+ removed'. AGI had a slightly different error, where it would
+ return success if the database was unavailable. This came from
+ confusion about the ast_db_deltree retval, which is -1 in the
+ event of a database error, or number of entries deleted
+ (including 0 for deleting nothing). * Adds a Doxygen comment to
+ process_db_keys explaining its retval * Changed some poorly named
+ res variables to num_deleted * Specified specific errors when
+ calling ast_db_deltree (database unavailable vs. entry not found
+ vs. success) * Fixed similar bug in AGI database deltree, where
+ 'Database unavailable' results in successful result (closes issue
+ AST-967) Reported by: John Bigelow Review:
+ https://reviewboard.asterisk.org/r/2138/
+
+2012-10-04 04:39 +0000 [r374365-374384] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * CHANGES, main/dsp.c, configs/dsp.conf.sample: dsp.c User
+ configuration of DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values
+ Asterisk's DTMF Specifications are based on AT&T specs, which may
+ not be compatible in other countries. Various countries have
+ different specifications for the maximum power level differences
+ between the DTMF low group and high group of frequencies. Power
+ level difference between frequencies for different
+ Administrations/RPOAs NTT = Max. 5 dB AT&T = 4dB(reverse) to
+ 8dB(normal) Danish = Max. 6 dB Australian = Max. 10 dB Brazilian
+ = Max. 9 dB ETSI = Max. 6 dB from ETSI ES 201 235-3 V1.3.1
+ (2006-03) Now allow 4 variables to be individually configured in
+ dsp.conf, with reasonable min/max of 2dB to 20dB. Default is AT&T
+ specifications Add's the following variables to dsp.conf
+ ;dtmf_normal_twist=6.31 ;dtmf_reverse_twist=2.51
+ ;relax_dtmf_normal_twist=6.31 ;relax_dtmf_reverse_twist=3.98
+ (closes issue ASTERISK-20442) Reported by: tbsky Tested by:
+ tbsky,alecdavis alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2141/
+
+ * main/dsp.c: _dsp_init: bring inline with trunk preparation for
+ clean merge of DTMF TWIST patch No functional changes, just
+ style. alecdavis (license 585) Reported by: Alec Davis Tested by:
+ alecdavis related https://reviewboard.asterisk.org/r/2141
+
+2012-10-04 02:09 +0000 [r374177-374335] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_jabber.c: Check for presence of buddy in info/dinfo
+ handlers The res_jabber resource module uses the ASTOBJ library
+ for managing its ref counted objects. After calling
+ ASTOBJ_CONTAINER_FIND to locate a buddy object, the pointer to
+ the object has to be checked to see if the buddy existed. Prior
+ to this patch, the buddy object was not checked for NULL; with
+ this patch in both aji_client_info_handler and aji_dinfo_handler
+ the pointer is checked before used and, if no buddy object was
+ found, the handlers return an error code. This patch does not
+ take the approach that our JID can be used to log in from another
+ resource. If that approach is desired, an improvement could be
+ made to this patch to create the buddy on the fly. This patch
+ seeks only to prevent Asterisk from crashing. Note that multiple
+ people have proposed patches for this issue; the patch being
+ committed here is based on those. (closes issue ASTERISK-19532)
+ Reported by: Karsten Wemheuer Tested by: Byron Clark patches:
+ fix-jabber uploaded by Karsten Wemheuer (license #5930)
+ xmpp_no_crash_with_ejabberd.patch uploaded by Byron Clark
+ (license #6157) (closes issue ASTERISK-19557) Reported by:
+ ulugutz
+
+ * main/ccss.c: Destroy the generic_monitors container after the
+ core_instances in ccss For each item in core_instances disposed
+ of in the shutdown of ccss, any generic monitor instances
+ referenced by the objects will be removed from generic_monitors
+ during their destruction. Hilarity ensues if generic_monitors no
+ longer exists. Thanks to the Asterisk Test Suite's generic_ccss
+ test for complaining loudly when it ran into this.
+
+ * main/asterisk.c: Ensure Shutdown AMI event is still fired during
+ Asterisk shutdown Richard pointed out that having the manager
+ dispose of itself gracefully during shutdown meant that the
+ Shutdown event will no longer get fired. This patch moves the AMI
+ event just prior to running the atexit callbacks.
+
+ * main/event.c, main/taskprocessor.c, res/res_musiconhold.c,
+ main/cel.c, main/indications.c, main/channel.c, main/data.c,
+ main/pbx.c, main/manager.c, main/ccss.c, main/features.c: Fix a
+ variety of ref counting issues This patch resolves a number of
+ ref leaks that occur primarily on Asterisk shutdown. It adds a
+ variety of shutdown routines to core portions of Asterisk such
+ that they can reclaim resources allocate duringd initialization.
+ Review: https://reviewboard.asterisk.org/r/2137
+
+2012-10-01 16:45 +0000 [r374108] Sean Bright <sean@malleable.com>
+
+ * tests/test_db.c, apps/app_queue.c, main/db.c,
+ include/asterisk/astdb.h: app_queue: Support persisting and
+ loading of long member lists. Greenlight in #asterisk brought up
+ that he was receiving an error message "Could not create
+ persistent member string, out of space" when running app_queue in
+ Asterisk 10. dump_queue_members() made an assumption that 8K
+ would be enough to store the generated string, but with queues
+ that have large member lists this is not always the case. This
+ patch removes the limitation and uses ast_str instead of a fixed
+ sized buffer. The complicating factor comes from the fact that
+ ast_db_get requires a buffer and buffer size argument, which
+ doesn't let us pull back more than what we pass in, so I
+ introduced a new ast_db_get_allocated() which returns an
+ ast_strdup()'d copy of the value from astdb. As an aside, I did
+ some testing on the maximum size of data that we can store in the
+ BDB library we distribute and was able to store a 10MB string and
+ retrieve it with no problems, so I feel this is a safe patch.
+ Review: https://reviewboard.asterisk.org/r/2136/
+
+2012-09-28 19:03 +0000 [r374032] Jonathan Rose <jrose@digium.com>
+
+ * res/res_jabber.c: res_jabber: Remove CLI command 'jabber test'
+ The opinion of development was that it is both improper to have
+ Matt's personal email address used in the source and that the
+ command wouldn't be useful without it. (closes issue AST-467)
+ Reported by: Malcolm Davenport
+
+2012-09-28 12:14 +0000 [r373989] Joshua Colp <jcolp@digium.com>
+
+ * res/res_agi.c: Update documentation to make it explicit that
+ "stream file" will not restart musiconhold. (issue
+ ASTERISK-17367) Reported by: oej
+
+2012-09-27 22:08 +0000 [r373945] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_senddtmf.c: Fix SendDTMF crash and channel reference
+ leak using channel name parameter. The SendDTMF channel name
+ parameter has two issues. 1) Crashes if the channel name does not
+ exist. 2) Leaks a channel reference if the channel is the current
+ channel. Problem introduced by ASTERISK-15956. * Updated SendDTMF
+ documentation. * Renamed app to senddtmf_name and tweaked the
+ type.
+
+2012-09-27 16:49 +0000 [r373878-373909] Joshua Colp <jcolp@digium.com>
+
+ * main/loader.c: loader: Ensure dependent modules are properly
+ initialized. If an Asterisk module specifies a dependency in
+ ast_module_info.nonoptreq, it is possible for Asterisk to skip
+ calling the modules's .load function. Asterisk was loading and
+ linking the module via load_dynamic_module() but was not adding
+ the module to the resource_heap. Therefore the module was not
+ initialized based on it's priority along with the other modules
+ in the heap. Now use load_resource() instead of
+ load_dynamic_module() for non-optional requirement. This will add
+ the module to the resource_heap so the module can be properly
+ initialized in the correct order. This is required if there are
+ any module global data structures initialized in the .load()
+ callback for the module on platforms which do not support weak
+ references. (issue ASTERISK-20439) Reported by: sruffell Patches:
+ 0001-loader-Ensure-dependent-modules-are-properly-initial.patch
+ uploaded by sruffell (license 5417)
+
+ * channels/chan_local.c: Fix an issue where Local channels dialed
+ by app_queue are considered in use immediately. The chan_local
+ channel driver returns a device state of in use even if a created
+ Local channel has not yet been dialed. This fix changes the logic
+ to return a state of not in use until the channel itself has been
+ dialed. (closes issue ASTERISK-20390) Reported by: tim_ringenbach
+ Review: https://reviewboard.asterisk.org/r/2116/
+
+2012-09-26 21:11 +0000 [r373848] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Move handling of 408 response so there is no
+ misleading warning message. (closes issue ASTERISK-20060)
+ Reported by: Walter Doekes
+
+2012-09-26 18:04 +0000 [r373815] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_meetme.c: Fixed meetme tab completion and command
+ documentation. * Removed unnecessary case sensitivity in meetme
+ list, lock, unlock, mute, unmute, and kick commands. * Separated
+ meetme lock/unlock, mute/unmute, and kick commands into their own
+ registered commands to simplify tab completion and parameter
+ checking. meetme_lock_cmd(), meetme_mute_cmd(), and
+ meetme_kick_cmd() * Simplified meetme_show_cmd() (closes issue
+ AST-1006) Reported by: John Bigelow Tested by: rmudgett
+
+2012-09-25 23:07 +0000 [r373735-373773] Mark Michelson <mmichelson@digium.com>
+
+ * main/say.c: Fix saying of date in Dutch. The Dutch say the date
+ before the month. (closes issue ASTERISK-20353) Reported by: Teun
+ Ouwehand
+
+ * configs/agents.conf.sample, channels/chan_agent.c: Remove dead
+ code and documentation for nonexistent feature. multiplelogin was
+ removed from chan_agent back in 1.6.0 when AgentCallbackLogin()
+ was removed. (closes issue AST-948) reported by Steve Pitts
+
+ * apps/app_voicemail.c: Fix error where improper IMAP greetings
+ would be deleted. (closes issue ASTERISK-20435) Reported by:
+ fhackenberger Patches: asterisk-20435-imap-del-greeting.diff
+ uploaded by Michael L. Young (License #5026) (with suggested
+ modification made by me)
+
+2012-09-25 20:10 +0000 [r373705] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_local.c: Fix T.38 support when used with chan_local
+ in between. Users of the T.38 API can indicate
+ AST_T38_REQUEST_PARMS on a channel to request that the channel
+ indicate a T.38 negotiation with the parameters present on the
+ channel. The return value of this indication is expected to be
+ AST_T38_REQUEST_PARMS upon success but with chan_local involved
+ this could never occur. This fix changes chan_local to always
+ return AST_T38_REQUEST_PARMS for this situation. If the
+ underlying channel technology on the other side does not support
+ T.38 this would have been determined ahead of time using
+ ast_channel_get_t38_state and an indication would not occur.
+ (closes issue ASTERISK-20229) Reported by: wdoekes Patches:
+ ASTERISK-20229.patch uploaded by wdoekes (license 5674) Review:
+ https://reviewboard.asterisk.org/r/2070/
+
+2012-09-25 19:32 +0000 [r373666-373702] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_rtp_asterisk.c: Fix an issue where media would not flow
+ for situations where the legacy STUN code is in use. The STUN
+ packets should *not* be blocked by strict RTP. (closes issue
+ ASTERISK-20415) Reported-by: Michele Cicciotti Patch-by: Josh
+ Colp (trunk r369817)
+
+ * apps/app_queue.c: "show" completion option for "queue" shouldn't
+ appear twice When tab-completing CLI commands starting with
+ "queue", "show" appeared twice in the list due to the way that
+ Asterisk's tab completion functions and the order in which the
+ commands were registered. The registration order has been altered
+ to resolve this issue. (closes issue AST-940) Reported-by: Steve
+ Pitts
+
+2012-09-25 17:21 +0000 [r373652] Terry Wilson <twilson@digium.com>
+
+ * configs/sip.conf.sample, channels/sip/include/sip.h,
+ channels/chan_sip.c: Properly handle UAC/UAS roles for SIP
+ session timers The SIP session timer mechanism contains a
+ mandatory 'refresher' parameter (included in the Session-Expires
+ header) which is used in the session timer offer/answer signaling
+ within a SIP Invite dialog. It looks like asterisk is
+ interpreting the uac resp. uas role only as the initial role of
+ client and server (caller is uac, callee is uas). The standard
+ rfc 4028 however assigns the client role to the ((RE)-Invite)
+ requester, the server role to the ((RE)-Invite) responder. This
+ patch has Asterisk track the actual refresher as "us" or "them"
+ as opposed to relying on just the configured "uas" or "uac"
+ properties. (closes issue AST-922) Reported by: Thomas Airmont
+ Review: https://reviewboard.asterisk.org/r/2118/
+
+2012-09-25 17:18 +0000 [r373618-373640] Richard Mudgett <rmudgett@digium.com>
+
+ * codecs/ilbc/iLBC_decode.c, codecs/ilbc/iLBC_encode.c: Fix
+ valgrind found memcpy issues in codec_ilbc. Valgrind found
+ codec_ilbc using memcpy instead of memmove for overlapping memory
+ blocks. (issue ASTERISK-19890) (closes issue ASTERISK-20231)
+ Reported by: Walter Doekes Patches: ASTERISK-20231.patch (license
+ #5674) patch uploaded by Walter Doekes
+
+ * codecs/Makefile: Make rebuild GSM, ilbc, or lpc10 codecs if the
+ respective sources change.
+
+2012-09-25 16:15 +0000 [r373617] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c: chan_sip: Set Quality of Service for video
+ rtp instance (closes issue ASTERISK-20201) Reported by: ddkprog
+ Patches: chan_sip.c.diff uploaded by ddkprog (license 6008)
+
+2012-09-25 13:27 +0000 [r373578] Kinsey Moore <kmoore@digium.com>
+
+ * configs/res_odbc.conf.sample: Fix documentation for default
+ username in res_odbc This was previously stated to be "root", but
+ is actually the name of the context if unspecified. (closes issue
+ ASTERISK-20258) Reported by: Stefan x
+
+2012-09-25 11:58 +0000 [r373532-373550] Joshua Colp <jcolp@digium.com>
+
+ * res/res_rtp_multicast.c: Fix an issue where a caller to ast_write
+ on a MulticastRTP channel would determine it failed when in
+ reality it did not. When sending RTP packets via multicast the
+ amount of data sent is stored in a variable and returned from the
+ write function. This is incorrect as any non-zero value returned
+ is considered a failure while a return value of 0 is success. For
+ callers (such as ast_streamfile) that checked the return value
+ they would have considered it a failure when in reality nothing
+ went wrong and it was actually a success. The write function for
+ the multicast RTP engine now returns -1 on failure and 0 on
+ success, as it should. (closes issue ASTERISK-17254) Reported by:
+ wybecom
+
+ * channels/chan_sip.c: Add missing checks that I neglected. The SIP
+ technology and SIP info technology should be considered equal.
+ (closes issue ASTERISK-20409) Reported by: michele cicciotti
+ privatewave
+
+2012-09-24 22:15 +0000 [r373504] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_rtp_asterisk.c: Revert change to res_rtp_asterisk
+ committed in r373236 (1.8) The change committed in r373236
+ attempted to account for endpoints that increased their RTP
+ timestamp in DTMF end of event re-transmissions. This change
+ attempted to make Asterisk continue to work with endpoints that
+ failed to follow the RFC while maintaining the fix that allowed
+ for out of order DTMF to be handled. Unfortunately, there is no
+ free lunch, and this patch broke any system that sent DTMF
+ immediately after an RTP session was established or when an SSRC
+ is updated. As such, that patch is being reverted for the
+ previous behavior. Endpoints that erroneously increase the RTP
+ timestamp in DTMF end of event packets will not work properly
+ with Asterisk. (issue ASTERISK-20424)
+
+2012-09-24 22:09 +0000 [r373500] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Be consistent, send From: "Anonymous"
+ <sip:anonymous@anonymous.invalid> When setting
+ CALLERID(pres)=unavailable in the dialplan, the From header in
+ the SIP message contains "Anonymous"
+ <sip:Anonymous@anonymous.invalid>. For consistency, Asterisk
+ should use a lowercase a in the userpart of the URI. * Make the
+ From header use a lowercase A in the userpart of the anonymous
+ URI. (closes issue ASTERISK-19838) Reported by: Antti Yrjola
+ Patches: chan_sip_patch_ASTERISK-19838.patch (license #6383)
+ patch uploaded by Antti Yrjola
+
+2012-09-24 20:57 +0000 [r373467] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_mixmonitor.c, funcs/func_audiohookinherit.c:
+ func_audiohookinherit: Document some missed sources. This patch
+ also mentions that AUDIOHOOK_INHERIT can be used to transfer
+ MixMonitor audiohooks. There is also wiki that addresses
+ audiohooks and the use of AUDIOHOOK_INHERIT at the following
+ link: https://wiki.asterisk.org/wiki/display/AST/Audiohooks
+ (closes issue ASTERISK-18220) Reported by: Ishfaq Malik
+
+2012-09-24 19:15 +0000 [r373438] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix a deadlock caused by a race condition
+ between removing a hint and reloading the dialplan and
+ subscribing to the removed hint. If conditions were right it was
+ possible for both the PBX core and chan_sip to deadlock by both
+ having a lock that the other wants. In the case of the PBX core
+ it had the contexts lock and wanted a SIP dialog lock, while in
+ the case of chan_sip it had the SIP dialog lock and wanted the
+ contexts lock. This fix unlocks the SIP dialog before getting the
+ extension state so that the other thread will not block on trying
+ to lock it. Once the extension state is retrieved the SIP dialog
+ is locked again and life carries on. As the SIP dialog is
+ reference counted it is not possible for it to go away after
+ unlocking. (closes issue ASTERISK-20437) Reported by: jhutchins
+
+2012-09-24 15:40 +0000 [r373424] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Fix potential reentrancy problems in
+ chan_sip. Asterisk v1.8 and later was not as vulnerable to this
+ issue. * Made find_call() lock each private as it processes the
+ found dialogs. (Primary cause of ABE-2876) * Made the other
+ functions that traverse the dialogs container lock each private
+ as it examines them. * Fix race condition in sip_call() if the
+ thread that sent the INVITE is held up long enough for a response
+ to be processed. The p->initid for the INVITE retransmission
+ could be added after it was canceled by the response processing.
+ * Made __sip_destroy() clean up resource pointers after freeing.
+ This is primarily defensive in case someone has a stale private
+ pointer. * Removed redundant memset() in reqprep(). The call to
+ init_req() already does the memset() and is the first reference
+ to req in reqprep(). * Removed useless set of req.method in
+ transmit_invite(). The calls to initreqprep() and reqprep() have
+ to do this because they memset() the req. JIRA ABE-2876
+ .......... Merged -r373423 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+
+2012-09-21 19:00 +0000 [r373298-373342] Jonathan Rose <jrose@digium.com>
+
+ * channels/iax2-provision.c: iax2-provision: Fix improper return on
+ failed cache retrieval (closes issue ASTERISK-20337) reported by:
+ John Covert Patches: iax2-provision.c.patch uploaded by John
+ Covert (license 5512)
+
+ * apps/app_queue.c: app_queue: Make queue reload members and
+ variants of that work Prior to this patch, 'queue reload members'
+ cli command did not work at all. This also affects the manager
+ function 'QueueReload' when supplied with the 'members: yes'
+ field. (closes issue AST-956) Reported by: John Bigelow
+
+2012-09-20 19:12 +0000 [r373242] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_meetme.c: Fix incorrect MeetME conference bridge
+ reference count decrementing and sometimes premature destruction.
+ When using the 'e' or 'E' option to MeetMe the configured
+ conference bridges are loaded and examined to see if any are
+ empty. If no conference bridges are empty the caller is prompted
+ to enter the number of one. This operation left around a pointer
+ to the last created conference bridge still containing
+ participants. When the caller that was not able to find any empty
+ conference bridge hung up this pointer was disposed of and the
+ reference count of the conference bridge decremented. If there
+ was only a single participant in the conference bridge it was
+ ultimately destroyed prematurely. (closes issue AST-994) Reported
+ by: John Bigelow
+
+2012-09-20 18:41 +0000 [r373236] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_rtp_asterisk.c: When processing RFC 2833 DTMF, accomodate
+ increasing timestamps in End events While endpoints should not be
+ changing the source timestamp between DTMF event packets, the
+ fact is there exists those endpoints that do exactly that. To
+ work around this, we absorb timestamps within the expected
+ re-transmit period. Note that this period only affects End of
+ Event packets, so it should not prevent the detection of new DTMF
+ digits that happen to arrive right on top of each other. (closes
+ issue ASTERISK-20424) Reported by: Vladimir Mikhelson Tested by:
+ mjordan, Vladimir Mikhelson Review:
+ https://reviewboard.asterisk.org/r/2124
+
+2012-09-19 16:02 +0000 [r373165] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix a regression where direct media was not
+ permitted for calls using SIP INFO DTMF. A change was committed
+ to fix direct media ACL support. This change wrongly assumed that
+ only a single channel technology structure exists for chan_sip.
+ This is in fact false as a second exists for calls using SIP INFO
+ DTMF. The code which performs direct media ACL checking now
+ checks for both the non-INFO DTMF and INFO DTMF channel
+ technology structures. (closes issue ASTERISK-20409) Reported by:
+ michele cicciotti privatewave
+
+2012-09-18 20:12 +0000 [r373131] Sean Bright <sean@malleable.com>
+
+ * main/manager.c: Don't crash when passing a NULL message to
+ __astman_get_header. Before this commit, __astman_get_header
+ would blindly dereference the passed in 'struct message *' to
+ traverse the header list. There are cases, however, such as
+ '*CLI> sip qualify peer foo' where the message pointer is NULL,
+ so we need to check for that.
+
+2012-09-15 00:13 +0000 [r373090] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_ss7.c: Made companding law for SS7 calls only
+ determined by SS7 signaling type. For SS7, the companding law for
+ a call was chosen inconsistently depending upon ss7type (ITU vs
+ ANSI) and the DAHDI companding default (T1 vs E1). For incoming
+ calls, the companding law was determined by ss7type. For outgoing
+ calls, the companding law was determined by the DAHDI default.
+ With the wrong combination you would get A-law/u-law conflicts.
+ An A-law/u-law conflict sounds like bad static on the line. SS7
+ ITU signaling with E1 line: ok SS7 ITU signaling with T1 line:
+ noise SS7 ANSI signaling with E1 line: noise SS7 ANSI signaling
+ with T1 line: ok * Fix the companding law used to be determined
+ by the SS7 signaling type only.
+
+2012-09-14 19:07 +0000 [r373061] Matthew Jordan <mjordan@digium.com>
+
+ * main/ssl.c, main/tcptls.c, channels/chan_sip.c: Resolve memory
+ leaks in TLS initialization and TLS client connections This patch
+ resolves two sources of memory leaks when using TLS in Asterisk:
+ 1) It removes improper initialization (and multiple
+ re-initializations) of portions of the SSL library. Asterisk
+ calls SSL_library_init and SSL_load_error_strings during SSL
+ initialization; collectively this obviates the need for calling
+ any of the following during initialization or client connection
+ handling: * ERR_load_crypto_strings (handled by
+ SSL_load_error_strings) * OpenSSL_add_all_algorithms (synonym for
+ SSL_library_init) * SSLeay_add_ssl_algorithms (synonym for
+ SSL_library_init) 2) Failure to completely clean up all memory
+ allocated by Asterisk and by the SSL library for TLS clients.
+ This included not freeing the SSL_CTX object in the SIP channel
+ driver, as well as not clearing the error stack when the TLS
+ client exited. Note that these memory leaks were found by Thomas
+ Arimont, and this patch was essentially written by him with some
+ minor tweaks. (closes issue AST-889) Reported by: Thomas Arimont
+ Tested by: Thomas Arimont patches: (bugAST-889.patch) by Thomas
+ Arimont (license 5525) Review:
+ https://reviewboard.asterisk.org/r/2105
+
+2012-09-13 18:39 +0000 [r373024] dlee <dlee@localhost>:
+
+ * include/asterisk/channel.h, main/channel.c: Fix timeouts for
+ ast_waitfordigit[_full]. ast_waitfordigit_full would simply pass
+ its timeout to ast_waitfor_nandfds, expecting it to decrement the
+ timeout by however many milliseconds were waited. This is a
+ problem if it consistently waits less than 1ms. The timeout will
+ never be decremented, and we wait... FOREVER! This patch makes
+ ast_waitfordigit_full manage the timeout itself. It maintains the
+ previously undocumented behavior that negative timeouts wait
+ forever. (closes issue ASTERISK-20375) Reported by: Mark
+ Michelson Tested by: Mark Michelson Review:
+ https://reviewboard.asterisk.org/r/2109/
+
+2012-09-13 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.17.0-rc1 Released.
+
+2012-09-12 15:42 +0000 [r372959] Matthew Jordan <mjordan@digium.com>
+
+ * main/astobj2.c, include/asterisk/astobj2.h: Constify
+ __ao2_ref_debug in astobj2 When REF_DEBUG is enabled in certain
+ files - most notably ccss.c - the 'tag' parameter passed to
+ __ao2_ref_debug will be a const char *. The function currently
+ expects that parameter to not be const. This causes a warning
+ when compiling, as the const qualifier is being discarded. With
+ dev-mode enabled, this prevents compiling Asterisk. This patch
+ makes __ao2_ref_debug's tag and file parameters const. (closes
+ issue ASTERISK-20408) Reported by: mjordan
+
+2012-09-12 14:51 +0000 [r372932] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Add channel name to a warning to make
+ debugging easier. The "autodestruct with owner in place" message
+ is typically indicative of a channel reference leak. Printing out
+ the name of the channel in the message may be helpful when trying
+ to debug the issue.
+
+2012-09-11 22:11 +0000 [r372902] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_local.c: chan_local: Switch from using a random 4
+ digit hex identifier to unique id Changes chan_local channels to
+ use an 8 digit hex identifier generated atomically and
+ sequentially in order to eliminate the chance of having multiple
+ channels with the same name during high call volume situations.
+ (issue ASTERISK-20318) Reported by: Dan Cropp Review:
+ https://reviewboard.asterisk.org/r/2104/
+
+2012-09-11 15:26 +0000 [r372840] Mark Michelson <mmichelson@digium.com>
+
+ * main/features.c: Fix bad channel application data reference. When
+ channels get bridged due to an AMI bridge action or a DTMF
+ attended transfer, the two channels that get bridged have their
+ application data pointing to the other channel's name. This means
+ that if one channel is hung up but the other moves on, it means
+ that the channel that moves on will have its application data
+ pointing at freed memory. (issue ASTERISK-20335) Reported by:
+ aragon
+
+2012-09-10 20:53 +0000 [r372804] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_iax2.c: Ensure iax2 debug output is displayed when
+ expected When IAX2 debug was changed from iax_showframe to
+ iax_outputframe, some instances were missed (or added afterward).
+ This was causing debug output to not be displayed when expected.
+ (closes issue ASTERISK-20338) Reported-by: John Covert Patch-by:
+ John Covert
+
+2012-09-10 18:35 +0000 [r372765] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_meetme.c: app_meetme: Document that 'p' option will
+ continue in dialplan. (closes issue AST-991) Reported by John
+ Bigelow
+
+2012-09-10 18:31 +0000 [r372763] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c: Warn on CLI when UDPTL init fails This adds
+ a CLI warning when a SDP offer is rejected due to UDPTL
+ initialization failure. Previously, there was no indication of
+ the reason for offer rejection in this case. (closes issue
+ ASTERISK-20357) Reported-by: Francesco Usseglio Gaudi
+
+2012-09-10 17:07 +0000 [r372736] Jonathan Rose <jrose@digium.com>
+
+ * main/channel.c: Masquerade: Retain parkinglot settings made by
+ CHANNEL function. Prior to this patch, the user would have a
+ parkinglot set on a channel that was parked and when the channel
+ was retrieved, any attempt by that channel to park would simply
+ use the default. This patch makes parkinglot values set in this
+ way be retained through the masquerade. (closes issue AST-990)
+ Reported by: Nick Huskinson Patches:
+ masquerade_parkinglot_patch.diff Uploaded by Jonathan Rose
+ (license 6182)
+
+2012-09-09 01:19 +0000 [r372709] Matthew Jordan <mjordan@digium.com>
+
+ * channels/sip/sdp_crypto.c: Only re-create an SRTP session when
+ needed; respond with correct crypto policy In r356604, SRTP
+ handling was fixed to accomodate multiple crypto keys in an SDP
+ offer and the ability to re-create an SRTP session when the
+ crypto keys changed. In certain circumstances - most notably when
+ a phone is put on hold after having been bridged for a
+ significant amount of time - the act of re-creating the SRTP
+ session causes problems for certain models of phones. The patch
+ committed in r356604 always re-created the SRTP session
+ regardless of whether or not the cryptographic keys changed.
+ Since this is technically not necessary, this patch modifies the
+ behavior to only re-create the SRTP session if Asterisk detects
+ that the remote key has changed. This allows models of phones
+ that do not handle the SRTP session changing to continue to work,
+ while also providing the behavior needed for those phones that do
+ re-negotiate cryptographic keys. In addition, in Asterisk 1.8
+ only, it was found that phones that offer AES_CM_128_HMAC_SHA1_32
+ will end up with no audio if the phone is the initiator of the
+ call. The phone will send an INVITE request specifying that
+ AES_CM_128_HMAC_SHA1_32 be used for the cryptographic policy;
+ Asterisk will set its policy to that value. Unfortunately, when
+ the call is Answered and a 200 OK is sent back to the UA, the
+ policy sent in the response's SDP will be the hard coded value
+ AES_CM_128_HMAC_SHA1_80. This potentially results in Asterisk
+ using the INVITE request's policy of AES_CM_128_HMAC_SHA1_32,
+ while the phone uses Asterisk's response of
+ AES_CM_128_HMAC_SHA1_80. Hilarity ensues as both endpoints think
+ the other is crazy. This patch fixes that by caching the policy
+ from the request and responding with it. Note that this is not a
+ problem in Asterisk 10 and later, as the ability to configure the
+ policy was added in that version. (issue ASTERISK-20194) Reported
+ by: Nicolo Mazzon Tested by: Nicolo Mazzon Review:
+ https://reviewboard.asterisk.org/r/2099
+
+2012-09-08 03:54 +0000 [r372682] dlee <dlee@localhost>:
+
+ * main/Makefile: Add OPENSSL_INCLUDE to the CFLAGS for ssl.c and
+ tcptls.c. Without this flag, those files will compile with the
+ system installed OpenSSL headers (if they exist). This is a real
+ bummer if a different path was specified using --with-ssl=
+ (closes issue ASTERISK-20392)
+
+2012-09-07 23:05 +0000 [r372620-372655] Richard Mudgett <rmudgett@digium.com>
+
+ * main/astmm.c: Fix MALLOC_DEBUG version of ast_strndup(). (closes
+ issue ASTERISK-20349) Reported by: Brent Eagles
+
+ * funcs/func_math.c: Remove annoying unconditional debug message
+ from INC/DEC functions. (closes issue AST-1001) Reported by:
+ Guenther Kelleter
+
+ * apps/app_queue.c: Fix exception path typo in app_queue.c
+ try_calling(). (closes issue ASTERISK-20380) Reported by: Jeremy
+ Pepper Patches: fix-local-channel-locking.patch (license #6350)
+ patch uploaded by Jeremy Pepper
+
+ * apps/app_voicemail.c: Fix VoicemailUserEntry event headers
+ ServerEmail and MailCommand reported values. The AMI action
+ VoicemailUsersList VoicemailUserEntry event headers ServerEmail
+ and MailCommand did not report the global values if they were not
+ overridden. The VoicemailUserEntry event header ServerEmail was
+ not populated with the global value if the voicemail user did not
+ override it. The VoicemailUserEntry event header MailCommand was
+ never populated with a value. * Removed unused struct ast_vm_user
+ member mailcmd[]. (closes issue AST-973) Reported by: John
+ Bigelow Tested by: rmudgett
+
+2012-09-07 02:24 +0000 [r372554-372581] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_minivm.c: Free ast_str objects when temp file fails to
+ be created in MiniVM The previous commit (r372554) was from a
+ patch that was written before r366880, which ensured that ast_str
+ objects allocated in the sendmail routine were free'd in off
+ nominal paths. This commit frees the string objects in the off
+ nominal path introduced in r372554. (issue ASTERISK-17133)
+ Reported by: Tzafrir Cohen
+
+ * apps/app_minivm.c: Fix file descriptor leak and pointer scope
+ issue in MiniVM when sending mail When MiniVM sends an e-mail and
+ it has the volgain option set, it will spawn sox in a separate
+ process to handle the manipulation of the sound file. In doing
+ so, it creates a temporary file. There are two problems here: 1)
+ The file descriptor returned from mkstemp is leaked 2) The
+ finalfilename character pointer points to a buffer that loses
+ scope once volgain processing is finished. Note that in r316265,
+ Russell fixed some gcc warnings by using the return value of the
+ mkstemp call. A warning was placed in minivm that the file
+ descriptor was going to be leaked. This patch reverts that
+ change, as it handles the leak and 'uses' the file descriptor
+ returned from mkstemp. (closes issue ASTERISK-17133) Reported by:
+ Tzafrir Cohen patches: minivm_18501_demo.diff uploaded by Tzafrir
+ Cohen (license #5035)
+
+2012-09-06 21:38 +0000 [r372517] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_queue.c: Ensure listed queues are not offered for
+ completion When using tab-completion for the list of queues on
+ "queue reset stats" or "queue reload
+ {all|members|parameters|rules}", the tab-completion listing for
+ further queues erroneously listed queues that had already been
+ added to the list. The tab-completion listing now only displays
+ queues that are not already in the list. (closes issue AST-963)
+ Reported-by: John Bigelow
+
+2012-09-06 18:54 +0000 [r372498] dsessions <dsessions@localhost>:
+
+ * configs/res_ldap.conf.sample, channels/chan_sip.c: LDAP Realtime
+ Peers Cannot Register Prior to 1.8, it was not necessary for an
+ explicit "type" to be set for an asterisk LDAP realtime peer. Now
+ the routine find_peer actually checks the type field during
+ registration and fails to find the peer if it is not set. The
+ attached patches make the realtime type equal whatever type is
+ being searched for if the type is 0 upon return from routine
+ build_peer. (closes issue ASTERISK-17222) Reported by: John
+ Covert Patch by: David Vossel Tested by: Darren Sessions Review:
+ https://reviewboard.asterisk.org/r/2095/
+
+2012-09-06 15:52 +0000 [r372471] Jonathan Rose <jrose@digium.com>
+
+ * UPGRADE.txt: chan_sip: Note change in behavior to how
+ directmediapermit/deny ACL works r366547 introduced a change to
+ the directmedia ACL for chan_sip which modified the behavior
+ significantly. Prior to the patch, this option would bridge peers
+ with directmedia if a peer's IP address matched its own
+ directmedia ACL. After that patch, the peer would check the
+ bridged peer's ACL instead. This change has been present since
+ 1.8.14.0. That patched failed to document the change in
+ Upgrade.txt, so this patch adds mention of that change to
+ UPGRADE.txt (UPGRADE-1.8.txt in newer branches) (issue AST-876)
+
+2012-09-06 14:28 +0000 [r372444] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_queue.c: Ensure "rules" is tab-completable for "queue
+ show" Previously, tabbing at the end of "queue show" produced a
+ list of available queues about which information could be shown,
+ but did not include an alternative command, "rules", to access
+ information about queue rules. The "rules" item should now be
+ shown in the list of tab-completable items. (closes issue
+ AST-958) Reported-by: John Bigelow
+
+2012-09-06 02:48 +0000 [r372390-372417] Matthew Jordan <mjordan@digium.com>
+
+ * pbx/pbx_dundi.c: Fix DUNDi message routing bug when neighboring
+ peer is unreachable Consider a scenario where DUNDi peer PBX1 has
+ two peers that are its neighbors, PBX2 and PBX3, and where PBX2
+ and PBX3 are also neighbors. If the connection is temporarily
+ broken between PBX1 and PBX3, PBX1 should not include PBX3 in the
+ list of peers it sends to PBX2 in a DPDISCOVER message, as it
+ cannot send messages to PBX3. If it does, PBX2 will assume that
+ PBX3 already received the message and fail to forward the message
+ on to PBX3 itself. This patch fixes this by only including peers
+ in a DPDISCOVER message that are reachable by the sending node.
+ This includes all peers with an empty address (00:00:00:00:00:00)
+ and that are have been reached by a qualify message. This patch
+ also prevents attempting to qualify a dynamic peer with an empty
+ address until that peer registers. (closes issue ASTERISK-19309)
+ Reported by: Peter Racz patches: dundi_routing.patch uploaded by
+ Peter Racz (license 6290) The patch uploaded by Peter was
+ modified slightly for this commit.
+
+ * apps/app_followme.c: Allow configured numbers for FollowMe to be
+ greater than 90 characters When parsing a 'number' defined in
+ followme.conf, FollowMe previously parsed the number in the
+ configuration file into a buffer with a length of 90 characters.
+ This can artificially limit some parallel dial scenarios. This
+ patch allows for numbers of any length to be defined in the
+ configuration file. Note that Clod Patry originally wrote a patch
+ to fix this problem and received a Ship It! on the JIRA issue.
+ The patch originally expanded the buffer to 256 characters.
+ Instead, the patch being committed duplicates the string in the
+ config file on the stack before parsing it for consumption by the
+ application. (closes issue ASTERISK-16879) Reported by: Clod
+ Patry Tested by: mjordan patches: followme_no_limit.diff uploaded
+ by Clod Patry (license #5138) Slightly modified for this commit.
+
+2012-09-05 19:20 +0000 [r372354] Kinsey Moore <kmoore@digium.com>
+
+ * main/manager.c: Correct documentation for ModuleLoad AMI action
+ The documentation incorrectly listed 'rtp' as a reloadable
+ subsystem and left out many other reloadable subsystems. It is
+ now also documented that subsystems may only be reloaded, not
+ loaded or unloaded. (closes issue AST-977) Reported-by: John
+ Bigelow
+
+2012-09-05 18:34 +0000 [r372339] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/dsp.c: dsp.c: in ast_mf_detect_init incorrectly sets
+ goertzel samples to 160, should be MF_GSIZE Related
+ https://reviewboard.asterisk.org/r/2097/
+
+2012-09-05 18:29 +0000 [r372337] Kinsey Moore <kmoore@digium.com>
+
+ * main/pbx.c: Ensure counts generated in
+ manager_show_dialplan_helper are correct When
+ manager_show_dialplan_helper was written, the counter increment
+ for the total number of contexts was placed with the extensions
+ increment instead of in the enclosing loop. This function should
+ now generate correct context counts. (closes issue AST-970)
+ Reported-by: John Bigelow
+
+2012-09-05 13:13 +0000 [r372268] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_voicemail.c: Fix memory leaks in app_voicemail when
+ using IMAP storage or realtime config This patch fixes two memory
+ leaks: 1. When find_user is called with NULL as its first
+ parameter, the voicemail user returned is allocated on the heap.
+ The inboxcount2 function uses find_user in such a fashion when
+ counting new messages, and fails to free the resulting voicemail
+ user object. 2. When populate_defaults is called on a voicemail
+ user, it wipes whatever flags have been set on the object by
+ copying over the global flags object. If the VM_ALLOCED flag was
+ ste on the voicemail user prior to doing so, that flag is
+ removed. This leaks the voicemail user when free_user is later
+ called. (closes issue ASTERISK-19155) Reported by: Filip Jenicek
+ patches: asterisk.patch2 uploaded by Filip Jenicek (license 6277)
+ Patch slightly modified for this commit. Review:
+ https://reviewboard.asterisk.org/r/2096
+
+2012-09-05 07:35 +0000 [r372212-372239] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/dsp.c: dsp.c: Fix multiple issues when no-interdigit delay
+ is present, and fast DTMF 50ms/50ms Revert DTMF hit/miss detector
+ to original -r349249 method with some changes, remove
+ unnecessary; 1. reseting of hits=0, when no signal, only need to
+ set it once. 2. incrementing of hits, when the hit is the same as
+ the current hit. 3. setting of lasthit, when it's the same as
+ before. Change HITS_TO_BEGIN to 2, MISSES_TO_END to 3 & 3
+ spelling mistakes (closes issue ASTERISK-19610) alecdavis
+ (license 585) Reported by: Jean-Philippe Lord Tested by:
+ alecdavis Review: https://reviewboard.asterisk.org/r/2085/
+
+ * main/dsp.c: dsp.c: optimize goerztzel sample loops, in
+ dtmf_detect, mf_detect and tone_detect use a temporary short int
+ when repeatedly used to call goertzel_sample. alecdavis (license
+ 585) Reported by: alecdavis Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/2093/
+
+2012-09-05 03:45 +0000 [r372185] Michael L. Young <elgueromexicano@gmail.com>
+
+ * res/res_rtp_asterisk.c: Fix Incrementing Sequence Number For
+ Retransmitted DTMF End Packets In Asterisk 1.4+, a fix was put in
+ place to increment the sequence number for retransmitted DTMF end
+ packets. With the introduction of the RTP engine API in 1.8, the
+ sequence number was no longer being incremented. This patch fixes
+ this regression as well as cleans up a few lines that were not
+ doing anything. (closes issue ASTERISK-20295) Reported by: Nitesh
+ Bansal Tested by: Michael L. Young Patches:
+ 01_rtp_event_seq_num.patch uploaded by Nitesh Bansal (license
+ 6418) asterisk-20295-dtmf-fix-cleanup.diff uploaded by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2083/
+
+2012-09-05 02:16 +0000 [r372158] Matthew Jordan <mjordan@digium.com>
+
+ * cel/cel_pgsql.c: Fix memory leak when CEL is successfully written
+ to PostgreSQL database PQClear is not called when the result
+ object of a call to PQExec has a status of PGRES_COMMAND_OK.
+ Interestingly enough, the off nominal case was handled properly,
+ so this memory leak only occurred when CEL records were
+ successfully written. This patch properly clears the result in
+ the nominal code path. (closes issue ASTERISK-19991) Reported by:
+ Etienne Lessard Tested by: Etienne Lessard patches:
+ mem_leak_cel_pgsql.patch uploaded by Etienne Lessard (license
+ #6394)
+
+2012-08-30 20:51 +0000 [r372048-372089] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Prevent crash on shutdown due to refcount error
+ on queues container. When app_queue is unloaded, the queues
+ container has its refcount decremented, potentially to 0. Then
+ the taskprocessor responsible for handling device state changes
+ is unreferenced. If the taskprocessor happens to be just about to
+ run its task, then it will create and destroy an iterator on the
+ queues container. This can cause the refcount on the queues
+ container to increase to 1 and then back to 0. Going back to 0 a
+ second time results in double frees. This failure was seen
+ periodically in the testsuite when Asterisk would shut down.
+
+ * apps/app_queue.c: Help prevent ringing queue members from being
+ rung when ringinuse set to no. Queue member status would not
+ always get updated properly when the member was called, thus
+ resulting in the member getting multiple calls. With this change,
+ we update the member's status at the time of calling, and we also
+ check to make sure the member is still available to take the call
+ before placing an outbound call. (closes issue ASTERISK-16115)
+ reported by nik600 Patches: app_queue.c-svn-r370418.patch
+ uploaded by Italo Rossi (license #6409)
+
+2012-08-30 16:21 +0000 [r371961-372015] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_iax2.c: AST-2012-013: Resolve ACL rules being
+ ignored during calls by some IAX2 peers When an IAX2 call is made
+ using the credentials of a peer defined in a dynamic Asterisk
+ Realtime Architecture (ARA) backend, the ACL rules for that peer
+ are not applied to the call attempt. This allows for a remote
+ attacker who is aware of a peer's credentials to bypass the ACL
+ rules set for that peer. This patch ensures that the ACLs are
+ applied for all peers, regardless of their storage mechanism.
+ (closes issue ASTERISK-20186) Reported by: Alan Frisch Tested by:
+ mjordan, Alan Frisch
+
+ * main/manager.c, README-SERIOUSLY.bestpractices.txt: AST-2012-012:
+ Resolve AMI User Unauthorized Shell Access through ExternalIVR
+ The AMI Originate action can allow a remote user to specify
+ information that can be used to execute shell commands on the
+ system hosting Asterisk. This can result in an unwanted
+ escalation of permissions, as the Originate action, which
+ requires the "originate" class authorization, can be used to
+ perform actions that would typically require the "system" class
+ authorization. Previous attempts to prevent this permission
+ escalation (AST-2011-006, AST-2012-004) have sought to do so by
+ inspecting the names of applications and functions passed in with
+ the Originate action and, if those applications/functions matched
+ a predefined set of values, rejecting the command if the user
+ lacked the "system" class authorization. As noted by IBM X-Force
+ Research, the "ExternalIVR" application is not listed in the
+ predefined set of values. The solution for this particular
+ vulnerability is to include the "ExternalIVR" application in the
+ set of defined applications/functions that require "system" class
+ authorization. Unfortunately, the approach of inspecting fields
+ in the Originate action against known applications/functions has
+ a significant flaw. The predefined set of values can be bypassed
+ by creative use of the Originate action or by certain dialplan
+ configurations, which is beyond the ability of Asterisk to
+ analyze at run-time. Attempting to work around these scenarios
+ would result in severely restricting the applications or
+ functions and prevent their usage for legitimate means. As such,
+ any additional security vulnerabilities, where an
+ application/function that would normally require the "system"
+ class authorization can be executed by users with the "originate"
+ class authorization, will not be addressed. Instead, the
+ README-SERIOUSLY.bestpractices.txt file has been updated to
+ reflect that the AMI Originate action can result in commands
+ requiring the "system" class authorization to be executed. Proper
+ system configuration can limit the impact of such scenarios.
+ (closes issue ASTERISK-20132) Reported by: Zubair Ashraf of IBM
+ X-Force Research
+
+ * doc/CODING-GUIDELINES (added): Restore CODING-GUIDELINES to doc
+ folder In r294740, the CODING-GUIDELINES was removed from the doc
+ folder in favor of the content on the Asterisk wiki. Some folks
+ still look in the doc folder initially for coding guideline
+ suggestions; as such, this patch adds a CODING-GUIDELINES file
+ back into the doc folder. The content of the file merely points
+ to the correct page on the Asterisk wiki where the coding
+ guidelines currently live. (closes issue ASTERISK-20279) Reported
+ by: Andrew Latham Patches: CODING-GUIDELINES.diff uploaded by
+ Andrew Latham (license 5985)
+
+2012-08-29 20:42 +0000 [r371919] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_meetme.c: app_meetme: Adding test events for following
+ activity in MeetMe.
+
+2012-08-29 19:38 +0000 [r371860-371888] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Initialize file descriptors for dummy channels to
+ -1. Dummy channels usually aren't read from, but functions like
+ SHELL and CURL use autoservice on the channel. (closes issue
+ ASTERISK-20283) Reported by: Gareth Palmer Patches:
+ svn-371580.patch (license #5169) patch uploaded by Gareth Palmer
+ (modified)
+
+ * apps/app_dial.c: Fix hangup cause passthrough regression. The
+ v1.8 -r369258 change to fix the F and F(x) action logic
+ introduced a regression in passing the hangup cause from the
+ called channel to the caller channel. (closes issue
+ ASTERISK-20287) Reported by: Konstantin Suvorov Patches:
+ app_dial_hangupcause.patch (license #6421) patch uploaded by
+ Konstantin Suvorov (modified) Tested by: rmudgett
+
+2012-08-29 16:59 +0000 [r371824] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c: chan_sip: Send 408 on retransmit timeout
+ instead of 603 (closes issue ASTERISK-20124) Reported by: Walter
+ Doekes
+
+2012-08-27 21:47 +0000 [r371747-371787] Mark Michelson <mmichelson@digium.com>
+
+ * configs/agents.conf.sample: Fix misleading documentation in
+ agents.conf.sample regarding ackcall usage. The documentation
+ made it sound as if the DTMF acknowledgment was needed at the
+ time the agent logs in, rather than when the agent is called.
+ This is likely a relic from the days when there were multiple
+ ways of logging in agents. (closes issue AST-962) reported by
+ Steve Pitts
+
+ * main/manager.c: Fix incorrect documentation of the MailboxStatus
+ manager command. The "Waiting" field was misdocumented as
+ reporting the number of messages waiting. In reality, it simply
+ indicated the presence or absence of waiting messages. (closes
+ issue AST-975) reported by John Bigelow
+
+ * configs/queues.conf.sample: Fix incorrectly documented option in
+ queues.conf sharedlastcall defaults to "no" not "yes" (closes
+ issue AST-979) reported by Steve Pitts
+
+2012-08-27 16:40 +0000 [r371718] dlee <dlee@localhost>:
+
+ * main/lock.c: Fixes ast_rwlock_timed[rd|wr]lock for BSD and
+ variants. The original implementations simply wrap pthread
+ functions, which take absolute time as an argument. The spinlock
+ version for systems without those functions treated the argument
+ as a delta. This patch fixes the spinlock version to be
+ consistent with the pthread version. (closes issue
+ ASTERISK-20240) Reported by: Egor Gorlin Patches: lock.c.patch
+ uploaded by Egor Gorlin (license 6416)
+
+2012-08-27 13:43 +0000 [r371690] Kinsey Moore <kmoore@digium.com>
+
+ * main/utils.c: Implement workaround for BETTER_BACKTRACES crash
+ When compiling with BETTER_BACKTRACES enabled, Asterisk will
+ sometimes crash when "core show locks" is run. This happens
+ regularly in the testsuite since several tests run "core show
+ locks" to help with debugging. This seems to be a fault with
+ libraries on certain operating systems (notably CentOS 6.2/6.3)
+ running on virtual machines and utilizing gcc 4.4.6. (closes
+ issue ASTERISK-20090)
+
+2012-08-26 23:03 +0000 [r371662] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/dsp.c: mf_detect: incorrectly used DTMF_GSIZE instead of
+ MF_GSIZE
+
+2012-08-21 20:35 +0000 [r371590] Mark Michelson <mmichelson@digium.com>
+
+ * main/utils.c, apps/app_queue.c, pbx/pbx_config.c,
+ res/res_jabber.c, apps/app_stack.c, channels/chan_oss.c,
+ res/res_config_sqlite.c, cdr/cdr_tds.c, main/xmldoc.c,
+ apps/app_dial.c, channels/chan_dahdi.c, channels/chan_sip.c,
+ funcs/func_odbc.c, main/file.c: Fix misuses of asprintf
+ throughout the code. This fixes three main issues * Change
+ asprintf() uses to ast_asprintf() so that it pairs properly with
+ ast_free() and no longer causes MALLOC_DEBUG to freak out. * When
+ ast_asprintf() fails, set the pointer NULL if it will be
+ referenced later. * Fix some memory leaks that were spotted while
+ taking care of the first two points. (Closes issue
+ ASTERISK-20135) reported by Richard Mudgett Review:
+ https://reviewboard.asterisk.org/r/2071
+
+2012-08-20 15:25 +0000 [r371544] Kinsey Moore <kmoore@digium.com>
+
+ * main/udptl.c: Ignore recovered zero-length secondary UDPTL
+ packets In some cases, recovering lost packets using the
+ secondary packet recovery mechanism with UDPTL/T.38 can result in
+ the recovery of zero-length packets. These must be ignored or the
+ frame generated from them can cause segfaults and allocation
+ failures. (closes issue ASTERISK-19762) (closes issue
+ ASTERISK-19373) Reported-by: Benjamin (bulkorok) Reported-by: Rob
+ Gagnon (rgagnon)
+
+2012-08-17 18:51 +0000 [r371469] Matthew Jordan <mjordan@digium.com>
+
+ * main/xmldoc.c: Fix memory leak in XML documentation When
+ formatting documentation fields, the XML documentation parser
+ calls xmldoc_get_formatted. This function allocates a string
+ buffer at the beginning of its routine. Unfortunately, on certain
+ code paths, it also calls xmldoc_string_cleanup, which assumes
+ that it will create the string buffer. The previously allocated
+ string buffer is then leaked by the xmldoc_string_cleanup
+ routine. Now: we don't do that. (closes issue AST-932) Reported
+ by: Alexander Homig
+
+2012-08-17 15:49 +0000 [r371393-371436] Kinsey Moore <kmoore@digium.com>
+
+ * main/loader.c: Add instrumentation to subsystem reloads When
+ Asterisk is built with TEST_FRAMEWORK defined, Asterisk will now
+ generate TestEvent AMI events on subsystem reloads such as cdr,
+ dnsmgr, extconfig, etc. (issue PQ-1126)
+
+ * main/loader.c: Add module reload instrumentation for
+ TEST_FRAMEWORK This adds AMI events for module reloads when
+ Asterisk is built with TEST_FRAMEWORK enabled and corrects
+ generation of the module load AMI event. (issue PQ-1126)
+
+2012-08-16 22:30 +0000 [r371392] Terry Wilson <twilson@digium.com>
+
+ * main/config.c: Handle integer over/under-flow in ast_parse_args
+ The strtol family of functions will return *_MIN/*_MAX on
+ overflow. To detect when an overflow has happened, errno must be
+ set to 0 before calling the function, then checked afterward.
+ (closes issue ASTERISK-20120) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/2073/
+
+2012-08-16 18:57 +0000 [r371337-371357] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c: chan_sip: Use pvt outgoing_call variable to
+ set Remote-Party-ID Header Previously the pvt SIP_OUTGOING flag
+ was used instead, which will frequently flip during reinvites.
+ (closes issue AST-897) Reported by: Thomas Arimont
+
+ * channels/chan_sip.c: chan_sip: Trigger reinvite if the SDP answer
+ is included in the SIP ACK Under certain conditions, a SIP
+ transaction involving directmedia wouldn't trigger a re-invite
+ because the SDP answer was included in an ACK instead of in a
+ message that we would have triggered the invite with. This patch
+ just queues a source change control frame if the dialog is using
+ directmedia when we find sdp for an ACK. (closes issue AST-913)
+ Reported by: Thomas Arimont
+
+2012-08-15 23:10 +0000 [r371306] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix bug where final queue member would not be
+ removed from memory. If a static queue had realtime members, then
+ there could be a potential for those realtime members not to be
+ properly deleted from memory. If the queue's members were loaded
+ from realtime and then all the members were deleted from the
+ backend, then the queue would still think these members existed.
+ The reason was that there was a short- circuit in code such that
+ if there were no members found in the backend, then the queue
+ would not be updated to reflect this. Note that this only
+ affected static queues with realtime members. Realtime queues
+ with realtime members were unaffected by this issue. (closes
+ issue ASTERISK-19793) reported by Marcus Haas
+
+2012-08-15 20:14 +0000 [r371270] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c: Avoid unconditional NULLing of mwipvt on
+ relatedpeer on SIP dialog destruction The other instance of this
+ bug was fixed by jcolp/file in r121496. If we are destroying a
+ dialog only set the MWI dialog pointer on the related peer to
+ NULL if it is the dialog currently being destroyed. (closes issue
+ ASTERISK-20119) Patch-by: Misha Vodsedalek
+
+2012-08-13 20:00 +0000 [r371201] Kinsey Moore <kmoore@digium.com>
+
+ * main/loader.c, apps/app_meetme.c: Add test instrumentation This
+ adds test instrumentation for loading and unloading of modules
+ and for certain actions in MeetMe to be used in the testsuite or
+ any other consumer of AMI events. These will only be generated
+ when Asterisk is built with TEST_FRAMEWORK enabled. (issue
+ PQ-1131) (issue PQ-1133)
+
+2012-08-13 19:49 +0000 [r371198] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix problem where incorrect pointer was
+ checked for nullity.
+
+2012-08-10 21:21 +0000 [r371141] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix a couple of documentation problems in
+ app_queue.c * The RemoveQueueMember app made mention of options
+ that could be passed in, but no options are supported. I have
+ removed the listing of options from the documentation. * The
+ RQMSTATUS variable did not list "NOTDYNAMIC" as a possible value
+ that could be set. (closes issue AST-949) reported by Steve Pitts
+ (closes issue AST-954) reported by Steve Pitts
+
+2012-08-10 16:40 +0000 [r371060-371089] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c: remove ALREADYGONE flag on ooh323 call data
+ by ooh323_indicate (CONGESTION/BUSY) due to call hasn't gone
+ there really. This indication arrive from asterisk core not h.323
+ stack (closes issue ASTERISK-19308) Reported by: Dmitry Melekhov
+ Patches: ASTERISK-19308.patch
+
+ * addons/ooh323c/src/ooGkClient.c: Send re-register packets by GRQ
+ (gatekeeper request) interval (close issue ASTERISK-20094)
+ Patches: ASTERISK-20094-2.patch
+
+2012-08-09 18:58 +0000 [r371012] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c, channels/sig_ss7.c, channels/chan_dahdi.c,
+ configure, include/asterisk/autoconfig.h.in, configure.ac: Use
+ better libss7 detection test and move libpri compile test.
+
+2012-08-09 18:58 +0000 [r370988-371011] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooGkClient.c: Fix to resend GRQ/RRQ if RRJ
+ (registration reject) is received (close issue ASTERISK-20094)
+ Patches: ASTERISK-20094.patch
+
+ * addons/ooh323c/src/ooh323ep.c: change opening h323 logfile with
+ append mode instead of overwrite
+
+2012-08-09 17:39 +0000 [r370985] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_meetme.c: Correct documentation for the MeetMe x flag
+ The documentation for the x flag for MeetMe incorrectly described
+ its function as closing down the conference when the last marked
+ user left. It actually causes the users with that flag to leave
+ the conference when the last marked user exits. The functionality
+ of this flag is not changing.
+
+2012-08-08 22:40 +0000 [r370952] Michael L. Young <elgueromexicano@gmail.com>
+
+ * apps/app_chanspy.c: Fix Not Unreferencing A Spied Channel When a
+ channel hangs up while being spied upon and the option to exit
+ the ChanSpy application when the spied on channel hangs up is
+ set, ast_autochan_destroy is not being called and therefore a
+ reference to the spied upon channel is not removed. The symptom
+ being reported was that when using func_group in the dialplan and
+ calling "group show channels" at the cli, the spied upon channel
+ was still being shown while "core show channels" showed that the
+ channel was not up. This patch calls ast_autochan_destroy when a
+ spied upon channel hangs up and the option to exit the ChanSpy
+ application is set, removing the reference to the channel
+ allowing the count for the group that the spied channel was part
+ of to be decremented. (closes issue ASTERISK-17515) Reported by:
+ Arkadiusz Malka Tested by: Alexandr Gordeev, Michael L. Young
+ Patches: asterisk-17515-destroy-autochan.diff uploaded by Michael
+ L. Young (license 5026)
+
+2012-08-08 20:28 +0000 [r370923] Kinsey Moore <kmoore@digium.com>
+
+ * main/channel.c: Do not define a cause that doesn't actually exist
+ AST_CAUSE_NOTDEFINED is a placeholder for usage when there is no
+ cause information. As such, it should not be defined and
+ translatable as a cause.
+
+2012-08-08 19:58 +0000 [r370900] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h: Fix the analog dial *0 flash-hook of
+ bridged peer feature. The flash-hook the bridged peer feature now
+ correctly determines if the bridged peer is another chan_dahdi
+ channel, that it is an analog channel, and that it has the
+ correct signaling for an FXO port. It now also flash-hooks the
+ correct channel.
+
+2012-08-07 19:19 +0000 [r370856] Kinsey Moore <kmoore@digium.com>
+
+ * main/channel.c: Add missing AST_CAUSE_* -> text translations
+
+2012-08-06 15:00 +0000 [r370797] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Improve debug message for temporary outbound
+ proxies. Thanks to Paul Belanger for pointing this out.
+
+2012-08-03 21:43 +0000 [r370769-370771] Mark Michelson <mmichelson@digium.com>
+
+ * channels/sip/config_parser.c: Seriously? Another compilation
+ error fixed. Somebody beat me.
+
+ * channels/chan_sip.c: Remove unused variable.
+
+ * channels/sip/config_parser.c, channels/sip/include/sip.h,
+ channels/chan_sip.c: Fix error in the "IPorHost" section of a SIP
+ dialstring. This is based on the review request posted by Walter
+ Doekes (referenced lower in the commit message) The main fix here
+ is to treat the IPorHost portion of the dial string as a
+ temporary outbound proxy. This ensures requests get sent to the
+ proper location. Due to the age of the request, some parts were
+ no longer relevant. For instance, the request moved outbound
+ proxy parsing code into a single method. This is done in a
+ previous commit, so it was not necessary to do again. Also, the
+ review request fixed some errors with regards to request routing
+ for CANCEL and ACK requests. This has also been fixed in more
+ recent commits. (closes issue ASTERISK-19677) reported by Walter
+ Doekes Review https://reviewboard.asterisk.org/r/1859
+
+2012-08-01 02:25 +0000 [r370697] Kinsey Moore <kmoore@digium.com>
+
+ * utils/extconf.c: Revert alloca changes for utils These changes
+ were a tad overzealous in the utils directory. Unfortunately,
+ these don't compile with a "make".
+
+2012-07-31 20:54 +0000 [r370666] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_sip.c: Schedule pokes of registered SIP peers
+ within a given timespan after SIP reload With a large number of
+ SIP peers registered, performing a SIP reload causes a flood of
+ SIP OPTIONS request packets. These are immediately sent out, and,
+ as responses come back, can cause peers to be flagged as 'lagged'
+ due to handling of the many response messages. This fix prevents
+ this "packet storm" and schedules the pokes for a random time.
+ That time varies between 1 ms and the peer's qualify time, or, if
+ the qualify time is unknown, the global qualifyfreq setting. The
+ committed patch has some very small modifications to the patch
+ schmidts wrote for the review. (closes issue ASTERISK-19154)
+ Reported by: Nicolo Mazzon patches: issue19154.patch license
+ #6034 uploaded by schmidts Review:
+ https://reviewboard.asterisk.org/r/1652
+
+2012-07-31 19:31 +0000 [r370642] Kinsey Moore <kmoore@digium.com>
+
+ * main/utils.c, funcs/func_logic.c, channels/chan_gtalk.c,
+ cdr/cdr_pgsql.c, channels/chan_iax2.c, res/res_jabber.c,
+ main/config.c, main/channel.c, res/ael/pval.c,
+ apps/app_osplookup.c, main/manager.c, pbx/pbx_spool.c,
+ main/strcompat.c, apps/app_minivm.c, main/features.c,
+ res/res_agi.c, main/http.c, main/logger.c, pbx/pbx_ael.c,
+ main/app.c, channels/chan_alsa.c, pbx/pbx_realtime.c,
+ addons/chan_mobile.c, apps/app_while.c, include/asterisk/utils.h,
+ main/pbx.c, res/res_config_pgsql.c, channels/chan_sip.c,
+ apps/app_festival.c, pbx/pbx_lua.c, funcs/func_cut.c,
+ tests/test_linkedlists.c, apps/app_getcpeid.c,
+ funcs/func_global.c, channels/chan_jingle.c, main/tcptls.c,
+ funcs/func_channel.c, apps/app_directed_pickup.c,
+ main/callerid.c, main/file.c, apps/app_macro.c, main/astmm.c,
+ apps/app_sms.c, main/event.c, pbx/pbx_dundi.c,
+ include/asterisk/strings.h, utils/extconf.c,
+ apps/app_mixmonitor.c, main/asterisk.c, main/dsp.c,
+ addons/res_config_mysql.c, apps/app_voicemail.c,
+ addons/app_mysql.c, apps/app_meetme.c, apps/app_dictate.c,
+ main/say.c, main/threadstorage.c, funcs/func_strings.c: Clean up
+ and ensure proper usage of alloca() This replaces all calls to
+ alloca() with ast_alloca() which calls gcc's __builtin_alloca()
+ to avoid BSD semantics and removes all NULL checks on memory
+ allocated via ast_alloca() and ast_strdupa(). (closes issue
+ ASTERISK-20125) Review: https://reviewboard.asterisk.org/r/2032/
+ Patch-by: Walter Doekes (wdoekes)
+
+2012-07-31 15:26 +0000 [r370618] Mark Michelson <mmichelson@digium.com>
+
+ * configs/sip.conf.sample, channels/sip/include/sip.h,
+ channels/chan_sip.c: Help mitigate potential reinvite glare
+ scenarios. When Asterisk servers are set up back-to-back, and
+ direct media is to be used betweeen endpoints, it is fairly
+ common for the two Asterisk servers to send direct media
+ reinvites to each other simultaneously. This results in 491s and
+ ACKs being exchanged between the servers. While the media
+ eventually gets set up properly, the problem is that there can be
+ a noticeable delay for the streams to stabilize. This patch adds
+ a new directmedia option called "outgoing". With this set, an
+ immediate direct media reinvite will only be sent if the call
+ direction is outgoing. For incoming dialogs, an immediate direct
+ media reinvite will not be sent, but further "reactionary" direct
+ media reinvites may be sent. For those who are having some deja
+ vu, that's because this patch was originally committed to trunk
+ since there is a new configuration option added. After seeing a
+ bug report about audio being slow to set up on SIP calls, it
+ became apparent that this patch would be the best solution for
+ resolving the issue. The patch is unintrusive and will have no
+ effect unless the option is explicitly enabled. (closes issue
+ AST-896) reported by Thomas Arimont (closes issue ASTERISK-19857)
+ reported by Matt Jordan
+
+2012-09-13 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.16.0 Released.
+
+2012-09-11 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.16.0-rc2 Released.
+
+ * AST-2012-013: Resolve ACL rules being ignored during calls by some
+ IAX2 peers
+
+ * AST-2012-012: Resolve AMI User Unauthorized Shell Access through
+ ExternalIVR
+
+ * r371860: Fix hangup cause passthrough regression.
+
+ The v1.8 -r369258 change to fix the F and F(x) action logic
+ introduced a regression in passing the hangup cause from the called
+ channel to the caller channel.
+
+ (closes issue ASTERISK-20287)
+ Reported by: Konstantin Suvorov
+ Patches:
+ app_dial_hangupcause.patch (license #6421) patch uploaded by
+ Konstantin Suvorov (modified)
+ Tested by: rmudgett
+
+ * r372709: Only re-create an SRTP session when needed; respond with
+ correct crypto policy
+
+ In r356604, SRTP handling was fixed to accomodate multiple crypto
+ keys in an SDP offer and the ability to re-create an SRTP session
+ when the crypto keys changed. In certain circumstances - most
+ notably when a phone is put on hold after having been bridged for a
+ significant amount of time - the act of re-creating the SRTP session
+ causes problems for certain models of phones. The patch committed in
+ r356604 always re-created the SRTP session regardless of whether or
+ not the cryptographic keys changed. Since this is technically
+ not necessary, this patch modifies the behavior to only re-create the
+ SRTP session if Asterisk detects that the remote key has changed.
+ This allows models of phones that do not handle the SRTP session
+ changing to continue to work, while also providing the behavior
+ needed for those phones that do re-negotiate cryptographic keys.
+
+ In addition, in Asterisk 1.8 only, it was found that phones that
+ offer AES_CM_128_HMAC_SHA1_32 will end up with no audio if the phone
+ is the initiator of the call. The phone will send an INVITE request
+ specifying that AES_CM_128_HMAC_SHA1_32 be used for the cryptographic
+ policy; Asterisk will set its policy to that value. Unfortunately,
+ when the call is Answered and a 200 OK is sent back to the UA, the
+ policy sent in the response's SDP will be the hard coded value
+ AES_CM_128_HMAC_SHA1_80. This potentially results in Asterisk using
+ the INVITE request's policy of AES_CM_128_HMAC_SHA1_32, while the
+ phone uses Asterisk's response of AES_CM_128_HMAC_SHA1_80. Hilarity
+ ensues as both endpoints think the other is crazy.
+
+ This patch fixes that by caching the policy from the request and
+ responding with it. Note that this is not a problem in Asterisk 10
+ and later, as the ability to configure the policy was added in that
+ version.
+
+ (issue ASTERISK-20194)
+ Reported by: Nicolo Mazzon
+ Tested by: Nicolo Mazzon
+
+ Review: https://reviewboard.asterisk.org/r/2099
+
+ * r372840: Fix bad channel application data reference.
+
+ When channels get bridged due to an AMI bridge action
+ or a DTMF attended transfer, the two channels that
+ get bridged have their application data pointing to
+ the other channel's name. This means that if one channel
+ is hung up but the other moves on, it means that the
+ channel that moves on will have its application data
+ pointing at freed memory.
+
+ (issue ASTERISK-20335)
+
+2012-07-31 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.16.0-rc1 Released.
+
+2012-07-30 16:47 +0000 [r370563] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: Release B channel allocation on error path
+ in chan_misdn.
+
+2012-07-25 21:00 +0000 [r370494] Jonathan Rose <jrose@digium.com>
+
+ * res/res_agi.c: res_agi: Add message indicating need for \n
+ character in verbose message The while loop responsible for
+ reading AGI messages from a fastAGI service can end up looping
+ indefinitely when an AGI script fails to indicate the end of a
+ message with a \n character. This patch adds an indication that
+ we are expecting a \n character to end the message to make it
+ more clear to users that this is necessary if they are receiving
+ this warning over and over. (issue ASTERISK-20061) Reported by:
+ Eike Kuiper
+
+2012-07-24 16:53 +0000 [r370429] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/frame.c: Rewrite a comment that didn't adequately explain
+ the code it was documenting.
+
+2012-07-24 16:49 +0000 [r370428] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_oss.c: chan_oss: fix "sample rate" error message
+
+2012-07-23 21:09 +0000 [r370360-370383] Kevin P. Fleming <kpfleming@digium.com>
+
+ * funcs/func_shell.c: Improve documentation for the SHELL()
+ dialplan function.
+
+ * main/channel.c: Free any datastores attached to dummy channels.
+ Revision 370205 added the use of a datastore attached to a dummy
+ channel to resolve a memory leak, but
+ ast_dummy_channel_destructor() in this branch did not free
+ datastores, resulting in a continued (but slightly smaller)
+ memory leak. This patch backports the change to free said
+ datastores from the Asterisk trunk. (related to issue AST-916)
+
+2012-07-19 22:07 +0000 [r370275] Richard Mudgett <rmudgett@digium.com>
+
+ * main/cel.c: Fix compiler warnings. gcc (GCC) 4.2.4 has problems
+ casting away constness.
+
+2012-07-19 22:00 +0000 [r370252-370273] Matthew Jordan <mjordan@digium.com>
+
+ * main/cel.c: Fix compilation error when MALLOC_DEBUG is enabled To
+ fix a memory leak in CEL, a channel datastore was introduced
+ whose destruction function pointer was pointed to the ast_free
+ macro. Without MALLOC_DEBUG enabled this compiles as fine, as
+ ast_free is defined as free. With MALLOC_DEBUG enabled, however,
+ ast_free takes on a definition from a different place then
+ utils.h, and became undefined. This patch resolves this by using
+ a reference to ast_free_ptr. When MALLOC_DEBUG is enabled, this
+ calls ast_free; when MALLOC_DEBUG is not enabled, this is defined
+ to be ast_free, which is defined to be free. (issue AST-916)
+ Reported by: Thomas Arimont
+
+ * res/res_rtp_asterisk.c: Handle extremely out of order RFC 2833
+ DTMF The current implementation of RFC 2833 DTMF handling in
+ res_rtp_asterisk will, if a packet arrives out of order, drop the
+ packet. This is to prevent duplicate ton generation in the
+ Asterisk core. Since the RTP layer does not buffer data itself,
+ this is the only option the RTP layer currently has for handling
+ packets that arrive out of order. For the most part, this doesn't
+ matter. For a particular digit, so long as a BEGIN packet arrives
+ before the first END packet, the digit will be produced. If
+ subsequent BEGIN packets arrive interleaved with the ENDs, they
+ will be dropped; likewise, if the BEGIN or END packets themselves
+ are out of order, those packets are dropped but sufficient
+ information is conveyed to the Asterisk core to produce the
+ appropriate digit. For certain sequences of DTMF packets - most
+ notably when, for a particular digit, an END packet arrives
+ before any BEGIN packet for that digit - this is a real problem.
+ When an END arrives before any BEGINs, the END packet is dropped
+ - but at the same time, it causes subsequent BEGIN packets for
+ that digit to be ignored. When the next in order END packet
+ arrives, it too is dropped - Asterisk believes that there was no
+ initial BEGIN. The solution this patch provides is to trust the
+ END packet to convey the information needed for the Asterisk core
+ to produce the DTMF digit. If we receive an END packet, and it: *
+ Has a timestamp greater then the last timestamp received from an
+ END packet * Does not have the same sequence number as the last
+ received sequence number (and is thus not an END packet
+ retransmission) Then we send the END frame up to the Asterisk
+ core. It contains enough DTMF information for Asterisk to produce
+ the digit. On the other hand, if we receive a BEGIN or
+ continuation packet that occurs with a timestamp equal to or less
+ then the last END timestamp, then we've received something out of
+ order - but we already have received enough information to
+ produce the digit. These packets are dropped. Much thanks goes to
+ Olle Johansson (oej) for providing the idea for this solution.
+ Review: https://reviewboard.asterisk.org/r/2033/ (issue
+ ASTERISK-18404) Reporter: Stephane Chazelas Tested by: Matt
+ Jordan
+
+2012-07-18 19:12 +0000 [r370183-370205] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/cel.c: Resolve severe memory leak in CEL logging modules. A
+ customer reported a significant memory leak using Asterisk 1.8.
+ They have tracked it down to
+ ast_cel_fabricate_channel_from_event() in main/cel.c, which is
+ called by both in-tree CEL logging modules (cel_custom.c and
+ cel_sqlite3_custom.c) for each and every CEL event that they log.
+ The cause was an incorrect assumption about how data attached to
+ an ast_channel would be handled when the channel is destroyed;
+ the data is now stored in a datastore attached to the channel,
+ which is destroyed along with the channel at the proper time.
+ (closes issue AST-916) Review:
+ https://reviewboard.asterisk.org/r/2053/
+
+ * apps/app_macro.c, channels/chan_iax2.c, apps/app_mixmonitor.c,
+ apps/app_stack.c, funcs/func_global.c, res/res_odbc.c,
+ main/channel.c, addons/app_mysql.c, main/pbx.c,
+ funcs/func_curl.c, main/ccss.c, funcs/func_odbc.c,
+ funcs/func_lock.c: Ensure that all ast_datastore_info structures
+ are 'const'. While addressing a bug, I came across a instance of
+ 'struct ast_datastore_info' that was not declared 'const'. Since
+ the API already expects them to be 'const', this patch changes
+ the declarations of all existing instances that were not already
+ declared that way.
+
+2012-07-16 19:50 +0000 [r370131] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * channels/chan_sip.c: Code cleanup and bugfix in chan_sip
+ outboundproxy parsing. The bug was clearing the global
+ outboundproxy when a peer-specific outboundproxy was bad. The
+ cleanup reduces duplicate code. Review:
+ https://reviewboard.asterisk.org/r/2034/ Reviewed by: Mark
+ Michelson
+
+2012-07-16 13:44 +0000 [r370081] Kinsey Moore <kmoore@digium.com>
+
+ * UPGRADE.txt, CHANGES: Add comments about the BUILD_NATIVE change
+ This is a significant change and mention of it should have gone
+ into UPGRADE.txt and CHANGES.
+
+2012-07-12 20:15 +0000 [r370017] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c: Add missing
+ ast_hangup() calls on some analog exception paths. Make starting
+ analog_ss_thread() or __analog_ss_thread() failure paths hangup
+ the channel.
+
+2012-07-12 20:05 +0000 [r369993-370014] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c: Include Expires header for SIP PUBLISH
+ requests RFC3903 requres SIP PUBLISH requests to have Expires
+ headers, so add them. Review:
+ https://reviewboard.asterisk.org/r/2003/ Patch-by: gareth
+
+ * channels/chan_sip.c: Prevent double uri_escaping in chan_sip when
+ pedantic is enabled If pedantic mode is enabled, outbound invites
+ will have double-escaped contacts. This avoids setting an
+ already-escaped string into a field where it is expected to be
+ unescaped. (closes issue ASTERISK-20023) Reported-by: Walter
+ Doekes
+
+2012-07-12 14:23 +0000 [r369970] Michael L. Young <elgueromexicano@gmail.com>
+
+ * funcs/func_math.c: Correct Documentation For DEC Function The
+ documentation for DEC in func_math.c was incorrect. Looks like a
+ copy and paste error. (Closes issue ASTERISK-20095) Reported by:
+ Billy Chia Tested by: Michael L. Young Patches: func_math.patch
+ uploaded by Billy Chia (license 6381)
+
+2012-07-11 17:08 +0000 [r369937] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * funcs/func_realtime.c: Allow the REALTIME() function to report
+ errors back to the caller. Also, do more error checking on the
+ arguments specified to the REALTIME() function and clarify the
+ documentation. While I was editing the file, a few coding
+ guidelines fixups, as well. Review:
+ https://reviewboard.asterisk.org/r/2031/
+
+2012-07-30 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.15.0 Released.
+
+2012-07-11 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.15.0-rc1 Released.
+
+2012-07-10 13:33 +0000 [r369869] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_stack.c, main/pbx.c: Improve Goto and GotoIf related
+ documentation Correct documentation on labeliftrue and
+ labeliffalse parameters of GotoIf() and update several other
+ locations that use the same syntax. (closes issue ASTERISK-20007)
+ Patch-by: Leif Madsen Reported-by: WIMPy
+
+2012-07-09 17:05 +0000 [r369818] Jason Parker <jparker@digium.com>
+
+ * configs/sip_notify.conf.sample: Add Digium phones context to
+ sip_notify sample config. This makes it so that they can be
+ reconfigured remotely. (closes issue ASTERISK-19910)
+
+2012-07-09 14:38 +0000 [r369792] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c: chan_sip: Fix small behavioral change
+ accidentally introduced in r369750 When removing the warning for
+ AST_CONTROL_FLASH from sip_indicate, I also inadvertently changed
+ the return value, which would likely make the indication not be
+ sent in audio. This fixes that while still removing the warning
+ message.
+
+2012-07-06 20:54 +0000 [r369750] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c: chan_sip: Add case for FLASH control frames
+ so that we don't display a warning. chan_sip channels can receive
+ flash control frames when connected to analog phones and possibly
+ for other reasons. There really isn't a reason to warn when these
+ frames are received, we can safely ignore them. Patches:
+ dahdi_sip_flash.diff uploaded by Jonathan Rose (license 6182)
+
+2012-07-06 18:40 +0000 [r369708-369731] Mark Michelson <mmichelson@digium.com>
+
+ * main/tcptls.c: Remove a superfluous and dangerous freeing of an
+ SSL_CTX. The problem here is that multiple server sessions share
+ a SSL_CTX. When one session ended, the SSL_CTX would be freed and
+ set NULL, leaving the other sessions unable to function. The code
+ being removed is superfluous because the SSL_CTX structures for
+ servers will be properly freed when ast_ssl_teardown is called.
+ (closes issue ASTERISK-20074) Reported by Trevor Helmsley
+ Patches: ASTERISK-20074.diff uploaded by Mark Michelson (license
+ #5049) Testers: Trevor Helmsley
+
+ * main/bridging.c: Fix bridging thread leak. The bridge thread was
+ exiting but was never being reaped using pthread_join(). This has
+ been fixed now by calling pthread_join() in ast_bridge_destroy().
+ (closes issue ASTERISK-19834) Reported by Marcus Hunger Review:
+ https://reviewboard.asterisk.org/r/2012
+
+2012-07-05 19:01 +0000 [r369652] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_voicemail.c: AST-2012-011: Resolve heap corruption issue
+ with voicemail The heard and deleted arrays in the voicemail
+ state structure were not handled properly following the memory
+ leak fix in r354890 and a fix for an invalid free in r356797.
+ This could result in accessing and writing into freed memory. The
+ allocation for these arrays has been reworked to avoid the
+ possibility of invalid frees, access of freed memory, and crashes
+ that were occurring as a result of this. Locking around accesses
+ and modifications of the voicemail state structure members
+ dh_arraysize, heard, and deleted has been added to prevent
+ simultaneous modification and access when IMAP storage is in use.
+ If IMAP storage is not in use, this locking is not compiled in.
+ Review: https://reviewboard.asterisk.org/r/1994/ (closes issue
+ ASTERISK-19923) Reported by: Dan Delaney Tested by: Dan Delaney,
+ Julian Yap Patches: vm_alloc_fix.diff uploaded by kmoore (license
+ 6273)
+
+2012-07-05 17:01 +0000 [r369626] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_sip.c: Do not send a BYE when a provisional
+ response arrives during a re-INVITE Commits r369557 and r369579
+ were done to improve handling of re-INVITEs when the UA that was
+ supposed to receive the re-INVITE fails to respond. A limitation
+ of those patches occurred when a UA sent a provisional response
+ to the re-INVITE. This triggered a sending of a BYE in
+ check_pending. This patch tweaks the handling of the re-INVITE
+ such that a BYE is not sent in response to those messages. (issue
+ ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies
+ patches: (reinvite_tweak.diff license #5012 by Steve Davies)
+
+2012-07-03 16:58 +0000 [r369557-369579] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: More improvements to re-INVITEs timing out
+ after a provisional response There is no need to call
+ check_pendings() on a final response to an INVITE when destroying
+ the scheduler entry as it will be done later during normal
+ processing. (issue ASTERISK-19992)
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: Better handle
+ re-INVITEs with provisional but no final repsonses A previous
+ attempt at fixing this issue had negative side effects related to
+ attended transfers which this patch should resolve. Many thanks
+ to Steve Davies for all of the good suggestions and testing.
+ (closes issue ASTERISK-19992) Reported by: Steve Davies Tested
+ by: Steve Davies, Terry Wilson Review:
+ https://reviewboard.asterisk.org/r/2009/
+
+2012-06-29 16:52 +0000 [r369471-369490] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: With some configurations a transport is not
+ actually specified so assume UDP in these cases.
+
+ * channels/chan_sip.c: Make the address family filter specific to
+ the transport. (closes issue ASTERISK-16618) Reported by: Leif
+ Madsen Review: https://reviewboard.asterisk.org/r/1667/
+
+2012-06-27 20:58 +0000 [r369436] Terry Wilson <twilson@digium.com>
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: AST-2012-010:
+ Clean up after a reinvite that never gets a final response The
+ basic problem is that if a re-INVITE is sent by Asterisk and it
+ receives a provisional response, but no final response, then the
+ dialog is never torn down. In addition to leaking memory, this
+ also leaks file descriptors and will eventually lead to Asterisk
+ no longer being able to process calls. This patch just keeps
+ track of whether there is an outstanding re-INVITE, and if there
+ is goes ahead and cleans up everything as though there was no
+ outstanding reinvite. Review:
+ https://reviewboard.asterisk.org/r/2009/ (closes issue
+ ASTERISK-19992) Reported by: Steve Davies Tested by: Steve
+ Davies, Terry Wilson
+
+2012-06-26 13:21 +0000 [r369366-369390] Matthew Jordan <mjordan@digium.com>
+
+ * main/adsi.c: Fix crash in unloading of res_adsi module When
+ res_adsi is unloaded, it removes the ADSI functions that it
+ previously installed by passing a NULL adsi_funcs pointer to
+ ast_adsi_install_funcs. This function was not checking whether or
+ not the adsi_funcs pointer passed in was NULL before
+ dereferencing it to check whether or not the version of the
+ functions matches what the core was expecting it. This patch
+ makes it so that the version is only checked if a potentially
+ valid adsi_funcs pointer was passed in. Passing in NULL removes
+ the installed functions, bypassing the version check.
+
+ * main/cdr.c: Tweak CDR change in r369351 As Tilghman pointed out
+ on review 1996, the check to see if a CDR end time has been set
+ is sufficient to know whether or not the duration value can be
+ used. The check-in done for r369351 forgot to include this
+ change.
+
+2012-06-25 19:13 +0000 [r369352] Mark Michelson <mmichelson@digium.com>
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: Re-fix how local
+ tag is generated when sending a 481 to an INVITE. Match our local
+ tag to whatever to-tag was sent in the initial INVITE. Because
+ the size of the to-tag may not fit in the buffer in the sip_pvt,
+ it has been changed to a string field. (closes issue
+ ASTERISK-19892) reported by Walter Doekes Review:
+ https://reviewboard.asterisk.org/r/1977
+
+2012-06-25 19:12 +0000 [r369351] Matthew Jordan <mjordan@digium.com>
+
+ * main/cdr.c: Fix incorrect duration reporting in CDRs created in
+ batch mode Certain places in core/cdr.c would, if the duration
+ value were 0, calculate the duration as being the delta between
+ the current time and the time at which the CDR record was
+ started. While this does not typically cause a problem in
+ non-batch mode, this can cause an issue in batch mode where CDR
+ records are gathered and written long after those calls have
+ ended. In particular, this affects calls that were never
+ answered, as those are expected to have a duration of 0. Often,
+ this would result in CDR logs with a significant number of calls
+ with lengthy durations, but dispositions of "BUSY". Note that
+ this does not affect cdr_csv, as that backend does not use
+ ast_cdr_getvar and instead directly reports the duration value.
+ The affected core backends include cdr_apative_odbc and
+ cdr_custom; other extended or deprecated CDR backends may
+ potentially still directly manipulate the duration values. (issue
+ ASTERISK-19860) Reported by: Thomas Arimont (issue AST-883)
+ Reported by: Thomas Arimont Tested by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1996/
+
+2012-06-25 15:57 +0000 [r369327] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: Fix Bridge application occasionally returning to
+ the wrong location. * Fix do_bridge_masquerade() getting the
+ resume location from the zombie channel. The code must not touch
+ a clone channel after it has masqueraded it. The clone channel
+ has become a zombie and is starting to hangup. (closes issue
+ ASTERISK-19985) Reported by: jamicque Patches:
+ jira_asterisk_19985_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: jamicque
+
+2012-06-25 15:50 +0000 [r369302-369324] Mark Michelson <mmichelson@digium.com>
+
+ * main/adsi.c (added): Forgot to svn add this file in my last
+ commit.
+
+ * res/res_adsi.exports.in (removed), include/asterisk/adsi.h,
+ main/Makefile, res/res_adsi.c: Eliminate embedding of res_adsi.so
+ module. The way this is done is to stop using the optional API.
+ Instead, res_adsi.so, when loaded fills in a table of function
+ pointers. Review: https://reviewboard.asterisk.org/r/1991
+
+ * channels/chan_sip.c: Be more consistent with the return code for
+ requests received from invalid domain. When Asterisk receives an
+ INVITE from an external domain when allowexternaldomains=no send
+ a 403 instead of a 404. This is consistent with Asterisk's
+ behavior when receiving a REGISTER in this situation. (Closes
+ issue ASTERISK-19601) Reported by Matthew Jordan Patches:
+ ASTERISK-19601-no401.patch uploaded by Mark Michelson (License
+ #5049)
+
+2012-06-23 00:04 +0000 [r369235-369282] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: Fix Bridge application and AMI Bridge action
+ error handling. * Fix AMI Bridge action disconnecting the AMI
+ link on error. * Fix AMI Bridge action and Bridge application not
+ checking if their masquerades were successful. * Fix Bridge
+ application running the h-exten when it should not. * Made
+ do_bridge_masquerade() return if the masquerade was successful so
+ the Bridge application and AMI Bridge action could deal with it
+ correctly. * Made bridge_call_thread_launch() hangup the passed
+ in channels if the bridge_call_thread fails to start. Those
+ channels would have been orphaned. * Made builtin_atxfer() check
+ the success of the transfer masquerade setup.
+
+ * apps/app_queue.c: Explicitly check caller hangup in app Queue
+ rather than a polluted res2 value.
+
+ * apps/app_dial.c: Check if PBX was started and fix F and F(x)
+ action logic in Dial application.
+
+ * main/ccss.c: Check if PBX was started for generic CCSS recall.
+
+ * channels/chan_sip.c: Change incorrect chan_sip zombie hangup
+ debug message. They are all zombies now.
+
+2012-06-22 19:28 +0000 [r369214] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Don't crash on a guest directmedia call A
+ sip_pvt may not have relatedpeer set if a call doesn't match up
+ with a peer. If there is no relatedpeer, there is no direct media
+ ACL to apply, so just return that it is allowed. (closes issue
+ ASTERISK-20040) Reported by: Terry Wilson
+
+2012-06-22 17:14 +0000 [r369195] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c: Don't parse media stream state for SIP video
+ streams The sendonly/recvonly/sendrecv/inactive media stream
+ attributes were parsed for video, but nothing was ever done with
+ them. With this code removed, an UNSUPPORTED message is produced
+ when these attributes are used in conjunction with a video stream
+ which is the better behavior since they were never really
+ supported in the first place.
+
+2012-06-20 17:33 +0000 [r369130-369146] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c: fix
+ locking issue on empty callList (issue ASTERISK-19298) Reported
+ by: Dmitry Melekhov Patches: ASTERISK-18322-2.patch
+
+ * addons/chan_ooh323.c: fix compile error (1.8 don't have
+ ast_channel_name macro)
+
+2012-06-20 02:03 +0000 [r369108] Michael L. Young <elgueromexicano@gmail.com>
+
+ * include/asterisk/netsock2.h, main/netsock2.c: Fix NULL pointer
+ segfault in ast_sockaddr_parse() While working with
+ ast_parse_arg() to perform a validity check, a segfault occurred.
+ The segfault occurred due to passing a NULL pointer to
+ ast_sockaddr_parse() from ast_parse_arg(). According to the
+ documentation in config.h, "result pointer to the result. NULL is
+ valid here, and can be used to perform only the validity checks."
+ This patch fixes the segfault by checking for a NULL pointer.
+ This patch also adds documentation to netsock2.h about why it is
+ necessary to check for a NULL pointer. (Closes issue
+ ASTERISK-20006) Reported by: Michael L. Young Tested by: Michael
+ L. Young Patches: asterisk-20006-netsock-null-ptr.diff uploaded
+ by Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/1990/
+
+2012-06-19 23:28 +0000 [r369090] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c: check rtptimeouts in ooh323 channels as per
+ config file (rtp voice, video, udptl except rtcp) (closes issue
+ ASTERISK-19179) Reported by: TSAREGORODTSEV Yury Patches:
+ 19179-ooh323-2.patch
+
+2012-06-19 15:30 +0000 [r369066] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix request routing issue when outboundproxy
+ is used. Asterisk was incorrectly setting the destination of
+ CANCELs and ACKs for error responses to the URI of the initial
+ INVITE. This resulted in further requests, such as INVITEs with
+ authentication credentials, to be routed incorrectly. Instead,
+ when these CANCEL or ACKs are to be sent, we should simply keep
+ the destination the same as what it previously was. There is no
+ need to alter it any. (closes issue ASTERISK-20008) Reported by
+ Marcus Hunger Patches: ASTERISK-20008.patch uploaded by Mark
+ Michelson (license #5049)
+
+2012-06-18 18:07 +0000 [r369043] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: Fix monitoring calls put in a parking lot. * Fix
+ a regression that was introduced by -r366167 which effectively
+ disabled monitoring parked calls. (closes issue ASTERISK-20012)
+ Reported by: sdolloff Tested by: rmudgett
+
+2012-06-15 15:57 +0000 [r369001-369002] Kevin P. Fleming <kpfleming@digium.com>
+
+ * build_tools/find_missing_support_level (added): Add a script to
+ enable finding source files without support-levels defined.
+
+ * main/devicestate.c, main/astfd.c, main/ssl.c,
+ main/taskprocessor.c, main/astobj2.c, main/indications.c,
+ main/config.c, main/loader.c, main/term.c, main/cli.c,
+ channels/sig_analog.c, main/framehook.c, main/strcompat.c,
+ main/plc.c, res/ais/evt.c, main/fskmodem_int.c, main/syslog.c,
+ main/stdtime/localtime.c, main/db.c, main/bridging.c,
+ channels/sig_ss7.c, main/datastore.c, main/sched.c,
+ channels/sip/sdp_crypto.c, main/pbx.c, main/strings.c,
+ channels/vcodecs.c, channels/iax2-provision.c, main/aoc.c,
+ pbx/dundi-parser.c, main/cel.c, channels/iax2-parser.c,
+ main/chanvars.c, main/netsock.c, main/data.c, main/srv.c,
+ channels/chan_misdn.c, main/privacy.c,
+ channels/sip/dialplan_functions.c, main/fixedjitterbuf.c,
+ main/test.c, main/audiohook.c, main/alaw.c, main/asterisk.c,
+ main/timing.c, main/global_datastores.c, main/fskmodem_float.c,
+ main/ccss.c, channels/sip/reqresp_parser.c,
+ channels/misdn/isdn_msg_parser.c, main/utils.c, main/xml.c,
+ main/autochan.c, main/enum.c, channels/misdn/isdn_lib.c,
+ main/fskmodem.c, channels/misdn_config.c, main/io.c,
+ res/ael/pval.c, main/channel.c, main/cdr.c, main/ulaw.c,
+ main/dial.c, main/tdd.c, main/heap.c, channels/console_gui.c,
+ channels/misdn/ie.c, main/logger.c, channels/console_board.c,
+ main/app.c, main/image.c, main/dns.c, main/lock.c, main/stun.c,
+ main/dnsmgr.c, channels/sip/srtp.c, main/translate.c,
+ main/slinfactory.c, main/jitterbuf.c, main/acl.c,
+ channels/sig_pri.c, main/tcptls.c, main/hashtab.c,
+ main/abstract_jb.c, main/callerid.c, main/file.c,
+ res/snmp/agent.c, main/astmm.c, channels/misdn/portinfo.c,
+ main/event.c, channels/sip/config_parser.c, channels/vgrabbers.c,
+ main/xmldoc.c, main/dsp.c, main/udptl.c, main/netsock2.c,
+ main/autoservice.c, main/rtp_engine.c, main/frame.c,
+ main/security_events.c, res/ais/clm.c, main/threadstorage.c,
+ main/say.c, channels/console_video.c: Add support-level
+ indications to many more source files. Since we now have tools
+ that scan through the source tree looking for files with specific
+ support levels, we need to ensure that every file that is a
+ component of a 'core' or 'extended' module (or the main Asterisk
+ binary) is explicitly marked with its support level. This patch
+ adds support-level indications to many more source files in tree,
+ but avoids adding them to third-party libraries that are included
+ in the tree and to source files that don't end up involved in
+ Asterisk itself.
+
+2012-06-14 15:23 +0000 [r368898-368927] Mark Michelson <mmichelson@digium.com>
+
+ * main/Makefile: Revert Makefile change to remove embedding
+ res_adsi.so The change has resulted in a linking error for
+ certain versions of GCC. This is much worse than the original
+ issue, so for now, temporarily revert the change. A more thorough
+ change will be sought out.
+
+ * funcs/func_volume.c: Fix a deadlock that occurs when func_volume
+ is used on a local channel. This was discovered by trying to
+ perform a call forward to an extension that makes use of
+ func_volume. When the local channel is optimized away, the
+ datastore on the local;2 channel would have its audiohook
+ destroyed rather than detaching the audiohook from the channel
+ and then destroying it. With this patch, func_volume's datastore
+ destructor takes the proper route of detaching the audiohook and
+ then destroying it. (closes issue ASTERISK-19611) reported by
+ Volker Sauer Patches: ASTERISK-19611.patch uploaded by Mark
+ Michelson (license #5049)
+
+2012-06-13 20:26 +0000 [r368894] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_smdi.c, res/res_adsi.c: Mark res_smdi/res_adsi as 'core'
+ supported modules Recently, various issues surrounding weak
+ symbols have caused problems with modules that rely on that
+ feature to be enabled in menuselect. This includes app_voicemail
+ and chan_dahdi, as they both rely upon res_smdi and res_adsi,
+ which, in certain circumstances, may not be enabled by default in
+ menuselect. Because res_smdi/res_adsi are dependencies for
+ chan_dahdi/app_voicemail, this patch marks both as 'core'
+ supported modules. This will allow both app_voicemail and
+ chan_dahdi to be enabled as well, regardless of whether or not
+ that system supports weak symbols. (issue AST-900) Reported by:
+ Thomas Arimont (issue AST-885) Reported by: Denis Alberto
+ Martinez
+
+2012-06-13 19:00 +0000 [r368873] Mark Michelson <mmichelson@digium.com>
+
+ * main/Makefile: Remove forced linking of res_adsi.o In GCC 4.5+
+ the result is that Asterisk has a phantom module loaded at
+ startup, claiming to be res_adsi. (closes issue ASTERISK-19920)
+ reported by Leif Madsen
+
+2012-06-13 14:27 +0000 [r368830-368852] Matthew Jordan <mjordan@digium.com>
+
+ * Makefile: Do not install empty directories; add ASTLIBDIR r368830
+ modified the installation script to only create a directory if
+ that directory does not exist. If some directory variable was
+ empty, it would attempt to create the empty location. It also
+ failed to create the ASTLIBDIR directory. This patch fixes it
+ such that the correct directories are made and only created if a
+ value specifying them actually exists.
+
+ * Makefile: Do not perform install on existing directories If a
+ directory already exists, performing a 'make install' will remove
+ the permissions associated with the current directory and replace
+ them with the permissions of the user executing the install. This
+ patch changes this behavior to only perform an install on the
+ directory if the directory does not exist. Thus, if a user later
+ changes the permissions on that directory, those permissions will
+ be preserved in subsequent installs. Review:
+ https://reviewboard.asterisk.org/r/1986 Review:
+ https://reviewboard.asterisk.org/r/1864 (closes issue
+ ASTERISK-19492) Reported by: Karl Fife Tested by: Paul Belanger,
+ Tilghman Lesher patches: ASTERISK-19492 by pabelanger (uploaded
+ by mjordan)
+
+2012-06-12 15:36 +0000 [r368807] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Set the Caller ID "tag" on peers even if
+ remote party information is present. On incoming calls, we were
+ setting the cid_tag on the dialog only if there was no remote
+ party information (Remote-Party-ID or P-Asserted-Identity)
+ present. The Caller ID tag is an invented parameter, though, and
+ should be set no matter the circumstance. (closes issue
+ ASTERISK-19859) Reported by Thomas Arimont (closes issue AST-884)
+ Reported by Trey Blancher
+
+2012-06-11 17:03 +0000 [r368759] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/channel.h, channels/chan_iax2.c, main/channel.c,
+ channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/chan_sip.c: Fix deadlock potential with
+ ast_set_hangupsource() calls. Calling ast_set_hangupsource() with
+ the channel lock held can result in a deadlock because the
+ function also locks the bridged channel. (issue ASTERISK-19537)
+ (closes issue AST-891) Reported by: Guenther Kelleter Tested by:
+ Guenther Kelleter (closes issue ASTERISK-19801) Reported by: Alec
+ Davis
+
+2012-06-11 15:13 +0000 [r368719-368738] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_queue.c, main/loader.c, channels/chan_dahdi.c,
+ res/res_config_odbc.c, channels/sip/dialplan_functions.c,
+ pbx/pbx_config.c, apps/app_directory.c, res/res_odbc.c,
+ res/res_speech.c, apps/app_voicemail.c, main/udptl.c,
+ channels/sip/sdp_crypto.c, channels/chan_sip.c, res/res_fax.c,
+ main/say.c, funcs/func_strings.c, channels/sip/reqresp_parser.c:
+ Fix coverity UNUSED_VALUE findings in core support level files
+ Most of these were just saving returned values without using them
+ and in some cases the variable being saved to could be removed as
+ well. (issue ASTERISK-19672)
+
+ * main/md5.c: Fix compilation in dev-mode Backport a compilation
+ fix in md5.c from trunk that only showed up in dev-mode under
+ certain compiler versions.
+
+2012-07-10 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.14.0 Released.
+
+2012-07-06 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.14.0-rc2 Released.
+
+ * AST-2012-010: Possible Resource Leak on Uncompleted Re-INVITE
+ transactions
+
+ * AST-2012-011: Remote Crash Vulnerability in VoiceMail Application
+
+ * Fix crash on a guest directmedia call
+
+ A sip_pvt may not have relatedpeer set if a call doesn't match up
+ with a peer. If there is no relatedpeer, there is no direct media
+ ACL to apply, so just return that is is allowed.
+
+ (closes issue ASTERISK-20040)
+
+ * Fix request routing issue when outboundproxy is used
+
+ Asterisk was incorrectly setting the destination of CANCELs and ACKs
+ for error responses to the URI of the initial INVITE. This resulted
+ in further requests, such as INVITEs with authentication
+ credentials, to be routed incorrectly. Instead when these CANCEL or
+ ACKs are to be esnt, we should simply keep the destination the same
+ as what it previously was. There is no need to alter it any.
+
+ (closes issue ASTERISK-20008)
+
+ * Fix monitoring calls put in a parking lot
+
+ Fix a regression that was introduced by r366167 which effectively
+ disabled monitoring parked calls.
+
+ (closes issue ASTERISK-20012)
+
+2012-06-08 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.14.0-rc1 Released.
+
+2012-06-06 21:27 +0000 [r368644] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c: Fix POTS flash hook
+ to orignate a second call deadlock. A deadlock can occur when a
+ POTS phone tries to flash hook to originate a second call for
+ 3-way or transfer. If another process is scanning the channels
+ container when the POTS line flash hooks then a deadlock will
+ occur. * Release the channel and private locks when creating a
+ new channel as a result of a flash hook. (closes issue
+ ASTERISK-19842) Reported by: rmudgett Tested by: rmudgett
+
+2012-06-06 19:13 +0000 [r368625] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix a specific scenario where ACKs are not
+ matched. If a dialog-starting INVITE contains a to-tag, then
+ Asterisk will respond with a 481. In this case, the resulting
+ incoming ACK would not be matched, so Asterisk would continue
+ retransmitting the 481 until the transaction times out. There
+ were two issues. Asterisk, upon creating a sip_pvt would generate
+ a local tag. However, when the time came to transmit the 481,
+ since there was a to-tag in the INVITE, Asterisk would place this
+ original to-tag in the 481 response. When the ACK came in,
+ Asterisk would attempt to match the to-tag in the ACK to the
+ generated local tag. Unfortunately, Asterisk never actually
+ transmitted a response with the generated local tag, so the
+ to-tag in the ACK would not match. The other problem was that
+ when the 481 was sent, nothing was set on the sip_pvt to indicate
+ what CSeq is expected in the ACK. To fix the first problem, we
+ zero out the to-tag seen in the incoming INVITE. This way,
+ Asterisk, when time to send a response, will send its generated
+ local tag instead. To fix the second problem, we set the
+ sip_pvt's pendinginvite to the CSeq of the INVITE when we send a
+ 481. (closes issue ASTERISK-19892) Reported by Mark Michelson
+
+2012-06-06 17:20 +0000 [r368604] Matthew Jordan <mjordan@digium.com>
+
+ * build_tools/make_version: Add feature modifier to versions
+ produced from branches Certain branches, such as Certified
+ Asterisk, may have a modifier added to them that specifies the
+ features available in that branch. For branches, this modifier is
+ expected to be reflected in the location of the branch in
+ subversion. For example, a subversion of URL of
+ /certified/branches/1.8.11 would have a feature modifier of
+ 'certified'. This is slightly different then how features are
+ determined for tags, where the feature is part of the actual tag
+ name, e.g., "10.5.0-digiumphones". In keeping with the
+ nomenclature used for tags, the feature specifier for branches is
+ translated and placed after the revision numbers. For the example
+ given previously, this would result in a branch version of
+ "Asterisk SVN-branch-1.8.11-cert-rXXXXXX".
+
+2012-06-06 16:07 +0000 [r368586] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c: Ensure overlapping hold flags do not
+ conflict When changing between different modes of hold, the flags
+ were not being cleared out properly causing a failure to change
+ hold states. (closes issue ASTERISK-19919) Patch-by: Morten
+ Tryfoss Reported-by: Morten Tryfoss
+
+2012-06-06 01:08 +0000 [r368567] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: Fix parked call performing a DTMF blind transfer
+ after being retrieved. When a parked call was retrieved from the
+ parking lot, it could not do a blind transfer because it caused
+ the involved calls to be hung up unconditionally. * Made the
+ ParkedCall application return the ast_bridge_call() return value.
+ (closes issue ABE-2862) Reported by: Vlad Povorozniuc
+
+2012-06-05 15:26 +0000 [r368520-368533] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_minivm.c: Resolve some build warnings My newly upgraded
+ compiler caught these usages of uninitialized values. They
+ weren't actually used.
+
+ * apps/app_voicemail.c: Ensure that pages and emails are sent using
+ RFC822-compliant date format When localization was added to
+ app_voicemail, these headers were altered when they should have
+ remained in en_US format for RFC compliance. This reverts the
+ changes to those two lines. (closes issue ASTERISK-19876)
+
+2012-06-04 21:56 +0000 [r368498] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Relay proper SIP responses on calling side.
+ Revision 351130 broke corect HANGUPCAUSE setting for the 404 case
+ in chan_sip. Other cases were also potentially broken. This patch
+ fixes the relaying of causes to be what they used to be. (closes
+ issue ASTERISK-19914) Reported by Pavel Troller Tested by Walter
+ Doekes (via a reviewboard test to be committed later) Patches:
+ chan_sip.diff uploaded by Pavel Troller (license #6302)
+
+2012-06-04 21:10 +0000 [r368405-368469] Richard Mudgett <rmudgett@digium.com>
+
+ * UPGRADE.txt: Document BLINDTRANSFER behavior change. (issue
+ ASTERISK-19322) (closes issue ASTERISK-19875) Reported by: call
+
+ * main/channel.c: Fix potential deadlock between masquerade and
+ chan_local. * Restructure ast_do_masquerade() to not hold channel
+ locks while it calls ast_indicate(). * Simplify many calls to
+ ast_do_masquerade() since it will never return a failure now. If
+ it does fail internally because a channel driver callback
+ operation failed, the only thing ast_do_masquerade() can do is
+ generate a warning message about strange things may happen and
+ press on. * Fixed the call to ast_bridged_channel() in
+ ast_do_masquerade(). This change fixes half of the deadlock
+ reported in ASTERISK-19801 between masquerades and chan_iax.
+ (closes issue ASTERISK-19537) Reported by: rmudgett Tested by:
+ rmudgett Review: https://reviewboard.asterisk.org/r/1915/
+
+2012-06-01 23:21 +0000 [r368308] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_stack.c: Fix deadlock when Gosub used with alternate
+ dialplan switches. Attempting to remove a channel from
+ autoservice with the channel lock held will result in deadlock. *
+ Restructured gosub_exec() to not call ast_parseable_goto() and
+ ast_exists_extension() with the channel lock held. (closes issue
+ ASTERISK-19764) Reported by: rmudgett Tested by: rmudgett
+
+2012-06-01 18:18 +0000 [r368218] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: Improve SDP parsing warning messages *
+ 'Unsupported media type' is only reported when that is in fact
+ the case, not when a supported media type is included in an 'm'
+ line that has an invalid format. * All warning messages related
+ to parsing 'm' lines now include the 'm' line contents. * (minor
+ bugfix) newline added to port-number-zero warning messages. *
+ Warning messages improved to use RFC-specified terminology for
+ various items. * Warnings for offers that include more than one
+ port for a single media type now include the media type. Review:
+ https://reviewboard.asterisk.org/r/1811/
+
+2012-06-01 03:25 +0000 [r368092] Michael L. Young <elgueromexicano@gmail.com>
+
+ * funcs/func_channel.c: Add documentation to function CHANNEL for
+ options echocan_mode and buffers The ability to set
+ "echocan_mode" and "buffers" through the dialplan was added to
+ chan_dahdi some time ago. This patch adds some documentation to
+ func_channel. (Closes issue ASTERISK-19911) Reported by: Dale
+ Noll Tested by: Michael L. Young Patches:
+ asterisk-19911-branch18.diff uploaded by Michael L. Young
+ (license 5026) Review: https://reviewboard.asterisk.org/r/1949/
+
+2012-05-31 18:00 +0000 [r367906-368039] Richard Mudgett <rmudgett@digium.com>
+
+ * main/db1-ast/btree/bt_open.c, apps/app_queue.c,
+ channels/chan_iax2.c, pbx/pbx_config.c, res/ael/pval.c,
+ main/tcptls.c, main/manager.c, res/res_config_odbc.c,
+ channels/chan_sip.c, channels/chan_agent.c, funcs/func_math.c,
+ main/features.c: Coverity Report: Fix issues for error type
+ REVERSE_INULL (core modules) * Fixes findings:
+ 0-2,5,7-15,24-26,28-31 (issue ASTERISK-19648) Reported by: Matt
+ Jordan
+
+ * channels/sig_pri.c, channels/sig_ss7.c: Use the
+ DEADLOCK_AVOIDANCE() macro instead. (issue ASTERISK-19854)
+
+ * channels/sig_pri.c, channels/sig_ss7.c: Fix deadlock when
+ executing CLI "pri show channels" and "ss7 show channels"
+ commands. * Fix sig_pri_lock_owner() to avoid deadlock properly.
+ * Code pri_grab() better. * Fix sig_ss7_lock_owner() to avoid
+ deadlock properly. * Code ss7_grab() better. (closes issue
+ ASTERISK-19854) Reported by: Jaxon Patches:
+ jira_asterisk_19854_v1.8.6.patch (license #5621) patch uploaded
+ by rmudgett (Modified to do the same thing to sig_ss7) Tested by:
+ Jaxon
+
+ * apps/app_meetme.c: Coverity Report: Fix issues for error type
+ REVERSE_INULL (deprecated modules) * Fix only issue pointed out
+ by deprecated_REVERSE_INULL.txt for app_meetme.c in find_user().
+ * Change use of %i to %d in sscanf() in find_user(). The use of
+ %i gives unexpected parsing because it can accept hex, octal, and
+ decimal integer formats. * Changed other uses of %i in
+ app_meetme() to use %d for consistency. (issue ASTERISK-19648)
+ Reported by: Matt Jordan
+
+2012-05-29 18:30 +0000 [r367843] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_skinny.c: AST-2012-008: Fix remote crash
+ vulnerability in chan_skinny When a skinny session is
+ unregistered, the corresponding device pointer is set to NULL in
+ the channel private data. If the client was not in the on-hook
+ state at the time the connection was closed, the device pointer
+ can later be dereferenced if a message or channel event attempts
+ to use a line's pointer to said device. The patches prevent this
+ from occurring by checking the line's pointer in message handlers
+ and channel callbacks that can fire after an unregistration
+ attempt. (closes issue ASTERISK-19905) Reported by: Christoph
+ Hebeisen Tested by: mjordan, Damien Wedhorn Patches:
+ AST-2012-008-1.8.diff uploaded by mjordan (license 6283)
+ AST-2012-008-10.diff uploaded by mjordan (license 6283)
+
+2012-05-25 16:28 +0000 [r367781] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_iax2.c: AST-2012-007: Fix IAX receiving HOLD
+ without suggested MOH class crash. * Made schedule_delivery() set
+ the received frame f->data.ptr to NULL if the datalen is zero. *
+ Fix queue_signalling() memcpy() size error. * Made
+ queue_signalling() not use C++ keyword variable names. (closes
+ issue ASTERISK-19597) Reported by: mgrobecker Patches:
+ jira_asterisk_19597_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: rmudgett, Michael L. Young
+
+2012-05-25 02:27 +0000 [r367730] Michael L. Young <elgueromexicano@gmail.com>
+
+ * channels/chan_sip.c: Fix pvt_sip for inbound call to use peer's
+ allowtransfer setting The pvt_sip allowtransfer was not being set
+ to that of the peer's setting. Therefore, the global
+ allowtransfer setting was being used instead which would lead to
+ calls not being transfered if the global setting was set to 'no'
+ despite the setting on the peer being 'yes' and vice versa, calls
+ would be allowed to transfer even if the peer's setting was 'no'
+ but the global setting was 'yes'. (Closes issue ASTERISK-19856)
+ Reported by: Jacek Tested by: Michael L. Young, Jacek Patches:
+ issue-asterisk-19856-branch10-v3.diff uploaded by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/1923/
+
+2012-05-24 22:21 +0000 [r367469-367678] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_queue.c, apps/app_dial.c: Fix Dial I option ignored if
+ dial forked and one fork redirects. The Dial and Queue I option
+ is intended to block connected line updates and redirecting
+ updates. However, it is a feature that when a call is locally
+ redirected, the I option is disabled if the redirected call runs
+ as a local channel so the administrator can have an opportunity
+ to setup new connected line information. Unfortunately, the Dial
+ and Queue I option is disabled for *all* forked calls if one of
+ those calls is redirected. * Make the Dial and Queue I option
+ apply to each outgoing call leg independently. Now if one
+ outgoing call leg is locally redirected, the other outgoing calls
+ are not affected. * Made Dial not pass any redirecting updates
+ when forking calls. Redirecting updates do not make sense for
+ this scenario. * Made Queue not pass any redirecting updates when
+ using the ringall strategy. Redirecting updates do not make sense
+ for this scenario. * Fixed deadlock potential with chan_local
+ when Dial and Queue send redirecting updates for a local
+ redirect. * Converted the Queue stillgoing flag to a boolean
+ bitfield. (closes issue ASTERISK-19511) Reported by: rmudgett
+ Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1920/
+
+ * main/pbx.c: Fix WaitExten(x,m(musicclass)) string termination.
+ The AST_CONTROL_HOLD MOH class from the WaitExten application can
+ now be queued onto a channel, passed over local channels with the
+ /m option, and passed over IAX channels.
+
+2012-05-23 20:27 +0000 [r367416] Mark Michelson <mmichelson@digium.com>
+
+ * main/tcptls.c: Only call SSL_CTX_free if DO_SSL is defined.
+ Thanks to Paul Belanger for pointing out this error.
+
+2012-05-23 13:06 +0000 [r367362] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_sip.c: Update a peer's LastMsgsSent when the peer
+ is notified of waiting messages Previously, MWI logic utilized a
+ counter called 'lastmsgssent' to know whether or not MWI NOTIFY
+ requests had been sent to a specific peer. When MWI notifications
+ were changed to use the internal event framework, this value was
+ no longer needed for its original purpose. Hence, it was no
+ longer updated with the new/old message counts for a peer.
+ However, the value was still presented when, either by AMI or
+ CLI, a 'sip show peer [peer]' command was executed. The output of
+ the command would always display the erroneous value of
+ 32767/65535 for 'LastMsgsSent'. This patch makes it so that the
+ value of lastmsgssent is updated appropriately. The value should
+ now display the new/old message counts for a particular peer.
+ (closes issue ASTERISK-17866) Reported by: Steve Davies patches
+ by: ast-17866-rb1272.patch (License #5041 by irroot) Modified
+ slightly for this commit Review:
+ https://reviewboard.asterisk.org/r/1939
+
+2012-05-22 17:14 +0000 [r367266-367292] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/channel.h, main/cel.c, main/asterisk.c,
+ main/channel.c, include/asterisk/cel.h: Fix race condition for
+ CEL LINKEDID_END event This patch fixes to situations that could
+ cause the CEL LINKEDID_END event to be missed. 1) During a core
+ stop gracefully, modules are unloaded when ast_active_channels ==
+ 0. The LINKDEDID_END event fires during the channel destructor.
+ This means that occasionally, the cel_* module will be unloaded
+ before the channel is destroyed. It seemed generally useful to
+ wait until the refcount of all channels == 0 before unloading, so
+ I added a channel counter and used it in the shutdown code. 2)
+ During a masquerade, ast_channel_change_linkedid is called. It
+ calls ast_cel_check_retire_linkedid which unrefs the linkedid in
+ the linkedids container in cel.c. It didn't ref the new linkedid.
+ Now it does. Review: https://reviewboard.asterisk.org/r/1900/
+
+ * channels/chan_sip.c: Resolve crash in subscribing for MWI
+ notifications ASTOBJ_UNREF sets the variable to NULL after
+ unreffing it, so the variable should definitely not be used after
+ that. To solve this in the two cases that affect subscribing for
+ MWI notifications, we instead save the ref locally, and unref
+ them in the error conditions. (closes issue ASTERISK-19827)
+ Reported by: B. R Review:
+ https://reviewboard.asterisk.org/r/1940/
+
+2012-05-18 17:47 +0000 [r367002-367027] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_dahdi.c, main/say.c: Address MISSING_BREAK static
+ analysis reports some more. This addresses core findings 4 and 6.
+ Moises Silva helped me by stating that a break could be safely
+ added to the case where it is added in chan_dahdi.c In say.c, I
+ have added a comment indicating that static analysis complains
+ but that it is currently unknown if this is correct. This fixes
+ all core findings of this type. (closes issue ASTERISK-19662)
+ reported by Matthew Jordan
+
+ * include/asterisk/tcptls.h, main/tcptls.c, channels/chan_sip.c:
+ Fix memory leak of SSL_CTX structures in TLS core. SSL_CTX
+ structures were allocated but never freed. This was a bigger
+ issue for clients than servers since new SSL_CTX structures could
+ be allocated for each connection. Servers, on the other hand,
+ typically set up a single SSL_CTX for their lifetime. This is
+ solved in two ways: 1. In __ssl_setup(), if a tcptls_cfg has an
+ ssl_ctx on it, it is freed so that a new one can take its place.
+ 2. A companion to ast_ssl_setup() called ast_ssl_teardown() has
+ been added so that servers can properly free their SSL_CTXs.
+ (issue ASTERISK-19278)
+
+2012-05-18 15:42 +0000 [r366944] Matthew Jordan <mjordan@digium.com>
+
+ * main/cli.c, channels/chan_sip.c, funcs/func_odbc.c: Fix more
+ memory leaks This patch adds to what was fixed in r366880.
+ Specifically, it addresses the following: * chan_sip: dispose of
+ an allocated frame in off nominal code paths in sip_rtp_read *
+ func_odbc: when disposing of an allocated resultset, ensure that
+ any rows that were appended to that resultset are also disposed
+ of * cli: free the created return string buffer in another off
+ nominal code path (issue ASTERISK-19665) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/1922/
+
+2012-05-18 14:16 +0000 [r366882] Kinsey Moore <kmoore@digium.com>
+
+ * channels/sip/config_parser.c: Reorder and renumber tests
+ appropriately It appears that a patch did not apply properly when
+ adding tests 12 and 13 and test 11 was duplicated. These tests
+ have been reordered and renumbered such that they make sense.
+
+2012-05-18 13:58 +0000 [r366880] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_calendar_caldav.c, res/res_musiconhold.c,
+ res/res_jabber.c, apps/app_queue.c, channels/chan_iax2.c,
+ main/enum.c, main/editline/term.c, main/config.c, res/res_srtp.c,
+ main/editline/tokenizer.c, main/cli.c, channels/chan_dahdi.c,
+ main/data.c, funcs/func_odbc.c, apps/app_minivm.c,
+ main/features.c, main/editline/readline.c,
+ channels/sip/config_parser.c, main/xmldoc.c, res/res_calendar.c,
+ apps/app_voicemail.c, res/res_rtp_asterisk.c, main/netsock2.c,
+ res/res_calendar_icalendar.c, res/res_calendar_exchange.c,
+ main/pbx.c, apps/app_page.c, channels/chan_sip.c,
+ funcs/func_dialgroup.c, apps/app_record.c: Fix a variety of
+ memory leaks This patch addresses a number of memory leaks in a
+ variety of modules that were found by a static analysis tool. A
+ brief summary of the changes: * app_minivm: free ast_str objects
+ on off nominal paths * app_page: free the ast_dial object if the
+ requested channel technology cannot be appended to the dialing
+ structure * app_queue: if a penalty rule failed to match any
+ existing rule list names, the created rule would not be inserted
+ and its memory would be leaked * app_read: dispose of the created
+ silence detector in the presence of off nominal circumstances *
+ app_voicemail: dispose of an allocated unique ID field for MWI
+ event un-subscribe requests in off nominal paths; dispose of
+ configuration objects when using the secret.conf option *
+ chan_dahdi: dispose of the allocated frame produced by
+ ast_dsp_process * chan_iax2: properly unref peer in CLI command
+ "iax2 unregister" * chan_sip: dispose of the allocated frame
+ produced by sip_rtp_read's call of ast_dsp_process; free memory
+ in parse unit tests * func_dialgroup: properly deref ao2 object
+ grhead in nominal path of dialgroup_read * func_odbc: free
+ resultset in off nominal paths of odbc_read * cli: free
+ match_list in off nominal paths of CLI match completion * config:
+ free comment_buffer/list_buffer when configuration file load is
+ unchanged; free the same buffers any time they were created and
+ config files were processed * data: free XML nodes in various
+ places * enum: free context buffer in off nominal paths *
+ features: free ast_call_feature in off nominal paths of
+ applicationmap config processing * netsock2: users of
+ ast_sockaddr_resolve pass in an ast_sockaddr struct that is
+ allocated by the method. Failures in ast_sockaddr_resolve could
+ result in the users of the method not knowing whether or not the
+ buffer was allocated. The method will now not allocate the
+ ast_sockaddr struct if it will return failure. * pbx: cleanup
+ hash table traversals in off nominal paths; free ignore pattern
+ buffer if it already exists for the specified context * xmldoc:
+ cleanup various nodes when we no longer need them *
+ main/editline: various cleanup of pointers not being freed before
+ being assigned to other memory, cleanup along off nominal paths *
+ menuselect/mxml: cleanup of value buffer for an attribute when
+ that attribute did not specify a value * res_calendar*: responses
+ are allocated via the various *_request method returns and should
+ not be allocated in the various write_event methods; ensure
+ attendee buffer is freed if no data exists in the parsed node;
+ ensure that calendar objects are de-ref'd appropriately *
+ res_jabber: free buffer in off nominal path * res_musiconhold:
+ close the DIR* object in off nominal paths * res_rtp_asterisk: if
+ we run out of ports, close the rtp socket object and free the rtp
+ object * res_srtp: if we fail to create the session in libsrtp,
+ destroy the temporary ast_srtp object (issue ASTERISK-19665)
+ Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1922
+
+2012-05-17 14:40 +0000 [r366791] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c: chan_sip: Fix missed locking of opposing pvt
+ for directmedia acl from r366547 It also required deadlock
+ avoidance since two sip_pvts structs needed to be locked
+ simultaneously. Trunk handles it differently, so this is a 1.8
+ and 10 patch only. (issue AST-876)
+
+2012-05-17 12:51 +0000 [r366740] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_calendar_ews.c, channels/chan_dahdi.c: Fix checking
+ bounds of array index after using it; improper sizeof This patch
+ fixes two problems pointed out by a static analysis tool. * In
+ chan_dahdi, when an event is handled the index of the sub channel
+ is first obtained. In very off nominal cases, the method that
+ determines the index can return a negative value. In the event
+ handling code, whether or not the index returned is valid was
+ being checked after that value was used to index into an array.
+ This patch makes it so the value is checked before any indexing
+ is done. * In res_calendar_ews, sizeof was being passed a pointer
+ instead of the struct to determine the amount of memory to
+ allocate. (issue ASTERISK-19651) Reported by: Matt Jordan (closes
+ issue ASTERISK-19671) Reported by: Matt Jordan
+
+2012-05-16 15:52 +0000 [r366597-366650] Mark Michelson <mmichelson@digium.com>
+
+ * main/http.c: Fix incorrect default port number for HTTP server.
+ Thanks to Tzafrir Cohen for bringing this up on the Asterisk
+ developers mailing list.
+
+ * channels/chan_sip.c: Correct misuse of ast_strip_quoted() when
+ getting a Diversion header's reason parameter. The use here was
+ assuming that the pointer would be updated, but the updated
+ string is actually returned by ast_strip_quoted() instead.
+
+2012-05-15 20:14 +0000 [r366547] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c: chan_sip: Check the right channel's host
+ address for directmediapermit/deny Prior to this patch, when
+ checking the addresses for directmediapermit and directmediadeny,
+ Asterisk would check the host address of the channel permit/deny
+ was specified, which differs from the expectations of both our
+ users and the development team. Instead, directmediapermit/deny
+ now checks against the address of the channel that the peer with
+ the ACL is connected to. (issue AST-876) Review:
+ https://reviewboard.asterisk.org/r/1899/
+
+2012-05-14 19:57 +0000 [r366389-366409] Mark Michelson <mmichelson@digium.com>
+
+ * pbx/dundi-parser.c: Fix two more coverity constant expression
+ result findings. These correspond to findings 0 and 1 in the core
+ findings of ASTERISK-19649. After contacting Mark Spencer, he was
+ unsure of what the intent behind these lines of code were, so
+ they are being axed. For Asterisk 1.8 and 10, the output of
+ debugging DUNDi frames will not be changed, but for trunk the
+ "Retry" portion will be omitted since it does not properly
+ distinguish retransmissions from initial frames. (closes issue
+ ASTERISK-19649) Reported by Matthew Jordan
+
+ * channels/chan_sip.c: Fix broken reinvite glare scenario. To make
+ a long story short, reinvite glares were broken because Asterisk
+ would invert the To and From headers when ACKing a 491 response.
+ The reason was because the initreq of the dialog was being
+ changed to the incoming glared reinvite instead of being set to
+ the outgoing glared reinvite. This change has three parts * In
+ handle_incoming, we never will reject an ACK because it has a
+ to-tag present, even if we think the request may be out of
+ dialog. * In handle_request_invite, we do not change the initreq
+ when receiving a reinvite to which we will respond with a 491. *
+ In handle_request_invite, several superflous settings up
+ pendinginvite have been removed since this is dones automatically
+ by transmit_response_reliable Review:
+ https://reviewboard.asterisk.org/r/1911
+
+2012-05-11 23:53 +0000 [r366296] Russell Bryant <russell@russellbryant.com>
+
+ * addons/format_mp3.c: format_mp3: Fix a possible crash mp3_read().
+ This patch fixes a potential crash in mp3_read() by not assuming
+ that dbuf has enough data to finish filling up the output buffer.
+ The patch also makes sure that the dbuf state gets reset after we
+ know we read everything out of it already. In passing, this patch
+ includes some other cleanups of this module, including stripping
+ trailing whitespace, formatting fixes based on coding guidelines,
+ and removing a number of unused members from the private state
+ struct. (closes issue ASTERISK-19761) Reported by: Chris
+ Maciejewsk Tested by: Chris Maciejewsk
+
+2012-05-10 23:38 +0000 [r366240] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: * Made ast_change_name() hold the channels
+ container lock while changing the channel name. * Eliminate
+ redundant list not empty check in clone_variables().
+
+2012-05-10 20:50 +0000 [r366167] Kinsey Moore <kmoore@digium.com>
+
+ * main/devicestate.c, pbx/dundi-parser.c, channels/chan_iax2.c,
+ channels/iax2-parser.c, main/config.c, res/res_monitor.c,
+ main/channel.c, main/cdr.c, res/ael/pval.c, main/data.c,
+ channels/chan_dahdi.c, main/tcptls.c, main/manager.c,
+ main/features.c, main/app.c, main/event.c, pbx/pbx_dundi.c,
+ res/res_odbc.c, main/xmldoc.c, apps/app_voicemail.c,
+ funcs/func_speex.c, main/pbx.c, res/res_calendar_icalendar.c,
+ channels/chan_sip.c, funcs/func_lock.c, channels/chan_agent.c,
+ channels/sip/reqresp_parser.c: Resolve FORWARD_NULL static
+ analysis warnings This resolves core findings from ASTERISK-19650
+ numbers 0-2, 6, 7, 9-11, 14-20, 22-24, 28, 30-32, 34-36, 42-56,
+ 82-84, 87, 89-90, 93-102, 104, 105, 109-111, and 115. Finding
+ numbers 26, 33, and 29 were already resolved. Those skipped were
+ either extended/deprecated or in areas of code that shouldn't be
+ disturbed. (Closes issue ASTERISK-19650)
+
+2012-05-10 16:47 +0000 [r366094] Jonathan Rose <jrose@digium.com>
+
+ * channels/iax2-provision.c, apps/app_queue.c,
+ channels/chan_iax2.c, res/ael/ael.flex, funcs/func_devstate.c,
+ main/asterisk.c, main/db.c, main/xmldoc.c, apps/app_voicemail.c,
+ main/pbx.c, channels/sig_analog.c, channels/chan_sip.c,
+ funcs/func_lock.c, main/features.c, main/acl.c: Coverity Report:
+ Fix issues for error type CHECKED_RETURN for core (issue
+ ASTERISK-19658) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1905/
+
+2012-05-10 16:10 +0000 [r366052] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Close the proper tcptls_session when session
+ creation fails. (issue AST-998) Reported by: Thomas Arimont
+ Tested by: Thomas Arimont
+
+2012-05-10 15:35 +0000 [r365989-366048] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_chanspy.c, apps/app_page.c, funcs/func_cdr.c,
+ main/features.c, apps/app_disa.c: Coverity Report: Fix issues for
+ error type UNINIT in Core supported modules (issue
+ ASTERISK-19652) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1909/
+
+ * codecs/codec_dahdi.c: Block on frameout if the hardware has
+ enough samples to complete a frame. Fixes some problems with
+ skipping audio in elaborate scenarios involving multiple codecs
+ by making codec_dahdi operate in a more synchronous fashion
+ similar to codec_g729. This change also fixes the use of file
+ conversion tools from Asterisk's CLI. This change may cause the
+ thread responsible for transcoding audio to block briefly (Shaun
+ Ruffell describes this as 'several milliseconds') while waiting
+ for the hardware transcoder. (closes issue ASTERISK-19643)
+ reported by: Shaun Ruffell Patches:
+ 0001-codec_dahdi-Block-on-frameout-the-hardware-has-enoug.patch
+ uploaded by Shaun Ruffell (license 5417)
+
+2012-05-09 16:11 +0000 [r365896] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Prevent sip_pvt refleak when an ast_channel
+ outlasts its corresponding sip_pvt. chan_sip was coded under the
+ assumption that a SIP dialog with an owner channel will always be
+ destroyed after the owner channel has been hung up. However,
+ there are situations where the SIP dialog can time out and auto
+ destruct before the corresponding channel has hung up. A typical
+ example of this would be if the 'h' extension in the dialplan
+ takes a long time to complete. In such cases,
+ __sip_autodestruct() would complain about the dialog being auto
+ destroyed with an owner channel still in place. The problem is
+ that even once the owner channel was hung up, the sip_pvt would
+ still be linked in its ao2_container because nothing would ever
+ unlink it. The fix for this is that if __sip_autodestruct() is
+ called for a sip_pvt that still has an owner channel in place,
+ the destruction is rescheduled for 10 seconds in the future. This
+ will continue until the owner channel is finally hung up. (closes
+ issue ASTERISK-19425) reported by David Cunningham Patches:
+ ASTERISK-19425.patch uploaded by Mark Michelson (License #5049)
+ (closes issue ASTERISK-19455) reported by Dean Vesvuio Tested by
+ Dean Vesvuio
+
+2012-05-08 20:14 +0000 [r365631-365692] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_followme.c: * Fix FollowMe memory leak on error paths in
+ app_exec(). * Fix FollowMe leaving recorded caller name file on
+ error paths in app_exec(). * Use correct buffer dimension define
+ in struct call_followme.moh[] and struct fm_args.namerecloc[].
+ This fixes unexpected namerecloc filename length restriction.
+
+ * apps/app_followme.c: * Fix accept/decline DTMF buffer overwrite
+ in FollowMe. * Made use MAX_YN_STRING define to make all
+ accept/decline DTMF buffers the same size. Just using 20 isn't
+ good enough when someone didn't get the memo. * Fix stupid use of
+ a global variable in FollowMe. (ynlongest) * Fix bit field
+ declarations in FollowMe. * Fix FollowMe n option documentation.
+
+2012-05-08 15:48 +0000 [r365574] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Send more accurate identification
+ information in dialog-info SIP NOTIFYs. This uses the calling
+ channel's caller ID and connected line information to populate
+ the remote and local identities in the dialog-info NOTIFY when an
+ extension is ringing. There is a bit of an oddity here, and that
+ is that we seed the remote target with the To header of the
+ outbound call rather than the from header. This is because it was
+ reported that seeding with the from header caused hints to be
+ broken with certain SNOM devices. A comment has been added to the
+ code to explain this. (closes issue ASTERISK-16735) reported by
+ Maciej Krajewski patches: local_remote_hint2.diff uploaded by
+ Mark Michelson (license #5049) 16735_tweak1.diff uploaded by Mark
+ Michelson (license #5049) Tested by Niccolo Belli
+
+2012-05-07 18:40 +0000 [r365476] Richard Mudgett <rmudgett@digium.com>
+
+ * tests/test_config.c: Fix type punned compiler warning in
+ test_config.c
+
+2012-05-07 18:36 +0000 [r365474] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_voicemail.c, main/pbx.c: Support VoiceMail d() option
+ when extension does not exist in channel's context The VoiceMail
+ d([c]) option is documented to accept digits for a new extension
+ in context <c>, if played during the greeting. This option works
+ fine if the extension being redirected to has an extension with
+ the same initial digit in the channel's current context. If that
+ digit did not happen to exist in some extension, a dialplan match
+ would fail and the user would not be redirected. This patch fixes
+ it such that if the <c> option is used, the extensions are
+ matched in that context as opposed to the caller's original
+ context. (closes issue ASTERISK-18243) Reported by: mjordan
+ Tested by: mjordan Review:
+ https://reviewboard.asterisk.org/r/1892
+
+2012-05-07 16:01 +0000 [r365460] Mark Michelson <mmichelson@digium.com>
+
+ * main/audiohook.c, res/res_speech.c, channels/sig_analog.c,
+ main/abstract_jb.c, res/res_agi.c: Fix findings 0-3, 5, and 8 for
+ Coverity MISSING_BREAK errors. (Issue ASTERISK-19662)
+
+2012-05-04 22:12 +0000 [r365398] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_followme.c, channels/chan_iax2.c,
+ channels/sip/config_parser.c, pbx/pbx_config.c,
+ apps/app_chanspy.c, apps/app_stack.c, main/config.c,
+ apps/app_voicemail.c, channels/chan_sip.c, funcs/func_aes.c,
+ main/features.c: Fix many issues from the NULL_RETURNS Coverity
+ report Most of the changes here are trivial NULL checks. There
+ are a couple optimizations to remove the need to check for NULL
+ and outboundproxy parsing in chan_sip.c was rewritten to avoid
+ use of strtok. Additionally, a bug was found and fixed with the
+ parsing of outboundproxy when "outboundproxy=," was set. (Closes
+ issue ASTERISK-19654)
+
+2012-05-04 16:24 +0000 [r365313] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_local.c: Fix local channel chains optimizing
+ themselves out of a call. * Made chan_local.c:check_bridge()
+ check the return value of ast_channel_masquerade(). In long
+ chains of local channels, the masquerade occasionally fails to
+ get setup because there is another masquerade already setup on an
+ adjacent local channel in the chain. * Made the outgoing local
+ channel (the ;2 channel) flush one voice or video frame per
+ optimization attempt. * Made sure that the outgoing local channel
+ also does not have any frames in its queue before the masquerade.
+ * Made do the masquerade immediately to minimize the chance that
+ the outgoing channel queue does not get any new frames added and
+ thus unconditionally flushed. * Made block indication -1 (Stop
+ tones) event when the local channel is going to optimize itself
+ out. When the call is answered, a chain of local channels pass
+ down a -1 indication for each bridge. This blizzard of -1 events
+ really slows down the optimization process. (closes issue
+ ASTERISK-16711) Reported by: Alec Davis Tested by: rmudgett, Alec
+ Davis Review: https://reviewboard.asterisk.org/r/1894/
+
+2012-05-04 15:48 +0000 [r365298] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_rtp_asterisk.c: Fix core FINDING 2, FINDING 3, and
+ FINDING 4 from Coverity's CONSTANT_EXPRESSION_RESULT report.
+ These three all are in RTP code that attempts to print the number
+ of sequence number cycles in an RTCP RR report. The code was
+ masking out the upper 16 bits and then shifting the number right
+ by 16 bits. This led to an all zero result in all cases. The fix
+ is to do the shift without the bit masking. (issue
+ ASTERISK-19649)
+
+2012-05-03 14:54 +0000 [r365143-365159] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/h323/H323-MESSAGES.h,
+ addons/ooh323c/src/h323/H323-MESSAGESEnc.c,
+ addons/ooh323c/src/ooh323.c: Fix warning of Coverity Static
+ analysis, change H225ProtocolIdentifier from value to pointer per
+ functions that use this. (close issue ASTERISK-19670) Reported
+ by: Matt Jordan Patches: ASTERISK-19670.patch (License #5415)
+
+ * addons/ooh323c/src/ooq931.c: Fix coverity static analysis
+ warning, allocate full ie structure instead of without data
+ buffer (close issue ASTERISK-19674) Reported by: Matt Jordan
+ Patches: ASTERISK-19674.patch (License #5415)
+
+2012-06-04 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.13.0 Released.
+
+2012-05-30 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.13.0-rc2 Released.
+
+ * Resolve crash in subscribing for MWI notifications.
+
+ ASTOBJ_UNREF sets the variable to NULL after unreffing it, so the
+ variable should definitely not be used after that. To solve this in
+ the two cases that affect subscribing for MWI notifications, we
+ instead save the ref locally, and unref them in the error
+ conditions.
+
+ (closes issue ASTERISK-19827)
+ Reported by: B. R.
+ Review: https://reviewboard.asterisk.org/r/1940/
+
+ * AST-2012-007
+
+ * AST-2012-008
+
+
+2012-05-03 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.13.0-rc1 Released.
+
+2012-05-02 17:02 +0000 [r365006-365068] Terry Wilson <twilson@digium.com>
+
+ * main/cel.c, channels/chan_local.c: Don't leak a ref if out of
+ memory and can't link the linkedid If the ao2_link fails, we are
+ most likely out of memory and bad things are going to happen.
+ Before those bad things happen, make sure to clean up the
+ linkedid references. This patch also adds a comment explaining
+ why linkedid can't be passed to both local channel allocations
+ and combines two ao2_ref calls into 1. Review:
+ https://reviewboard.asterisk.org/r/1895/
+
+ * main/cel.c, channels/chan_local.c: Fix a CEL LINKEDID_END race
+ and local channel linkedids This patch has the ;2 channel inherit
+ the linkedid of the ;1 channel and fixes the race condition by no
+ longer scanning the channel list for "other" channels with the
+ same linkedid. Instead, cel.c has an ao2 container of linkedid
+ strings and uses the refcount of the string as a counter of how
+ many channels with the linkedid exist. Not only does this
+ eliminate the race condition, but it also allows us to look up
+ the linkedid by the hashed key instead of traversing the entire
+ channel list. Review: https://reviewboard.asterisk.org/r/1895/
+
+2012-05-01 23:11 +0000 [r364902] Richard Mudgett <rmudgett@digium.com>
+
+ * main/astobj2.c: Fixed __ao2_ref() validating user_data twice.
+ (closes issue ASTERISK-19755) Reported by: Gunther Kelleter
+ Patches: ao2_ref.patch (license #6372) patch uploaded by Gunther
+ Kelleter
+
+2012-05-01 23:08 +0000 [r364899] Mark Michelson <mmichelson@digium.com>
+
+ * funcs/func_volume.c: Fix Coverity-reported ARRAY_VS_SINGLETON
+ error. As it turned out, this wasn't a huge deal. We were calling
+ ast_app_parse_options() for a set of options of which none took
+ arguments. The proper thing to do for this case is to pass NULL
+ for the "args" parameter here. We were instead passing a
+ seemingly-randomly chosen char * from the function. While this
+ would never get written to, you can rest assured things would
+ have gotten bad had new options (which took arguments) been added
+ to func_volume. (closes issue ASTERISK-19656)
+
+2012-05-01 21:37 +0000 [r364841] Jason Parker <jparker@digium.com>
+
+ * main/manager.c: Prevent a potential crash when using manager
+ hooks. Found by me while poking at DPMA-127.
+
+2012-05-01 21:36 +0000 [r364840] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_local.c: * Fix error path resouce leak in
+ local_request(). * Restructure local_request() to reduce
+ indentation.
+
+2012-05-01 19:03 +0000 [r364786] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_confbridge.c: Play conf-placeintoconf message to the
+ correct channel Correct the code in app_confbridge to play the
+ conf-placeintoconf message to the marked user entering the bridge
+ instead of to the conference while the marked user hears silence.
+ (closes issue ASTERISK-19641) Reported-by: Mark A Walters
+
+2012-05-01 18:16 +0000 [r364769] Jonathan Rose <jrose@digium.com>
+
+ * main/app.c: Fix bad check in voicemail functions for
+ ast_inboxcount2_func Check looks for ast_inboxcount_func instead
+ of ast_inboxcount2_func on ast_inboxcount2_func calls. (closes
+ issue ASTERISK-19718) Reported by: Corey Farrell Patches:
+ ast_app_inboxcount2-null-refcheck.patch uploaded by Corey Farrell
+ (license 5909)
+
+2012-04-30 19:39 +0000 [r364706] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Revert improved identities sent in
+ dialog-info NOTIFY requests in r360862 Revision 360862 was
+ intended to improve identities sent in dialog-info NOTIFY
+ requests. Some users reported that hint became broken once this
+ was done. It's not clear exactly what part of the patch has
+ caused this regression, but broken hints are bad. For now, this
+ revision is being reverted so that the next releases of Asterisk
+ do not have bad behavior in them. The original reported issue
+ will have to be fixed differently in the next version of
+ Asterisk. (issue ASTERISK-16735)
+
+2012-04-30 16:37 +0000 [r364649] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323cDriver.c: Fix use freed pointer in return value
+ from call thread (issue ASTERISK-19663) Reported by: Matt Jordan
+ Patches: ASTERISK-19663-ooh323.patch (License #5415)
+
+2012-04-30 15:51 +0000 [r364635] Mark Murawki <markm@intellasoft.net>
+
+ * main/logger.c: Sanatize result from bfd_find_nearest_line
+ (BETTER_BACKTRACES) bfd_find_nearest_line can possibly set file
+ to null resulting in a crash when strrchr(file) runs (closes
+ issue ASTERISK-19815) Reported by Mark Murawski Tested by Mark
+ Murawski
+
+2012-04-29 19:31 +0000 [r364578] Matthew Jordan <mjordan@digium.com>
+
+ * formats/format_g723.c, formats/format_h263.c,
+ formats/format_h264.c, formats/format_sln16.c,
+ formats/format_wav_gsm.c, formats/format_siren14.c,
+ formats/format_gsm.c, formats/format_g719.c,
+ formats/format_siren7.c, formats/format_g729.c,
+ formats/format_ilbc.c, formats/format_sln.c,
+ formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c:
+ Fix error that caused truncate operations to fail Another very
+ inappropriate placement of a ')' (again introduced in r362151)
+ caused the various truncate operations to attempt to truncate the
+ sound file at a position of '0'. (issue ASTERISK-19655) Reported
+ by: Matt Jordan (issue ASTERISK-19810) Reported by: colbec
+
+2012-04-27 21:48 +0000 [r364341] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Don't attempt to make use of the
+ dynamic_exclude_static ACL if DNS lookup fails. (closes issue
+ ASTERISK-18321) Reported by Dan Lukes Patches:
+ ASTERISK-18321.patch by Mark Michelson (license #5049)
+
+2012-04-27 21:45 +0000 [r364340] Terry Wilson <twilson@digium.com>
+
+ * tests/test_config.c (added), main/config.c: Fix ast_parse_arg
+ numeric type range checking and add tests ast_parse_arg wasn't
+ checking for strto* parse errors or limiting the results by the
+ actual range of the numeric types. This patch fixes that and adds
+ unit tests as well. Review:
+ https://reviewboard.asterisk.org/r/1879/
+
+2012-04-27 19:26 +0000 [r364277] Matthew Jordan <mjordan@digium.com>
+
+ * include/asterisk/time.h: Prevent overflow in calculation in
+ ast_tvdiff_ms on 32-bit machines The method ast_tvdiff_ms
+ attempts to calculate the difference, in milliseconds, between
+ two timeval structs, and return the difference in a 64-bit
+ integer. Unfortunately, it assumes that the long tv_sec/tv_usec
+ members in the timeval struct are large enough to hold the
+ calculated values before it returns. On 64-bit machines, this
+ might be the case, as a long may be 64-bits. On 32-bit machines,
+ however, a long may be less (32-bits), in which case, the
+ calculation can overflow. This overflow caused significant
+ problems in MixMonitor, which uses the method to determine if an
+ audio factory, which has not presented audio to an audiohook, is
+ merely late in providing said audio or will never provide audio.
+ In an overflow situation, the audiohook would incorrectly
+ determine that an audio factory that will never provide audio is
+ merely late instead. This led to situations where a MixMonitor
+ never recorded any audio. Note that this happened most frequently
+ when that MixMonitor was started by the ConfBridge application
+ itself, or when the MixMonitor was attached to a Local channel.
+ (issue ASTERISK-19497) Reported by: Ben Klang Tested by: Ben
+ Klang Patches: 32-bit-time-overflow-10-2012-04-26.diff (license
+ #6283) by mjordan (closes issue ASTERISK-19727) Reported by: Mark
+ Murawski Tested by: Michael L. Young Patches:
+ 32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan)
+ (closes issue ASTERISK-19471) Reported by: feyfre Tested by:
+ feyfre (issue ASTERISK-19426) Reported by: Johan Wilfer Review:
+ https://reviewboard.asterisk.org/r/1889/
+
+2012-04-27 18:57 +0000 [r364258] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c: Allow SIP pvts involved in Replaces
+ transfers to fall out of reference sooner Unref the SIP pvt
+ stored in the refer structure as soon as it is no longer needed
+ so that the pvt and associated file descriptors can be freed
+ sooner. This change makes a reference decrement unnecessary in
+ code that handles SIP BYE/Also transfers which should not touch
+ the reference anyway. (related to issue ASTERISK-19579)
+
+2012-04-27 14:42 +0000 [r364203] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_sip.c: Allow for reloading SRTP crypto keys within
+ the same SIP dialog As a continuation of the patch in r356604,
+ which allowed for the reloading of SRTP keys in re-INVITE
+ transfer scenarios, this patch addresses the more common case
+ where a new key is requested within the context of a current SIP
+ dialog. This can occur, for example, when certain phones request
+ a SIP hold. Previously, once a dialog was associated with an SRTP
+ object, any subsequent attempt to process crypto keys in any SDP
+ offer - either the current one or a new offer in a new SIP
+ request - were ignored. This patch changes this behavior to only
+ ignore subsequent crypto keys within the current SDP offer, but
+ allows future SDP offers to change the keys. (issue
+ ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas
+ Arimont Review: https://reviewboard.asteriskorg/r/1885/
+
+2012-04-26 21:10 +0000 [r364060-364108] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_directed_pickup.c: Update Pickup application
+ documentation. (With feeling this time.)
+
+ * main/features.c: Fix DTMF atxfer running h exten after the wrong
+ bridge ends. When party B does an attended transfer of party A to
+ party C, the attending bridge between party B and C should not be
+ running an h exten when the bridge ends. Running an h exten now
+ sets a softhangup flag to ensure that an AGI will run in dead AGI
+ mode. * Set the AST_FLAG_BRIDGE_HANGUP_DONT on the party B
+ channel for the attending bridge between party B and C. (closes
+ issue AST-870) (closes issue ASTERISK-19717) Reported by: Mario
+ (closes issue ASTERISK-19633) Reported by: Andrey Solovyev
+ Patches: jira_asterisk_19633_v1.8.patch (license #5621) patch
+ uploaded by rmudgett Tested by: rmudgett, Andrey Solovyev, Mario
+
+2012-04-26 19:24 +0000 [r364046] Terry Wilson <twilson@digium.com>
+
+ * main/asterisk.c: Add more constness to the end_buf pointer in the
+ netconsole issue ASTERISK-18308 Review:
+ https://reviewboard.asterisk.org/r/1876/
+
+2012-04-26 13:24 +0000 [r363986] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c: Fix reference leaks involving SIP Replaces
+ transfers The reference held for SIP blind transfers using the
+ Replaces header in an INVITE was never freed on success and also
+ failed to be freed in some error conditions. This caused a file
+ descriptor leak since the RTP structures in use at the time of
+ the transfer were never freed. This reference leak and another
+ relating to subscriptions in the same code path have now been
+ corrected. (closes issue ASTERISK-19579)
+
+2012-04-26 09:44 +0000 [r363934] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/chan_sip.c: chan_sip: [general] maxforwards, not checked
+ for a value greater than 255 The peer maxforwards is checked for
+ both '< 1' and '> 255', but the default 'maxforwards' in the
+ [general] section is only checked for '< 1' alecdavis (license
+ 585) Reported by: alecdavis Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1888/
+
+2012-04-26 03:11 +0000 [r363375-363875] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_directed_pickup.c: Update Pickup application
+ documentation. (Even better)
+
+ * apps/app_directed_pickup.c: Update Pickup application
+ documentation.
+
+ * channels/sig_pri.c, channels/chan_dahdi.c: Make
+ DAHDISendCallreroutingFacility wait 5 seconds for a reply before
+ disconnecting the call. Some switches may not handle the
+ call-deflection/call-rerouting message if the call is
+ disconnected too soon after being sent. Asteisk was not waiting
+ for any reply before disconnecting the call. * Added a 5 second
+ delay before disconnecting the call to wait for a potential
+ response if the peer does not disconnect first. (closes issue
+ ASTERISK-19708) Reported by: mehdi Shirazi Patches:
+ jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: rmudgett
+
+ * channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c:
+ Clear ISDN channel resetting state if the peer continues to use
+ it. Some ISDN switches occasionally fail to send a RESTART
+ ACKNOWLEDGE in response to a RESTART request. * Made the second
+ SETUP received after sending a RESTART request clear the channel
+ resetting state as if the peer had sent the expected RESTART
+ ACKNOWLEDGE before continuing to process the SETUP. The peer may
+ not be sending the expected RESTART ACKNOWLEDGE. (issue
+ ASTERISK-19608) (issue AST-844) (issue AST-815) Patches:
+ jira_ast_815_v1.8.patch (license #5621) patch uploaded by
+ rmudgett (modified)
+
+ * main/features.c: Fix recalled party B feature flags for a failed
+ DTMF atxfer. 1) B calls A with Dial option T 2) B DTMF atxfer to
+ C 3) B hangs up 4) C does not answer 5) B is called back 6) B
+ answers 7) B cannot initiate transfers anymore * Add dial
+ features datastore to recalled party B channel that is a copy of
+ the original party B channel's dial features datastore. *
+ Extracted add_features_datastore() from
+ add_features_datastores(). * Renamed struct ast_dial_features
+ features_caller and features_callee members to my_features and
+ peer_features respectively. These better names eliminate the need
+ for some explanatory comments. * Simplified code accessing the
+ struct ast_dial_features datastore. (closes issue ASTERISK-19383)
+ Reported by: lgfsantos
+
+ * main/features.c: Hangup affected channel in error paths of
+ bridge_call_thread().
+
+2012-04-23 16:02 +0000 [r363209] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/astfd.c: On some platforms, O_RDONLY is not a flag to be
+ checked, but merely the absence of O_RDWR and O_WRONLY. The POSIX
+ specification does not mandate how these 3 flags must be
+ specified, only that one of the three must be specified in every
+ call.
+
+2012-04-23 14:33 +0000 [r363141] Jonathan Rose <jrose@digium.com>
+
+ * main/manager.c, /: AST-2012-004: Fix an error that allows AMI
+ users to run shell commands sans authorization. As detailed in
+ the advisory, AMI users without write authorization for SYSTEM
+ class AMI actions were able to run system commands by going
+ through other AMI commands which did not require that
+ authorization. Specifically, GetVar and Status allowed users to
+ do this by setting their variable/s options to the SHELL or EVAL
+ functions. Also, within 1.8, 10, and trunk there was a similar
+ flaw with the Originate action that allowed users with originate
+ permission to run MixMonitor and supply a shell command in the
+ Data argument. That flaw is fixed in those versions of this
+ patch. (closes issue ASTERISK-17465) Reported By: David Woolley
+ Patches: 162_ami_readfunc_security_r2.diff uploaded by jrose
+ (license 6182) 18_ami_readfunc_security_r2.diff uploaded by jrose
+ (license 6182) 10_ami_readfunc_security_r2.diff uploaded by jrose
+ (license 6182) ........ Merged revisions 363117 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2
+
+2012-04-23 14:05 +0000 [r363102-363106] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_sip.c: AST-2012-006: Fix crash in UPDATE handling
+ when no channel owner exists If Asterisk receives a SIP UPDATE
+ request after a call has been terminated and the channel has been
+ destroyed but before the SIP dialog has been destroyed, a
+ condition exists where a connected line update would be attempted
+ on a non-existing channel. This would cause Asterisk to crash.
+ The patch resolves this by first ensuring that the SIP dialog has
+ an owning channel before attempting a connected line update. If
+ an UPDATE request is received and no channel is associated with
+ the dialog, a 481 response is sent. (closes issue ASTERISK-19770)
+ Reported by: Thomas Arimont Tested by: Matt Jordan Patches:
+ ASTERISK-19278-2012-04-16.diff uploaded by Matt Jordan (license
+ 6283)
+
+ * /, channels/chan_skinny.c: AST-2012-005: Fix remotely exploitable
+ heap overflow in keypad button handling When handling a keypad
+ button message event, the received digit is placed into a fixed
+ length buffer that acts as a queue. When a new message event is
+ received, the length of that buffer is not checked before placing
+ the new digit on the end of the queue. The situation exists where
+ sufficient keypad button message events would occur that would
+ cause the buffer to be overrun. This patch explicitly checks that
+ there is sufficient room in the buffer before appending a new
+ digit. (closes issue ASTERISK-19592) Reported by: Russell Bryant
+ ........ Merged revisions 363100 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2
+
+2012-04-21 01:44 +0000 [r362997] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c: Update app_dial M and U option GOTO return value
+ documentation.
+
+2012-04-20 16:09 +0000 [r362815-362868] Terry Wilson <twilson@digium.com>
+
+ * main/asterisk.c: OpenBSD doesn't have rawmemchr, use strchr
+ (closes issue ASTERISK-19758) Reported by: Barry Miller Tested
+ by: Terry Wilson Patches: 362758-diff uploaded by Barry Miller
+ (license 5434)
+
+ * apps/app_speech_utils.c: Document Speech* apps hangup on failure
+ and suggest TryExec The Speech API apps return -1 on failure,
+ which will hang up the channel. This may not be desirable
+ behavior for some, but it isn't something that can be changed
+ without breaking people's dialplans or writing an option to all
+ of the Speech apps that does what TryExec already does. This
+ patch documents the hangup behavior of the apps, and suggests
+ TryExec as the solution. (closes issue AST-813)
+
+2012-04-19 21:58 +0000 [r362729] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * funcs/func_version.c: Fix documentation for
+ ${VERSION(ASTERISK_VERSION_NUM)}.
+
+2012-04-19 21:05 +0000 [r362680] Michael L. Young <elgueromexicano@gmail.com>
+
+ * tests/test_linkedlists.c, tests/test_poll.c: Add leading and
+ trailing backslashes A couple of unit tests did not have have
+ leading or trailing backslashes when setting their test category
+ resulting in a warning message being displayed. Added the
+ backslash where needed.
+
+2012-04-19 20:59 +0000 [r362677] Richard Mudgett <rmudgett@digium.com>
+
+ * configs/queues.conf.sample: Update membermacro and membergosub
+ documentation in queues.conf.sample.
+
+2012-04-19 15:53 +0000 [r362586] Sean Bright <sean@malleable.com>
+
+ * apps/app_externalivr.c: Prevent a crash in ExternalIVR when the
+ 'S' command is sent first. If the first command sent from an
+ ExternalIVR client is an 'S' command, we were blindly removing
+ the first element from the play list and deferencing it, even if
+ it was NULL. This corrects that and also locks appropriately in
+ one place. (issue ASTERISK-17889) Reported by: Chris Maciejewski
+
+2012-04-19 14:26 +0000 [r362536] Terry Wilson <twilson@digium.com>
+
+ * main/asterisk.c: Handle multiple commands per connection via
+ netconsole Asterisk would accept multiple NULL-delimited CLI
+ commands via the netconsole socket, but would occasionally miss a
+ command due to the command not being completely read into the
+ buffer. This patch ensures that any partial commands get moved to
+ the front of the read buffer, appended to, and properly sent.
+ (closes issue ASTERISK-18308) Review:
+ https://reviewboard.asterisk.org/r/1876/
+
+2012-04-19 02:08 +0000 [r362485] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_sms.c, main/stdtime/localtime.c, utils/extconf.c,
+ addons/chan_mobile.c, main/asterisk.c, channels/chan_unistim.c,
+ main/frame.c, main/tdd.c, main/jitterbuf.c: Fix a variety of
+ potential buffer overflows * chan_mobile: Fixed an overrun where
+ the cind_state buffer (an integer array of size 16) would be
+ overrun due to improper bounds checking. At worst, the buffer can
+ be overrun by a total of 48 bytes (assuming 4-byte integers),
+ which would still leave it within the allocated memory of struct
+ hfp. This would corrupt other elements in that struct but not
+ necessarily cause any further issues. * app_sms: The array imsg
+ is of size 250, while the array (ud) that the data is copied into
+ is of size 160. If the size of the inbound message is greater
+ then 160, up to 90 bytes could be overrun in ud. This would
+ corrupt the user data header (array udh) adjacent to ud. *
+ chan_unistim: A number of invalid memmoves are corrected. These
+ would move data (which may or may not be valid) into the ends of
+ these buffers. * asterisk: ast_console_toggle_loglevel does not
+ check that the console log level being set is less then or equal
+ to the allowed log levels of 32. * frame: In
+ ast_codec_pref_prepend, if any occurrence of the specified codec
+ is not found, the value used to index into the array pref->order
+ would be one greater then the maximum size of the array. *
+ jitterbuf: If the element being placed into the jitter buffer
+ lands in the last available slot in the jitter history buffer,
+ the insertion sort attempts to move the last entry in the buffer
+ into one slot past the maximum length of the buffer. Note that
+ this occurred for both the min and max jitter history buffers. *
+ tdd: If a read from fsk_serial returns a character that is
+ greater then 32, an attempt to read past one of the statically
+ defined arrays containing the values that character maps to would
+ occur. * localtime: struct ast_time and tm are not the same size
+ - ast_time is larger, although it contains the elements of tm
+ within it in the same layout. Hence, when using memcpy to copy
+ the contents of tm into ast_time, the size of tm should be used,
+ as opposed to the size of ast_time. * extconf: this treats
+ ast_timing's minmask array as if it had a length of 48, when it
+ has defined the size of the array as 24. pbx.h defines minmask as
+ having a size of 48. (issue ASTERISK-19668) Reported by: Matt
+ Jordan
+
+2012-04-18 16:20 +0000 [r362428] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample: Add ability to ignore layer 1
+ alarms for BRI PTMP lines. Several telcos bring the BRI PTMP
+ layer 1 down when the line is idle. When layer 1 goes down,
+ Asterisk cannot make outgoing calls. Incoming calls could fail as
+ well because the alarm processing is handled by a different code
+ path than the Q.931 messages. * Add the layer1_presence
+ configuration option to ignore layer 1 alarms when the telco
+ brings layer 1 down. This option can be configured by span while
+ the similar DAHDI driver teignorered=1 option is system wide.
+ This option unlike layer2_persistence does not require libpri
+ v1.4.13 or newer. Related to JIRA AST-598 JIRA ABE-2845
+
+2012-04-17 21:18 +0000 [r362355-362368] Matthew Jordan <mjordan@digium.com>
+
+ * main/frame.c: Handle case where an unknown format is used to get
+ the preferred codec size In ast_codec_pref_getsize, if an unknown
+ format is passed to the method, no preferred codec will be
+ selected and a negative number will be used to index into the
+ format list. The method now logs an unknown format as a warning,
+ and returns an empty format list. (issue ASTERISK-19655) Reported
+ by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/
+
+ * res/res_musiconhold.c, res/res_rtp_asterisk.c, res/res_agi.c: Fix
+ places in resources where a negative return value could impact
+ execution This patch addresses a number of modules in resources
+ that did not handle the negative return value from function calls
+ adequately. This includes: * res_agi.c: if the result of the read
+ function is a negative number, indicating some failure, the
+ result would instead be treated as the number of bytes read. This
+ patch now treats negative results in the same manner as an end of
+ file condition, with the exception that it also logs the error
+ code indicated by the return. * res_musiconhold.c: if spawn_mp3
+ fails to assign a file descriptor to srcfd, and instead assigns a
+ negative value, that file descriptor could later be passed to
+ functions that require a valid file descriptor. If spawn_mp3
+ fails, we now immediately retry instead of continuing in the
+ logic. * res_rtp_asterisk.c: if no codec can be matched between
+ two RTP instances in a peer to peer bridge, we immediately return
+ instead of attempting to use the codec payload type as an index
+ to determine the appropriate negotiated codec. (issue
+ ASTERISK-19655) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1863/
+
+ * main/asterisk.c, main/manager.c, main/translate.c: Fix places in
+ main where a negative return value could impact execution This
+ patch addresses a number of modules in main that did not handle
+ the negative return value from function calls adequately, or were
+ not sufficiently clear that the conditions leading to improper
+ handling of the return values could not occur. This includes: *
+ asterisk.c: A negative return value from the read function would
+ be used directly as an index into a buffer. We now check for
+ success of the read function prior to using its result as an
+ index. * manager.c: Check for failures in mkstemp and lseek when
+ handling the temporary file created for processing data returned
+ from a CLI command in action_command. Also check that the result
+ of an lseek is sanitized prior to using it as the size of a
+ memory map to allocate. * translate.c: Note in the appropriate
+ locations where powerof cannot return a negative value, due to
+ proper checks placed on the inputs to that function. (issue
+ ASTERISK-19655) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1863/
+
+ * funcs/func_env.c: Fix places where a negative return from ftello
+ could be used as invalid input In a variety of locations in both
+ reading and writing a file, the result from the C library
+ function ftello is used as input to other functions. For the
+ parameters and functions in question, a negative value is invalid
+ input. This patch checks the return value from the ftello
+ function to determine if we were able to determine the current
+ position in the file stream and, if not, fail gracefully. (issue
+ ASTERISK-19655) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1863/
+
+2012-04-17 20:43 +0000 [r362354] Jonathan Rose <jrose@digium.com>
+
+ * main/utils.c, res/res_config_curl.c, res/res_config_pgsql.c,
+ res/res_config_odbc.c: Make use of va_args more appropriate to
+ form in various res_config modules plus utils. A number of
+ va_copy operations weren't matched with a corresponding va_end in
+ res_config_odbc. Also, there was a potential for va_end to be
+ invoked twice on the same va_arg in utils, which would mean
+ invoking va_end on an undefined variable... which is bad. va_end
+ is removed from various functions in config_pgsql and config_curl
+ since they aren't making their own copy. The invokers of those
+ functions are responsible for calling va_end on them. (issue
+ ASTERISK-19451) Reported by: Walter Doekes Review:
+ https://reviewboard.asterisk.org/r/1848/
+
+2012-04-17 18:25 +0000 [r362304] Matthew Jordan <mjordan@digium.com>
+
+ * formats/format_sln16.c, formats/format_wav_gsm.c,
+ formats/format_siren14.c, formats/format_gsm.c,
+ formats/format_g719.c, formats/format_siren7.c,
+ formats/format_sln.c, formats/format_vox.c, formats/format_wav.c,
+ formats/format_pcm.c: Fix error that caused seek format
+ operations to set max file size to '1' or '0' A very
+ inappropriate placement of a ')' (introduced in r362151) caused
+ the maximum size of a file to be set as the result of a
+ comparison operation, as opposed to the result of the ftello
+ operation. This resulted in seeking being restricted to the
+ beginning of the file, or 1 byte into the file. Thanks to the
+ Asterisk Test Suite for properly freaking out about this on at
+ least one test. (issue ASTERISK-19655) Reported by: Matt Jordan
+
+2012-04-17 02:37 +0000 [r362253] Michael L. Young <elgueromexicano@gmail.com>
+
+ * channels/chan_sip.c: Turn off warning message when bind address
+ is set to any. When a bind address is set to an ANY address
+ (udpbindport=::), a warning message is displayed stating that
+ "Address remapping activated in sip.conf but we're using IPv6,
+ which doesn't need it. Please remove 'localnet' and/or
+ 'externaddr' settings." But if one is running dual stack, we
+ shouldn't be told to turn those settings off. This patch checks
+ if the bind address is an ANY address or not. The warning message
+ will now only be displayed if the bind address is NOT an ANY
+ address and IPv6 is being used. Also, updated the copyright year.
+ (closes issue ASTERISK-19456) Reported by: Michael L. Young
+ Tested by: Michael L. Young Patches: chan_sip_ipv6_message.diff
+ uploaded by Michael L. Young (license 5026)
+
+2012-04-16 21:56 +0000 [r362151-362204] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_dahdi.c, channels/chan_agent.c: Fix negative return
+ handling in channel drivers In chan_agent, while handling a
+ channel indicate, the agent channel driver must obtain a lock on
+ both the agent channel, as well as the channel the agent channel
+ is using. To do so, it attempts to lock the other channel first,
+ then unlock the agent channel which is locked prior to entry into
+ the indicate handler. If this unlock fails with a negative return
+ value, which can occur if the object passed to agent_indicate is
+ an invalid ao2 object or is NULL, the return value is passed
+ directly to strerror, which can only accept positive integer
+ values. In chan_dahdi, the return value of dahdi_get_index is
+ used to directly index into the sub-channel array. If
+ dahd_get_index returns a negative value, it would use that value
+ to index into the array, which could cause an invalid memory
+ access. If dahdi_get_index returns a negative number, we now
+ default to SUB_REAL. (issue ASTERISK-19655) Reported by: Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/1863/
+
+ * apps/app_voicemail.c: Fix handling of negative return code when
+ storing voicemails in ODBC storage When storing a voicemail
+ message using an ODBC connection to a database, the voicemail
+ message is first stored on disk. The sound file associated with
+ the message is read into memory before being transmitted to the
+ database. When this occurs, a failure in the C library's lseek
+ function would cause a negative value to be passed to the mmap as
+ the size of the memory map to create. This would almost certainly
+ cause the creation of the memory map to fail, resulting in the
+ message being lost. (issue ASTERISK-19655) Reported by: Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/1863
+
+ * formats/format_g723.c, formats/format_h263.c,
+ formats/format_h264.c, formats/format_sln16.c,
+ formats/format_wav_gsm.c, formats/format_siren14.c,
+ formats/format_gsm.c, formats/format_g719.c,
+ formats/format_siren7.c, formats/format_g729.c,
+ formats/format_ilbc.c, formats/format_sln.c,
+ formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c:
+ Check for IO stream failures in various format's truncate/seek
+ operations For the formats that support seek and/or truncate
+ operations, many of the C library calls used to determine or set
+ the current position indicator in the file stream were not being
+ checked. In some situations, if an error occurred, a negative
+ value would be returned from the library call. This could then be
+ interpreted inappropriately as positional data. This patch checks
+ the return values from these library calls before using them in
+ subsequent operations. (issue ASTERISK-19655) Reported by: Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/1863/
+
+2012-04-13 15:54 +0000 [r362079-362082] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_forkcdr.c: Make ForkCDR e option not set end time of the
+ newly forked CDR log Prior to this patch, ForkCDR's e option
+ would immediately set the end time of the forked CDR to that of
+ the CDR that is being terminated. This resulted in the new CDR's
+ end time being roughly the same as it's beginning time (which is
+ in turn roughly the same as the original's end time). (closes
+ issue ASTERISK-19164) Reported by: Steve Davies Patches:
+ cdr_fork_end.v10.patch uploaded by Steve Davies (license 5012)
+
+ * apps/app_meetme.c: Send relative path named recordings to the
+ meetme directory instead of sounds Prior to this patch, no effort
+ was made to parse the path name to determine a proper destination
+ for recordings of MeetMe's r option. This fixes that. Review:
+ https://reviewboard.asterisk.org/r/1846/
+
+2012-04-12 16:18 +0000 [r361955-361972] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_iax2.c: Make trunkfreq take effect when set
+ Previously, setting trunkfreq had no effect on initial load or on
+ reload and only ever used the default value. This causes
+ trunkfreq to be used appropriately on initial load and reload.
+ (closes issue ASTERISK-19521) Patch-by: Jaco Kroon
+
+ * Makefile.rules, makeopts.in, codecs/lpc10/Makefile, Makefile,
+ build_tools/cflags.xml, build_tools/menuselect-deps.in,
+ codecs/gsm/src/k6opt.s, configure, codecs/gsm/Makefile,
+ configure.ac: Simplify build system architecture optimization
+ This change to the build system rips out any usage of PROC along
+ with architecture-specific optimizations in favor of using
+ -march=native where it is supported. This fixes broken builds on
+ 64bit Intel systems and results in better optimized code on
+ systems running GCC 4.2+. Review:
+ https://reviewboard.asterisk.org/r/1852/ (closes issue
+ ASTERISK-19462)
+
+2012-04-10 21:43 +0000 [r361854] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Prevent invalid access of free'd memory if
+ DAHDI channel during an MWI event In the MWI processing loop,
+ when a valid event occurs the temporary caller ID information is
+ deallocated. If a new DAHDI channel is successfully created, the
+ event is passed up to the analog_ss_thread without error and the
+ loop exits. If, however, the DAHDI channel is not created, then
+ the caller ID struct has been free'd, and the gains reset to
+ their previous level. This will almost certainly cause an invalid
+ access to the free'd memory, either in subsequent calls to
+ callerid_free or calls to callerid_feed. * Rework the -r361705
+ patch to better manage the cs and mtd allocated resources. *
+ Fixed use of mwimonitoractive flag to be correct if the
+ mwi_thread() fails to start.
+
+2012-04-10 19:57 +0000 [r361657-361803] Matthew Jordan <mjordan@digium.com>
+
+ * main/http.c: Fix crash caused by unloading or reloading of
+ res_http_post When unlinking itself from the registered HTTP
+ URIs, res_http_post could inadvertently free all URIs registered
+ with the HTTP server. This patch modifies the unregister method
+ to only free the URI that is actually being unregistered, as
+ opposed to all of them.
+
+ * funcs/func_curl.c: Allow func_curl to exit gracefully if list
+ allocation fails during write If the global_curl_info data
+ structure could not be allocated, the datastore associated with
+ the operation would be free'd, but the function would not return.
+ This would later dereference the datastore, almost certainly
+ causing Asterisk to crash. With this patch, if the data structure
+ is not allocated the method will return an error code, and not
+ attempt any further operation.
+
+ * channels/chan_dahdi.c: Prevent invalid access of free'd memory if
+ DAHDI channel during an MWI event In the MWI processing loop,
+ when a valid event occurs the temporary caller ID information is
+ deallocated. If a new DAHDI channel is successfully created, the
+ event is passed up to the analog_ss_thread without error and the
+ loop exits. If, however, the DAHDI channel is not created, then
+ the caller ID struct has been free'd, and the gains reset to
+ their previous level. This will almost certainly cause an invalid
+ access to the free'd memory, either in subsequent calls to
+ callerid_free or calls to callerid_feed. This patch makes it so
+ that we only free the caller ID structure if a DAHDI channel is
+ successfully created, and we bump the gains back up if we fail to
+ make a DAHDI channel.
+
+ * funcs/func_global.c: Change SHARED function to use a safe
+ traversal when modifying a variable When the SHARED function
+ modifies a variable, it removes it from its list of variables and
+ reinserts the new value at the head of the list of variables.
+ Doing this inside a standard list traversal can be dangerous, as
+ the standard list traversal does not account for the list being
+ changed. While the code in question should not cause a use after
+ free violation due to its breaking out of the loop after freeing
+ the variable, it could lead to a maintenance issue if the loop
+ was modified. This also fixes a violation reported by a static
+ analysis tool, which also makes this code easier to maintain in
+ the future.
+
+2012-04-06 21:50 +0000 [r361558-361606] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_calendar_ews.c: Fix memory leak in res_calendar_ews when
+ event email address node is empty If the XML calendar data
+ returned by a Microsoft Exchange Web Service specifies an XML
+ Event E-Mail Address ("EmailAddress"), and no e-mail address is
+ provided, a condition existed where an ast_calendar_attendee
+ struct would be allocated but not appended to the list of
+ attendees. Because of that, the memory associated with the
+ attendee would never be freed. This patch frees the memory if no
+ e-mail address is provided.
+
+ * apps/app_meetme.c: Fix memory leak when using MeetMeAdmin 'e'
+ option with user specified A memory leak/reference counting leak
+ occurs if the MeetMeAdmin 'e' command (eject last user that
+ joined) is used in conjunction with a specified user. Regardless
+ of the command being executed, if a user is specified for the
+ command, MeetMeAdmin will look up that user. Because the 'e'
+ option kicks the last user that joined, as opposed to the one
+ specified, the reference to the user specified by the command
+ would be leaked when the user variable was assigned to the last
+ user that joined.
+
+2012-04-06 18:09 +0000 [r361471] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_ices.c, channels/chan_gtalk.c, channels/chan_iax2.c,
+ res/res_config_sqlite.c, res/res_srtp.c, main/cdr.c,
+ main/tcptls.c, funcs/func_channel.c, channels/console_gui.c,
+ apps/app_sms.c, apps/app_chanspy.c, addons/chan_mobile.c,
+ channels/chan_mgcp.c, main/xmldoc.c, apps/app_voicemail.c,
+ res/res_clioriginate.c, channels/chan_unistim.c, main/pbx.c,
+ channels/chan_sip.c, res/res_fax.c, funcs/func_strings.c,
+ channels/console_video.c, formats/format_ogg_vorbis.c: Add
+ missing newlines to CLI logging
+
+2012-04-06 16:27 +0000 [r361403-361412] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * funcs/func_sysinfo.c: Fix typo in svn:keywords
+
+ * bridges/bridge_multiplexed.c, bridges/bridge_builtin_features.c:
+ Fix typo in svn:keywords
+
+2012-04-06 15:47 +0000 [r361380] Russell Bryant <russell@russellbryant.com>
+
+ * apps/rpt_flow.pdf (removed), configs/rpt.conf.sample (removed),
+ configs/usbradio.conf.sample (removed): Remove a few more files
+ related to chan_usbradio and app_rpt.
+
+2012-04-06 14:01 +0000 [r361332] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_sip.c: Fix a typo in the warning messages for an
+ ignored media stream Added a '\n' to the warning messages when we
+ ignore a media stream due to the port number being '0'. (closes
+ issue ASTERISK-19646) Reported by: Badalian Vyacheslav
+
+2012-04-06 13:30 +0000 [r361329] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_dial.c: Remove unnecessary error message in app_dial.c
+ The error message for failure to stop autoservice after a gosub
+ or macro call during a dial was removed for macro while Asterisk
+ 1.4 was still being actively developed. The corresponding gosub
+ error message was never removed. (closes issue ASTERISK-19551)
+
+2012-04-05 16:36 +0000 [r361201-361269] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_meetme.c: Fix MusicOnHold in MeetMe so that it always
+ uses the class if it's been defined There were a few instances of
+ restarting music on hold in meetme that would cause Asterisk to
+ revert to the default class of music on hold for no adequate
+ reason. Review: https://reviewboard.asterisk.org/r/1844/
+
+ * addons/ooh323cDriver.c: Fix some stuff involving calls to memcpy
+ and memset The important parts of the patch were already applied
+ through other updates. (closes issue ASTERISK-19445) Reported by:
+ Makoto Dei Patches: memset-memcpy-length.patch uploaded by Makoto
+ Dei (license 5027)
+
+ * funcs/func_devstate.c: Make 'help devstate change' display
+ properly (get rid of excess comma) (closes issue ASTERISK-19444)
+ Reported by: Makoto Dei Patches:
+ devstate-change-usage-truncate.patch uploaded by Makoto Dei
+ (license 5027)
+
+2012-05-02 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.12.0 Released.
+
+2012-05-01 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.12.0-rc3 Released.
+
+ * channels/chan_sip.c: Revert revision 360862
+
+ Revision 360862 was intended to improve identities sent in
+ dialog-info NOTIFY requests. Some users reported that hint became
+ broken once this was done. It's not clear exactly what part of
+ the patch has caused this regression, but broken hints are bad.
+
+ For now, this revision is being reverted so that the next releases of
+ Asterisk do not have bad behavior in them. The original reported
+ issue will have to be fixed differently in the next version of
+ Asterisk.
+
+ (issue ASTERISK-16735)
+
+2012-04-24 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.12.0-rc2 Released.
+
+ * AST-2012-004
+
+ * AST-2012-005
+
+ * AST-2012-006
+
+2012-04-04 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.12.0-rc1 Released.
+
+2012-04-04 16:29 +0000 [r361090-361142] Jonathan Rose <jrose@digium.com>
+
+ * main/app.c, pbx/pbx_realtime.c, apps/app_externalivr.c,
+ channels/chan_iax2.c, apps/app_milliwatt.c, main/channel.c,
+ pbx/pbx_loopback.c, addons/chan_ooh323.c, channels/chan_sip.c:
+ Replace GNU old-style field designator extensions to fix clang
+ warnings (issue ASTERISK-19540) Reported by: Makoto Dei Patches:
+ clang-gnu-designator.patch uploaded by Makoto Dei (license 5027)
+
+ * apps/app_meetme.c: Make the MeetMeAdmin N command (mute all
+ nonadmins) not mute admins (Closes Issue ASTERISK-19335) Reported
+ by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1843/
+
+2012-04-03 20:08 +0000 [r360987-361040] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_transfer.c: Fix the display of documentation for
+ Transfer This came up while fixing documentation generation for
+ many other cases where the argument separator was not being
+ displayed properly. Now that it is displayed properly, it shows
+ up in the wrong place for Transfer since the '/' is only required
+ if Tech is present. (related to issue ASTERISK-18168)
+
+ * channels/chan_sip.c: Stop sending out RTCP if RTP is inactive
+ This change prevents Asterisk from sending RTCP receiver reports
+ during a remote bridge since it is no longer receiving media and
+ should not be reporting anything. (related to ASTERISK-19366)
+
+2012-03-30 21:26 +0000 [r360933] Richard Mudgett <rmudgett@digium.com>
+
+ * main/logger.c: Fix logger deadlock on Asterisk shutdown. The
+ logger_thread() had an exit path that failed to release the
+ logmsgs list lock. * Make logger_thread() exit path unlock the
+ logmsgs list lock. * Made ast_log() not queue any messages to the
+ logmsgs list if the close_logger_thread flag is set. (issue
+ ASTERISK-19463) Reported by: Matt Jordan
+
+2012-03-29 23:32 +0000 [r360862-360884] Mark Michelson <mmichelson@digium.com>
+
+ * main/features.c: Fix potential race condition during call pickup.
+ Prior to this patch, a connected line update was queued during
+ call pickup and then an answer frame was queued. The original
+ caller would presumably then have his connected line updated and
+ then the call would be answered. In actuality, the answer frame
+ was not how the call ended up being answered. Rather, an odd
+ section in app_dial that checks if the called channel's state is
+ up. The result is that the order of the connected line update and
+ the answer were variable. In most cases, this wasn't actually a
+ bad thing. However, if the 'I' option was passed to dial, the
+ connected line update would be inhibited. The fix is to queued
+ the connected line after the answer frame is queued. This way the
+ race in app_dial is between two conditions resulting in an
+ answer. This way the connected line update occurs after the
+ answer every time. (closes issue ASTERISK-19183) Reported by:
+ Thomas Arimont Tested by: Thomas Arimont Mark Michelson Patches:
+ ASTERISK-19183.patch uploaded by Mark Michelson (license 5049)
+
+ * channels/chan_sip.c: Improve accuracy of identifying information
+ sent in dialog-info SIP NOTIFY requests. This change makes use of
+ connected party information in addition to caller ID in order to
+ populate local and remote XML elements in the dialog-info
+ NOTIFYs. (closes issue ASTERISK-16735) Reported by: Maciej
+ Krajewski Tested by: Maciej Krajewski Patches:
+ local_remote_hint2.diff uploaded by Mark Michelson (license 5049)
+
+2012-03-28 19:06 +0000 [r360712] Terry Wilson <twilson@digium.com>
+
+ * cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c,
+ channels/chan_gtalk.c, channels/chan_jingle.c,
+ addons/chan_ooh323.c: Destroy configs when they are no longer
+ used https://reviewboard.asterisk.org/r/1834/
+
+2012-03-27 16:59 +0000 [r360625] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Make a debug message regarding subscription
+ changes more accurate. I was getting confused during some testing
+ why Asterisk was saying that a subscription was being added when
+ it was clearly being removed. This fixes that confusion.
+
+2012-03-27 14:32 +0000 [r360488-360574] Jonathan Rose <jrose@digium.com>
+
+ * configure: Updates config with bootstrap where I changed
+ configure.ac in r360488 (issue ASTERISK-17842) Reported by: Bryon
+ Clark
+
+ * configure.ac: Fix BETTER_BACKTRACES library detection for
+ Fedora/RedHat/CentOS (closes ASTERISK-17842) Reported by: Bryon
+ Clark Patches: 20110512__issue19278.diff.txt uploaded by Tilghman
+ Lesher (license 5003) configure_bfd_with_dl_and_iberty.patch
+ uploaded by Bryon Clark (license 6157)
+
+2012-03-26 18:37 +0000 [r360471-360474] Paul Belanger <pabelanger@digium.com>
+
+ * CHANGES: Update CHANGES for r360471
+
+ * CHANGES: Fix Asterisk version typo
+
+ * main/dnsmgr.c: Increase verbosity level for ast_verb messages
+ While this does not fix the issue of the CLI being flooded by
+ 'doing dnsmgr_lookup' messages, increasing the verbosity level
+ above 5 should help minimize it.
+
+2012-03-24 23:46 +0000 [r360356-360413] Russell Bryant <russell@russellbryant.com>
+
+ * funcs/func_curl.c: func_curl: Fix leak of an ast_str in error
+ handling code path.
+
+ * apps/app_page.c: app_page: Fix a memory leak on every Page().
+ dial_list is a dynamically allocated array that is allocated at
+ the beginning of Page() based on how many devices will be dialed.
+ This was never being freed.
+
+ * apps/app_jack.c: app_jack: fix datastore memory leak in error
+ handling path.
+
+ * res/ael/ael.tab.h, main/ast_expr2.c, main/ast_expr2.h,
+ res/ael/ael.tab.c, main/ast_expr2f.c, res/ael/ael_lex.c: Rebuild
+ parsers. This is needed to include the last fix to
+ main/ast_expr2.y. The changes look much bigger as this
+ regeneration of the code was done with newer versions of flex and
+ bison.
+
+ * main/ast_expr2.y: expression parser: Fix (theoretical) memory
+ leak. Fix a memory leak that is very unlikely to actually happen.
+ If a malloc() succeeded, but the following strdup() failed, the
+ memory from the original malloc() would be leaked.
+
+2012-03-24 00:35 +0000 [r360262-360309] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c, main/channel.c: Make number not available
+ presentation also set screening to network provided. Q.951
+ indicates that when the presentation indicator is "Number not
+ available due to interworking" for a number then the screening
+ indicator field should be "Network provided". * Made
+ ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE
+ when the presentation is "Number not available due to
+ interworking". This fix makes Asterisk consistent and it also
+ makes it consistent with earlier branches as far as this
+ presentation value is concerned. * Made pri_to_ast_presentation()
+ and ast_to_pri_presentation() conversions handle the "Number not
+ available due to interworking" case better in sig_pri.c. This
+ change is possible because the minimum required libpri version
+ (v1.4.11) has the necessary defines in libpri.h.
+
+ * channels/chan_sip.c: Add missing initialization of
+ update_redirecting in chan_sip.c
+
+2012-03-21 14:51 +0000 [r360138] Jonathan Rose <jrose@digium.com>
+
+ * contrib/scripts/install_prereq: Update install_prereq script to
+ include missing GSM library for debian amd move SQLite3. (closes
+ issue ASTERISK-19367) Reported by: Andrew Latham Patches:
+ debian_install_prereq.diff uploaded by Andrew Latham (license
+ 5985)
+
+2012-03-21 13:19 +0000 [r360087] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * configure, configure.ac: Also detect gmime 2.6 Also detect gmime
+ version 2.6 (Michael Biebl) Signed-off-by: Tzafrir Cohen (License
+ #5035) <tzafrir.cohen@xorcom.com>
+
+2012-03-21 13:19 +0000 [r360086] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_sip.c: Ensure Asterisk sends a BYE when pending on
+ the final response to a re-INVITE When Asterisk detects a hangup
+ and cannot send a BYE due to a pending INVITE, it sets the
+ pendingbye flag and waits for the final response to that INVITE.
+ When the response is received, it transmits the BYE. If, however,
+ that INVITE request is a pending re-INVITE, it needs to first
+ send a CANCEL request to terminate the pending re-INVITE. In that
+ circumstance, Asterisk was, in some scenarios, clearing the
+ pendingbye flag after processing the CANCEL request and not
+ checking for a pending BYE when receiving the final 487 response
+ to the INVITE. This patch ensures that if the pendingbye flag is
+ set, it is honored regardless of the nature of the INVITE request
+ currently in flight. (closes issue ASTERISK-19365) Reported by:
+ Thomas Arimont Tested by: Thomas Arimont Patches:
+ bugASTERISK-19365_2012_03_08.patch uploaded by mjordan (license
+ 6283) Review: https://reviewboard.asterisk.org/r/1807
+
+2012-03-20 20:32 +0000 [r360033] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_echo.c: Prevent Echo() from relaying control, null, and
+ modem frames Echo()'s description states that it echoes audio,
+ video, and DTMF except for # while it actually echoes any frame
+ that it receives other than DTMF #. This was causing frame storms
+ in the test suite in some circumstances where Echo() was attached
+ to both ends of a pair of local channels and control frames were
+ being periodically generated. Echo()'s behavior and description
+ have been modifed so that it only echoes media and non-# DTMF
+ frames.
+
+2012-03-20 17:21 +0000 [r359979] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/manager.h, main/manager.c: Allow AMI action
+ callback to be reentrant. Fix AMI module reload deadlock
+ regression from ASTERISK-18479 when it tried to fix the race
+ between calling an AMI action callback and unregistering that
+ action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change.
+ Locking the ao2 object guaranteed that there were no active
+ callbacks that mattered when ast_manager_unregister() was called.
+ Unfortunately, this causes the deadlock situation. The patch
+ stops locking the ao2 object to allow multiple threads to invoke
+ the callback re-entrantly. There is no way to guarantee a module
+ unload will not crash because of an active callback. The code
+ attempts to minimize the chance with the registered flag and the
+ maximum 5 second delay before ast_manager_unregister() returns.
+ The trunk version of the patch changes the API to fix the race
+ condition correctly to prevent the module code from unloading
+ from memory while an action callback is active. * Don't hold the
+ lock while calling the AMI action callback. (closes issue
+ ASTERISK-19487) Reported by: Philippe Lindheimer Review:
+ https://reviewboard.asterisk.org/r/1818/ Review:
+ https://reviewboard.asterisk.org/r/1820/
+
+2012-03-16 20:13 +0000 [r359892] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_chanspy.c: Prevent chanspy from binding to zombie
+ channels This patch addresses a bug with chanspy on local
+ channels which roughly 50% of the time would create a situation
+ where chanspy can latch onto a zombie channel, keeping the zombie
+ alive forever and causing the channel doing the spying to never
+ be able to hang up. (closes issue ASTERISK-19493) Reported by:
+ lvl Review: https://reviewboard.asterisk.org/r/1819/
+
+2012-03-16 08:22 +0000 [r359809] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/sip/include/sip.h: Missed lastinvite CSeq int to
+ uint32_t change from Review:
+ https://reviewboard.asterisk.org/r/1699/
+
+2012-03-15 19:01 +0000 [r359656-359706] Matthew Jordan <mjordan@digium.com>
+
+ * main/utils.c: Fix remotely exploitable stack overflow in HTTP
+ manager There exists a remotely exploitable stack buffer overflow
+ in HTTP digest authentication handling in Asterisk. The
+ particular method in question is only utilized by HTTP AMI. When
+ parsing the digest information, the length of the string is not
+ checked when it is copied into temporary buffers allocated on the
+ stack. This patch fixes this behavior by parsing out pre-defined
+ key/value pairs and avoiding unnecessary copies to the stack.
+ (closes issue ASTERISK-19542) Reported by: Russell Bryant Tested
+ by: Matt Jordan
+
+ * apps/app_milliwatt.c, /: Fix remotely exploitable stack overrun
+ in Milliwatt Milliwatt is vulnerable to a remotely exploitable
+ stack overrun when using the 'o' option. This occurs due to the
+ milliwatt_generate function not accounting for
+ AST_FRIENDLY_OFFSET when calculating the maximum number of
+ samples it can put in the output buffer. This patch resolves this
+ issue by taking into account AST_FRIENDLY_OFFSET when determining
+ the maximum number of samples allowed. Note that at no point is
+ remote code execution possible. The data that is written into the
+ buffer is the pre-defined Milliwatt data, and not custom data.
+ (closes issue ASTERISK-19541) Reported by: Russell Bryant Tested
+ by: Matt Jordan Patches: milliwatt_stack_overrun.rev1.txt by
+ Russell Bryant (license 6283) Note that this patch was written by
+ Russell, even though Matt uploaded it ........ Merged revisions
+ 359645 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
+
+2012-03-15 18:17 +0000 [r359609] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_queue.c, apps/app_dial.c: Add missing connected line
+ macro calls to initial dial for Dial and Queue apps. The
+ connected line interception macros do not get executed when the
+ outgoing channel is initially created and that channel's
+ caller-id is implicitly imported into the incoming channel's
+ connected line data. If you are using the interception macros,
+ you would expect that they get run for every change to a
+ channel's connected line information outside of normal dialplan
+ execution. Review: https://reviewboard.asterisk.org/r/1817/
+
+2012-03-15 00:52 +0000 [r359452-359558] Russell Bryant <russell@russellbryant.com>
+
+ * channels/chan_iax2.c: chan_iax2: Fix use of uninitialized
+ sockaddr_in in try_transfer(). Initialize a struct sockaddr_in in
+ try_transfer() so that the code isn't (potentially) trying to
+ read from it while uninitialized.
+
+ * channels/chan_gtalk.c: chan_gtalk: Fix use of uninitialized vars
+ in config handling. Fix potential use of context, parkinglot, and
+ prefs before they are initialized.
+
+ * channels/chan_gtalk.c: chan_gtalk: Fix potential use of
+ uninitialized variable. Avoid potential use of idroster in
+ gtalk_alloc() before it has been initialized.
+
+ * apps/app_chanisavail.c: app_chanisavail: Fix use of uninitialized
+ variable. Ensure that status is set before it is used by
+ resetting it during each loop iteration. This could have resulted
+ in incorrect results from this app.
+
+ * main/udptl.c: udptl: Ensure fec[] in udptl_build_packet() is
+ initialized. Scan results indicated that this array could be used
+ uninitialized. At a quick look, it looks correct. In any case,
+ initializing it is a Good Thing (tm).
+
+ * include/asterisk/app.h: app.h: Always initialize
+ AST_DECLARE_APP_ARGS(). This patch ensures that the struct
+ defined by AST_DECLARE_APP_ARGS() is always fully initialized.
+ I'm not sure if this fixes any real bugs, but it silences a bunch
+ of warnings from coverity, and is generally a good thing to do
+ anyway.
+
+2012-03-14 22:20 +0000 [r359451] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/channel.h, main/channel.c,
+ channels/chan_agent.c: Fix deadlock potential with some
+ ast_indicate/ast_indicate_data calls. Calling
+ ast_indicate()/ast_indicate_data() with the channel lock held can
+ result in a deadlock with a local channel because of how local
+ channels need to avoid deadlock.
+
+2012-03-14 17:32 +0000 [r359356] Matthew Jordan <mjordan@digium.com>
+
+ * main/jitterbuf.c: Fix incorrect jitter buffer overflow due to
+ missed resynchronizations When a change in time occurs, such that
+ the timestamps associated with frames being placed into an
+ adaptive jitter buffer (implemented in jitterbuf.c) are
+ significantly different then the previously inserted frames, the
+ jitter buffer checks to see if it needs to be resynched to the
+ new time frame. If three consecutive packets break the threshold,
+ the jitter buffer resynchs itself to the new timestamps. This
+ currently only occurs when history is calculated, and hence only
+ on JB_TYPE_VOICE frames. JB_TYPE_CONTROL frames, on the other
+ hand, are never passed to the history calculations. Because of
+ this, if the jump in time is greater then the maximum allowed
+ length of the jitter buffer, the JB_TYPE_CONTROL frames are
+ dropped and no resynchronization occurs. Alterntively, if the
+ overfill logic is not triggered, the JB_TYPE_CONTROL frame will
+ be placed into the buffer, but with a time reference that is not
+ applicable. Subsequent JB_TYPE_VOICE frames will quickly trigger
+ the overflow logic until reads from the jitter buffer reach the
+ errant JB_TYPE_CONTROL frame. This patch allows JB_TYPE_CONTROL
+ frames to resynch the jitter buffer. As JB_TYPE_CONTROL frames
+ are unlikely to occur in multiples, it perform the
+ resynchronization on any JB_TYPE_CONTROL frame that breaks the
+ resynch threshold. Note that this only impacts chan_iax2, as
+ other consumers of the adaptive jitter buffer use the abstract
+ jitter buffer API, which does not use JB_TYPE_CONTROL frames.
+ Review: https://reviewboard.asterisk.org/r/1814/ (closes issue
+ ASTERISK-18964) Reported by: Kris Shaw Tested by: Kris Shaw, Matt
+ Jordan Patches: jitterbuffer-2012-2-26.diff uploaded by Kris Shaw
+ (license 5722)
+
+2012-03-14 17:17 +0000 [r359344] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, main/channel.c: Fix Dial m and r options and
+ forked calls generating warnings for voice frames. When connected
+ line support was added, the wait_for_answer() variable single
+ changed its meaning slightly. Unfortunately, the places where
+ single was used did not necessarily get updated to reflect that
+ change. Also audio/video frames were sent to all forked calls
+ when the endpoints were never made compatible. * Don't pass
+ audio/video media frames when the channels have not been made
+ compatible. * Added handling of AST_CONTROL_SRCCHANGE to
+ app_dial.c. * Fixed app_dial.c passing on AST_CONTROL_HOLD
+ because that frame can also pass a requested MOH class. (closes
+ issue ASTERISK-16901) Reported by: Chris Gentle (closes issue
+ ASTERISK-17541) Reported by: clint Review:
+ https://reviewboard.asterisk.org/r/1805/
+
+2012-03-14 10:52 +0000 [r359050-359259] Russell Bryant <russell@russellbryant.com>
+
+ * include/asterisk/logger.h, main/logger.c: Fix bogus reads/writes
+ of console log levels in asterisk.c This patch updates the
+ NUMLOGLEVELS define in logger.h to 32, to match the fact that
+ logger.c implements 32 log levels (because of the custom log
+ level stuff). asterisk.c uses this define to size an array of
+ levels per remote console. This array is modified in
+ ast_console_toggle_loglevel(), which is called by the "logger set
+ level" CLI command. While the documentation for the CLI command
+ doesn't make it terribly obvious, you can use this CLI command to
+ toggle a custom log level on a remote console, as well. However,
+ doing so led to an invalid array index in asterisk.c. This array
+ is read from any time a log message is written to a console. So,
+ all custom log level messages resulted in a bogus read if a
+ remote console was connected.
+
+ * apps/app_externalivr.c, channels/chan_iax2.c: Fix invalid
+ reads/writes due to incorrect sizeof(). These few places in the
+ code used sizeof() on h_addr in struct hostent. This is
+ sizeof(char *). The correct way to get the size of this address
+ is to use h_length. This error would result in reads/writes of 8
+ bytes instead of 4 on 64-bit machines.
+
+ * main/sched.c: Fix inaccurate sizeof() in sched.c. This code just
+ needed sizeof(int), not sizeof(int *).
+
+ * utils/astman.c: Fix incorrect sizeof() in astman.
+
+ * res/res_crypto.c: Fix incorrect usage of sizeof() in res_crypto.
+ In this case, just remove the memset(). There was a redundant
+ memset that is done correctly just 2 lines later.
+
+ * res/res_adsi.c: Fix broken usage of sizeof() in res_adsi.
+
+ * main/features.c: Fix incorrect sizeof() usage in features.c. This
+ didn't actually result in a bug anywhere, luckily. The only place
+ where the result of these memcpys was used is in app_dial, and
+ the only field that it read out of ast_call_feature was the first
+ one, which is an int, so these memcpys always copied just enough
+ to avoid a problem.
+
+ * main/md5.c: Fix incorrect sizeof() on a pointer in MD5Final().
+
+ * main/pbx.c: Don't use a buffer after it goes out of scope. 's' is
+ set to 'workspace'. Make sure 'workspace' doesn't go out of scope
+ while the reference to it via 's' is still used.
+
+ * res/ais/ais.h, res/res_ais.c, res/ais/clm.c, res/ais/evt.c: Dump
+ cache of published events when a node joins the cluster. Also use
+ a more reliable method for stopping the poll() thread.
+
+ * makeopts.in, apps/app_rpt.c (removed), channels/chan_usbradio.c
+ (removed), channels/xpmr (removed),
+ build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Remove
+ chan_usbradio and app_rpt. These modules are being maintained
+ outside of the tree and have been for a long time now, so it
+ doesn't make sense to keep them here. Review:
+ https://reviewboard.asterisk.org/r/1764/
+
+2012-03-13 20:31 +0000 [r358943-358978] Terry Wilson <twilson@digium.com>
+
+ * main/features.c: Fix setting CDR variables in the hangup
+ extension A previous CDR fix for setting CDR variables during a
+ bridge via custom dialplan features broke setting CDR variables
+ in the hangup extension. This patch fixes the issue. Review:
+ https://reviewboard.asterisk.org/r/1794/
+
+ * main/devicestate.c, include/asterisk/devicestate.h,
+ channels/chan_sip.c, tests/test_devicestate.c: Make hints for
+ invalid SIP devices return Unavail, not idle This patch
+ drastically simplifies the device state aggegation code. The old
+ method was not only overly complex, but also made it impossible
+ to return AST_DEVICE_INVALID from the aggregation code. The unit
+ test update is as a result of fixing that bug. The SIP change
+ stems from a bug introduced by removing a DNS lookup for
+ hostname-based SIP channels. (closes issue ASTERISK-16702)
+ Review: https://reviewboard.asterisk.org/r/1808/
+
+2012-03-13 16:54 +0000 [r358810-358859] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * UPGRADE.txt, CHANGES: Requested changes documenting the fixed AEL
+ functionality.
+
+ * utils/ael_main.c, apps/app_stack.c, utils/conf2ael.c,
+ res/ael/pval.c, funcs/func_dialplan.c, tests/test_gosub.c: Enable
+ macros in 1.8 to find the next highest "h" extension in a
+ context, like in 1.4. This change restores functionality that was
+ present in 1.4, when AEL macros were implemented with the Macro
+ dialplan application. Macros are fraught with functionality
+ issues, because they consume a large portion of the underlying
+ application stack. This limits the ability of AEL users to call
+ many layers of subroutines, an issue which Gosub does not have
+ (originally tested to 100,000 levels deep). Therefore, starting
+ in 1.6.0, AEL macros were implemented with Gosub. However, there
+ were some implicit behaviors of Macro, which were not replicated
+ at the same time as with the transition to Gosub, one of which is
+ documented in the related issue. In particular, the "h" extension
+ is designed to execute not in the Macro context, but in the
+ topmost calling context. Due to legacy issues with a misapplied
+ bugfix many years ago, when a macro exited in 1.4, it looks in
+ all calling contexts, bubbling up from the deepest level until it
+ finds an "h" extension. Since AEL hides the complexity of the
+ underlying dialplan logic from the AEL programmer, it's
+ reasonable to assume that this behavior should not change in the
+ transition from Asterisk 1.4 LTS to Asterisk 1.8 LTS, lest we
+ break working AEL configurations in the transition to Asterisk
+ 1.8 LTS. This fix is the result, which implements a search for
+ the "h" extension in all calling Gosub contexts. Fixes
+ ASTERISK-19336 Patch: 20120308__ael_bugfix_for_trunk__2.diff
+ (License #5003) by Tilghman Lesher (with slight modifications for
+ 1.8) Tested by: Johan Wilfer Review:
+ https://reviewboard.asterisk.org/r/1776/
+
+2012-03-08 16:39 +0000 [r358643] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c: Make transfer not ignore port information
+ with SIP. Attempting to transfer with SIP to an address like
+ 1XXXXX@ip.ad.re.ss:5061 would fail because port would be cut from
+ the host string and ignored. This simply keeps chan_sip from
+ cutting off the port number during these kinds of transfers.
+ (closes issue ASTERISK-19321) Reported by: Federico Alves Review:
+ https://reviewboard.asterisk.org/r/1790/diff/#index_header
+
+2012-03-07 18:25 +0000 [r358530] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_ss7.c: Change directly setting _softhangup in
+ sig_ss7.c to use ast_softhangup_nolock(). Update to: (issue
+ ASTERISK-19372)
+
+2012-03-07 16:11 +0000 [r358484] Sean Bright <sean@malleable.com>
+
+ * codecs/codec_dahdi.c: Return g729 and g723.1 frames with the
+ number of samples set properly. If the wctc4xxp returns more than
+ a single packet, we need to update the number of samples in the
+ returned frame accordingly. Acked-by: Shaun Ruffell
+ <sruffell@digium.com>
+
+2012-03-07 15:16 +0000 [r358435-358438] Terry Wilson <twilson@digium.com>
+
+ * configs/cdr_adaptive_odbc.conf.sample: Set snarkiness = 0 in
+ cdr_adaptive_odbc.conf.sample
+
+ * cdr/cdr_adaptive_odbc.c, cel/cel_odbc.c: Add detection for ODBC
+ WCHAR fields Without detecting these types, cel_odbc blows up
+ when the character set for the table is utf8. This also wraps
+ cdr_adaptive_odbc's use of those types in the HAVE_ODBC_WCHAR
+ #ifdef seen in other parts of the code.
+
+2012-03-06 17:44 +0000 [r358260-358377] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Fix ring cadance setup for outgoing calls
+ on FXS ports. * Fix referencing the wrong variable in
+ chan_dahdi.c:my_set_cadence(). Thanks to Sean Bright for
+ compiling with -Wshadow and finding this bug.
+
+ * channels/sig_ss7.c: Drop SS7 call if not connected yet when
+ INCOMPLETE/BUSY/CONGESTION. SS7 is a trunk protocol and should
+ clear a failed call as soon as possible. * Made SS7 hangup a call
+ immediately if it has not connected yet for
+ INCOMPLETE/BUSY/CONGESTION causes. Otherwise, play an appropriate
+ inband tone. (closes issue ASTERISK-19372) Reported by: Igor
+ Nikolaev
+
+ * channels/sig_ss7.c, channels/chan_dahdi.c, channels/sig_ss7.h:
+ Setup DSP when SS7 call is connected or early media is available.
+ Outgoing SS7 calls fail to detect incoming DTMF so any bridged
+ channel that requires out-of-band DTMF will not work. * Added
+ sig_ss7_open_media() calls at appropriate places in sig_ss7.c.
+ The new call converts conditionaled out unconverted code and
+ shows that the code really did something useful. * Improved some
+ chan_dahdi DTMF debug messages to help track DTMF handling.
+ (closes issue ASTERISK-19312) Reported by: Igor Nikolaev
+
+2012-03-05 18:49 +0000 [r358214] Jonathan Rose <jrose@digium.com>
+
+ * main/manager.c: Eliminate double close of file descriptor in
+ manager.c The process_output function in manager.c attempted to
+ call fclose and close immediately afterwards. Since fclose
+ implies close, this resulted in a potential double free on file
+ descriptors. This patch changes that behavior and also adds error
+ checking to fclose and close depending on which was deemed
+ necessary. Also error messages. Thanks to Rosen Iliev for
+ pointing out the location of the problem. (closes issue
+ ASTERISK-18453) Reported By: Jaco Kroon Review:
+ https://reviewboard.asterisk.org/r/1793/
+
+2012-03-05 16:41 +0000 [r358162] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Defer sending the connected line reinvite if
+ a reinvite is already in progress. (issue ASTERISK-19355)
+ Reported by: tomaso (closes issue AST-825)
+
+2012-03-05 15:54 +0000 [r358115] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c: Ensure Asterisk acknowledges ACKs to 4xx on
+ Replaces errors Asterisk was not setting pendinginvite in the
+ upper half of handle_request_invite such that the 4xx was
+ retransmitted repeatedly even though an ack was received for
+ every retransmission. (closes issue ASTERISK-19303) Patch-by:
+ Jeremiah Gowdy
+
+2012-03-02 23:27 +0000 [r357986-358029] Terry Wilson <twilson@digium.com>
+
+ * channels/xpmr/xpmr.c, channels/chan_usbradio.c: Fix
+ unused-but-set-variable warnings All of these were pretty
+ obviously unused. Some were unused because the code that used
+ them was #if 0'd. In those cases, I just commented out the
+ unused-but-set variables.
+
+ * channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c,
+ channels/chan_misdn.c: Correct some set-but-unused variable
+ warnings in the mISDN library. (from kpfleming's commit to trunk
+ r356292)
+
+ * channels/xpmr/xpmr.c: Make chan_usbradio compile under dev mode
+ x=++x and x=x=1? Really?
+
+2012-03-02 21:02 +0000 [r357940] Kinsey Moore <kmoore@digium.com>
+
+ * main/event.c, include/asterisk/strings.h, main/ccss.c,
+ tests/test_event.c: Fix case-sensitivity for device-specific
+ event subscriptions and CCSS This change fixes case-sensitivity
+ for device-specific subscriptions such that the technology
+ identifier is case-insensitive while the remainder of the device
+ string is still case-sensitive. This should also preserve the
+ original case of the device string as passed in to the event
+ system. CCSS is the only feature affected as it is the only
+ consumer of device-specific event subscriptions. The second part
+ of this patch addresses similar case-sensitivity issues within
+ CCSS itself that prevented it from functioning correctly after
+ the fix to the events system. This adds a unit test to verify
+ that the event system works as expected. (closes issue
+ ASTERISK-19422) Review: https://reviewboard.asterisk.org/r/1780/
+
+2012-03-02 18:34 +0000 [r357894] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c, main/channel.c: Remove ISDN hold restriction
+ for non-bridged calls. The check if an ISDN call is bridged
+ before it could be placed on hold is not necessary and is overly
+ restrictive. The check was originally done to prevent problems
+ with call transfers in case a user tried to transfer a call
+ connected to an application to another call connected to an
+ application. The ISDN transfer code has not required this
+ restriction for quite some time because ECT could transfer any
+ two active calls to each other. * Remove ISDN hold restriction
+ for calls connected to applications. * Made
+ ast_waitfordigit_full() ignore AST_CONTROL_HOLD and
+ AST_CONTROL_UNHOLD instead of generating a warning message.
+ (closes issue ASTERISK-19388) Reported by: Birger Harzenetter
+ Tested by: rmudgett
+
+2012-03-02 15:58 +0000 [r357811] Sean Bright <sean@malleable.com>
+
+ * channels/chan_iax2.c: The default value for mohinterpret is the
+ empty string, so when resetting to default values don't
+ explicitly set the value to "default."
+
+2012-03-02 15:45 +0000 [r357809] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_chanspy.c: Fix channel reference leak in ChanSpy. * Fix
+ next_channel() channel reference leak in ChanSpy. (closes issue
+ ASTERISK-19461) Reported by: Irontec Patches:
+ app_chanspy_iteartor_next_unref.patch (license #6213) patch
+ uploaded by Irontec (issue ASTERISK-17515)
+
+2012-03-02 00:59 +0000 [r357760-357761] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c: Fix race condition that can cause important
+ control frames (such as a hangup) to be missed. This takes two
+ actions. 1. Move the reading of the alertpipe in __ast_read() to
+ immediately before the removal of frames from the readq. This
+ means we won't do something silly like read from the alertpipe,
+ then ignore the fact that there's a frame to get from the readq
+ since channel's fdno is the AST_TIMING_FD. 2. When
+ ast_settimeout() sets the rate to 0 and the timingfunc to NULL,
+ if the channel's fdno is the AST_TIMING_FD, then set the fdno to
+ -1. This is because if the rate is 0 and the timingfunc is NULL,
+ it means that the channel's timing fd is being invalidated, so
+ any pending reads should not occur. This may actually solve more
+ issues than the referenced one below, but it's not known at this
+ time for sure. (closes issue ASTERISK-19223) reported by
+ Frank-Michael Wittig Review:
+ https://reviewboard.asterisk.org/r/1779
+
+ * main/translate.c: Second attempt to get optimal translation paths
+ when codec_resample is used. This borrows code heavily from
+ changes made in translation code in Asterisk 10. This uses the
+ quality and sample rate change of translation in order to pick
+ paths rather than the computational cost of translations.
+ Computational cost is used solely in determining if a single
+ translation step from a specific translator is better than the
+ same translation step provided by a different translator. (closes
+ issue ASTERISK-16821) reported by Andrew Lindh Review:
+ https://reviewboard.asterisk.org/r/1772
+
+2012-03-01 14:18 +0000 [r357665] Kinsey Moore <kmoore@digium.com>
+
+ * main/acl.c: Prevent outbound SIP NOTIFY packets from displaying a
+ port of 0 In the change from 1.6.2 to 1.8, ast_sockaddr was
+ introduced which changed the behavior of ast_find_ourip such that
+ port number was wiped out. This caused the port in internip
+ (which is used for Contact and Call-ID on NOTIFYs) to be 0. This
+ change causes ast_find_ourip to be port-preserving again. (closes
+ issue ASTERISK-19430)
+
+2012-02-29 19:41 +0000 [r357575] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * apps/app_dial.c: Fix copying of CDR(accountcode) to local
+ channels. In r203638, during the addition of the Channel Event
+ Logging, in mid-2009, this got broken in trunk and ended up in
+ asterisk 1.8 and higher. This fixes so the CDR(accountcode) from
+ the calling channel is available to dialed channels again as well
+ as showing up properly in the CDR's. (closes issue
+ ASTERISK-19384) Patches: accountcode.patch (License #6033) by
+ jamicque Review: https://reviewboard.asterisk.org/r/1775/
+ Reviewed by: Richard Mudgett
+
+2012-02-28 22:27 +0000 [r357455-357490] Jonathan Rose <jrose@digium.com>
+
+ * UPGRADE.txt, configs/sip.conf.sample: Adding transport=udp to
+ sample sip.conf - Also changes version of Asterisk 1.8 in UPGRADE
+ (issue ASTERISK-19352) Reported by: jamicque Patches:
+ asterisk-19352-transport-warning-message-v1.patch uploaded by
+ Michael L. Young (license 5026)
+
+ * cdr/cdr_adaptive_odbc.c: Add additional character type types to
+ supported data types for cdr_adaptive_odbc The reporter was uable
+ to use varchar utf8_unicode_ci with cdr_adaptive_odbc, so this
+ patch adds those along with some other character types to the
+ list of types cdr_adaptive_odbc will work using the varchar
+ conditions. The problem wasn't really UTF8 characters as much as
+ it was a failure to respond to the exact type that was
+ declared/in use on that database. (closes issue ASTERISK-19334)
+ Reported By: Igor Nikolaev Patches: cdr_adaptive_odbc.patch
+ uploaded by Igor Nikolaev (license 6236)
+
+2012-02-28 21:19 +0000 [r357416] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * apps/app_stack.c: Correctly reset the dialplan priority. When the
+ stack frame is allocated, we save the address to which we should
+ return, when the Gosub returns. However, if we just want to
+ restore the priority, then we need to subtract 1 before setting
+ it. Otherwise, when a Gosub goes to a nonexistent address, it
+ will skip a priority in the dialplan. This is because when we
+ return from an application, the PBX increments the priority for
+ us.
+
+2012-02-28 20:57 +0000 [r357407] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Use more reasonable cause code when rejecting
+ incoming call waiting calls. (closes issue ASTERISK-19397)
+ Reported by: Birger Harzenetter Patches: nochannel-cause.patch
+ (license #5870) patch uploaded by Birger Harzenetter
+
+2012-02-28 20:26 +0000 [r357356-357386] Jonathan Rose <jrose@digium.com>
+
+ * UPGRADE.txt: Moves UPGRADE.txt notes from r357356 to a new
+ section specific to 1.8.12 (issue ASTERISK-19352) reported by:
+ jamicque
+
+ * UPGRADE.txt: Adds UPGRADE.txt notes to r357266 indicating changes
+ to transport option (issue ASTERISK-19352) Reported by: jamicque
+
+2012-02-28 19:32 +0000 [r357352] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_page.c: Remove dupliate 'i' option table entry in
+ app_page.c. (closes issue ASTERISK-19310) Reported by: Makoto Dei
+ Patches: app_page-duplicate-i-option.patch (license #5027) patch
+ uploaded by Makoto Dei
+
+2012-02-28 18:00 +0000 [r357266] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c: Changes transport option in sip.conf so that
+ using multiple instances doesn't stack. Prior to this patch,
+ Using "transport=" multiple times would cause them to add to one
+ another like allow/deny. This patch changes that behavior to
+ simply use the transport option specified last. Also, if no
+ transport option is applied now, the default will automatically
+ be UDP. (closes ASTERISK-19352) Reported by: jamicque Patches:
+ asterisk-19352-transport-warning-message-v1.patch uploaded by
+ Michael L. Young (license 5026)
+ issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes
+ (license 5674) Review:
+ https://reviewboard.asterisk.org/r/1745/diff/#index_header
+
+2012-02-28 14:45 +0000 [r357212] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile.rules: Make COMPILE_DOUBLE magic actually work. The
+ build system has some special magic to ensure that if Asterisk is
+ built with --enable-dev-mode *and* DONT_OPTIMIZE, that all the
+ source is still compiled with the optimizer enabled (even though
+ the result will be thrown away), because the compiler is able to
+ find a great deal of coding errors and bugs as a result of
+ running its optimizers. Unfortunately at some point this mode got
+ broken, and the 'throwaway' compile of the code was no longer
+ done with the optimizer enabled. This patch corrects that
+ problem.
+
+2012-02-27 23:34 +0000 [r357093] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Fix callerid of Originated calls. Thanks to Matt
+ Riddell for tracking this down. (closes issue ASTERISK-19385)
+ Reported by: ornix
+
+2012-03-29 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.11.0 Released.
+
+2012-03-26 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.11.0-rc3 Released.
+
+ * AST-2012-003
+
+ * AST-2012-002
+
+ * /main/manager.c, /include/asterisk/manager.h: Fix AMI deadlock
+ regression by allowing AMI action callback to be reentrant
+
+ Fix AMI module reload deadlock from ASTERISK-18479 when it tried
+ to fix the race between calling an AMI action callback and
+ unregistering that action. Refixes ASTERISK-13784 broken by
+ ASTERISK-17785 change.
+
+ Locking the ao2 object guaranteed that there were no active
+ callbacks that mattered when ast_manager_unregister() was called.
+ Unfortunately, this causes the deadlock situation. The patch stops
+ locking the ao2 object to allow multiple threads to invoke the
+ callback re-entrantly. There is no way to guarantee a module unload
+ will not crash because of an active callback. The code attempts to
+ minimize the chance with the registered flag and the maximum 5
+ second delay before ast_manager_unregister() returns.
+
+ The trunk version of the patch changes the API to fix the race
+ condition correctly to prevent the module code from unloading from
+ memory while an action callback is active.
+
+ * Don't hold the lock while calling the AMI action callback.
+
+ (closes issue ASTERISK-19487)
+ Reported by: Philippe Lindheimer
+
+ Review: https://reviewboard.asterisk.org/r/1818/
+
+2012-03-06 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.11.0-rc2 Released.
+
+ * main/acl.c: Prevent outbound SIP NOTIFY packets from displaying
+ a port of 0.
+
+ In the change from 1.6.2 to 1.8, ast_sockaddr was
+ introduced which changed the behavior of ast_find_ourip such
+ that port number was wiped out. This caused the port in
+ internip (which is used for Contact and Call-ID on NOTIFYs) to be
+ 0. This change causes ast_find_ourip to be port-preserving again.
+
+2012-01-30 21:57 +0000 [r353368-353320] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/sip/include/sip.h, channels/sip/include/dialog.h,
+ channels/chan_sip.c: RFC3261 Section 8.1.1.5. The sequence number
+ value MUST be expressible as a 32-bit unsigned integer * fix: use
+ %u instead of %d when dealing with CSeq numbers - to remove
+ possibility of -ve numbers. * fix: change all uses of seqno and
+ friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t.
+ Summary of CSeq numbers. An initial CSeq number must be less than
+ 2^31 A CSeq number can increase in value up to 2^32-1 An
+ incrementing CSeq number must not wrap around to 0. Tested with
+ Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585)
+ Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1699/
+
+ * channels/chan_sip.c: prevent debug messsges displaying -ve Cseq
+ numbers. Missed in R353320
+
+2012-01-30 23:17 +0000 [r353371] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/dnsmgr.h, main/dnsmgr.c, channels/chan_sip.c:
+ Re-link peers by IP when dnsmgr changes the IP Asterisk's dnsmgr
+ currently takes a pointer to an ast_sockaddr and updates it
+ anytime an address resolves to something different. There are a
+ couple of issues with this. First, the ast_sockaddr is usually
+ the address of an ast_sockaddr inside a refcounted struct and we
+ never bump the refcount of those structs when using dnsmgr. This
+ makes it possible that a refresh could happen after the
+ destructor for that object is called (despite ast_dnsmgr_release
+ being called in that destructor). Second, the module using dnsmgr
+ cannot be aware of an address changing without polling for it in
+ the code. If an action needs to be taken on address update (like
+ re-linking a SIP peer in the peers_by_ip table), then polling for
+ this change negates many of the benefits of having dnsmgr in the
+ first place. This patch adds a function to the dnsmgr API that
+ calls an update callback instead of blindly updating the address
+ itself. It also moves calls to ast_dnsmgr_release outside of the
+ destructor functions and into cleanup functions that are called
+ when we no longer need the objects and increments the refcount of
+ the objects using dnsmgr since those objects are stored on the
+ ast_dnsmgr_entry struct. A helper function for returning the
+ proper default SIP port (non-tls vs tls) is also added and used.
+ This patch also incorporates changes from a patch posted by Timo
+ Teräs to ASTERISK-19106 for related dnsmgr issues. (closes issue
+ ASTERISK-19106) Review: https://reviewboard.asterisk.org/r/1691/
+
+2012-01-31 16:51 +0000 [r353454] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/channel.h, main/manager.c: Fix memory leak in
+ error paths for action_originate(). * Fix memory leak of vars in
+ error paths for action_originate(). * Moved struct
+ fast_originate_helper tech and data members to stringfields. *
+ Simplified ActionID header handling for fast_originate(). * Added
+ doxygen note to ast_request() and ast_call() and the associated
+ channel callbacks that the data/addr parameters should be treated
+ as const char *. Review: https://reviewboard.asterisk.org/r/1690/
+
+2012-01-31 23:41 +0000 [r353502] Terry Wilson <twilson@digium.com>
+
+ * res/res_calendar.c: Allow res_calendar to be unloaded The
+ calendaring tech modules depend on res_calendar and initially
+ res_calendar just bumped the use count so that it couldn't be
+ unloaded. res_calendar can potentially create many threads and
+ I've seen issues where the Asterisk shutdown has failed where it
+ looked like these threads could be the culprit. This patch adds
+ unload support for res_calendar. Unloading res_calendar will also
+ unload the dependant tech modules as well. (closes issue
+ ASTERISK-16744) Review: https://reviewboard.asterisk.org/r/1657/
+
+2012-02-01 15:02 +0000 [r353550] Matthew Jordan <mjordan@digium.com>
+
+ * contrib/init.d/etc_default_asterisk: Added clarification for the
+ VERBOSITY setting to etc_default_asterisk Clarified that using
+ the VERBOSITY setting in etc_default_asterisk is the same as
+ using the -v command line switch, which causes Asterisk to launch
+ in console mode. (closes issue ASTERISK-17030) Reported by: Jonas
+
+2012-02-01 15:50 +0000 [r353598] Sean Bright <sean@malleable.com>
+
+ * include/asterisk/audiohook.h: Resolve an overlap in the
+ ast_audiohook_flags values. AST_AUDIOHOOK_TRIGGER_WRITE and
+ AST_AUDIOHOOK_WANTS_DTMF were overlapping which may have caused
+ unintended side effects. This patch moves
+ AST_AUDIOHOOK_TRIGGER_WRITE, and updates
+ AST_AUDIOHOOK_TRIGGER_MODE to reflect the original intention.
+ This will affect existing modules that use these flags, so be
+ sure to recompile as necessary. (closes issue ASTERISK-19246)
+ Reported by: feyfre
+
+2012-02-01 21:05 +0000 [r353769-353720] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c: Use ast_sockaddr_stringify_fmt wrappers for
+ various functions in chan_sip There are a number of cleaner
+ looking wrappers for ast_sockaddr_stringify_fmt available which
+ are slightly more readable than using a direct call to
+ ast_sockaddr_stringify_fmt. This patch switches a number of those
+ calls in chan_sip to use those wrappers and is generally
+ harmless. (Closes issue ASTERISK-16930) Reported by: Michael L.
+ Young Patches: chan_sip-broken-registration-1.8.diff uploaded by
+ Michael L. Young (license 5026)
+
+ * channels/chan_sip.c: Fix sip show peers port output, align
+ columns, and fix ami port output. A previous patch I committed
+ from ASTERISK-16930 unexpectedly changed some output for the AMI
+ action "sippeers" which this patch changes back. Also, this
+ aligns the output for the cli command "sip show peers" and fixes
+ another issue that patch introduced by using
+ ast_sockaddr_stringify calls multiple times without immediately
+ using the pointer. I also went ahead and did a little janitorial
+ work to clean up whitespace in _sip_show_peers. (issue
+ ASTERISK-16930) (closes issue ASTERISK-19281) Reported by:
+ Patrick El Youssef Patches: ASTERISK-19281.diff uploaded by
+ Walter Doekes (license 5674)
+
+2012-02-02 16:58 +0000 [r353770] Mark Michelson <mmichelson@digium.com>
+
+ * UPGRADE.txt, configs/manager.conf.sample,
+ include/asterisk/manager.h, configs/http.conf.sample,
+ main/manager.c, main/http.c: Fix TLS port binding behavior as
+ well as reload behavior: * Removes references to tlsbindport from
+ http.conf.sample and manager.conf.sample * Properly bind to port
+ specified in tlsbindaddr, using the default port if specified. *
+ On a reload, properly close socket if the service has been
+ disabled. A note has been added to UPGRADE.txt to indicate how
+ ports must be set for TLS. (closes issue ASTERISK-16959) reported
+ by Olaf Holthausen (closes issue ASTERISK-19201) reported by
+ Chris Mylonas (closes issue ASTERISK-19204) reported by Chris
+ Mylonas Review: https://reviewboard.asterisk.org/r/1709
+
+2012-02-02 18:31 +0000 [r353818] Jonathan Rose <jrose@digium.com>
+
+ * funcs/func_curl.c: Backports some documentation for func_curl
+ from 10 to 1.8 For some reason this function was completely
+ undocumented in 1.8. I copied the 10 docs over to 1.8 and removed
+ references to an enumerator that was added in the Asterisk 10
+ version of func_curl. That was the only change I noted. (closes
+ issue ASTERISK-19186) Reported by: Olivier Krief
+
+2012-02-02 20:01 +0000 [r353867] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c:
+ Restore the 'w' modifier support for ISDN spans.
+ Dial(DAHDI/g0/1234w888) This feature also causes the sending
+ complete ie to be sent for switch types that do not automatically
+ send the ie. (EuroISDN/ETSI) The main difference between dialing
+ Dial(DAHDI/g0/1234w888) and Dial(DAHDI/g0/1234,,D(888)) is the
+ sending of the sending complete ie. (closes issue ASTERISK-19176)
+ Reported by: rmudgett Tested by: rmudgett
+
+2012-02-02 22:26 +0000 [r353915] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c: Ensure entering T.38 passthrough does not
+ cause an infinite loop After R340970 Asterisk was still polling
+ the RTCP file descriptor after RTCP is shut down and removed. If
+ the descriptor happened to have data ready when the removal
+ occured then Asterisk would go into an infinite loop trying to
+ read data that it can never actually access. This change disables
+ the audio RTCP file descriptor for the duration of the T.38
+ transaction. (closes issue ASTERISK-18951) Reported-by: Kristijan
+ Vrban
+
+2012-02-03 21:24 +0000 [r353999] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_agent.c: Fixes deadlocks occuring in chan_agent due
+ to r335976 Bad locking order was added to chan_agent to prevent
+ segfaults from having no locking in a patch by irroot. This patch
+ addresses the bad locking order by releasing locks before getting
+ the right locking order to stop deadlocks from occuring when
+ doing multiple interactions with agents. (closes issue
+ ASTERISK-19285) Reported by: Alex Villacis Lasso Review:
+ https://reviewboard.asterisk.org/r/1708/
+
+2012-02-06 17:28 +0000 [r354216-354116] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: Add missing headers to AMI UnParkedCall event to
+ uniquely identify the call. The AMI UnParkedCall event was
+ missing the Parkinglot and Uniqueid headers that the AMI
+ ParkedCall event contains. (closes issue ASTERISK-19240) Reported
+ by: Michael Yara
+
+ * pbx/pbx_config.c: Improved documentation of CLI "dialplan add
+ extension" command. * Documented dialplan add extension
+ <exten>,<priority>,<app(<app-data>)> format. * Allow acceptance
+ of command without the app-data value. There are many
+ applications that do no need any parameters so it is silly to
+ require that field for all commands. * Fixed a couple
+ ast_malloc/ast_free mismatches with ast_add_extension2() calls.
+ (closes issue ASTERISK-19222) Reported by: Andrey Solovyev Tested
+ by: rmudgett
+
+2012-02-07 15:04 +0000 [r354263] Jonathan Rose <jrose@digium.com>
+
+ * cdr/cdr_pgsql.c: Fix column duplication bug in module reload for
+ cdr_pgsql. Prior to this patch, attempts to reload cdr_pgsql.so
+ would cause the column list to keep its current data and then add
+ a second copy during the reload. This would cause attempts to log
+ the CDR to the database to fail. This patch also cleans up some
+ unnecessary null checks for ast_free and deals with a few
+ potential locking problems. (closes issue ASTERISK-19216)
+ Reported by: Jacek Konieczny Review:
+ https://reviewboard.asterisk.org/r/1711/
+
+2012-02-07 20:53 +0000 [r354348] Terry Wilson <twilson@digium.com>
+
+ * contrib/realtime/postgresql/realtime.sql, channels/chan_sip.c:
+ Fix multiple SIP realtime issues 1. Set lastms to 0 when clearing
+ instead of "" 2. Don't set ipaddr or port to the string "(null)"
+ when they are empty 3. Add missing required fields, set default
+ for lastms to 0, and modify the length of the ipaddr field to 45
+ in the Postgresql realtime.sql file. (closes issue
+ ASTERISK-19172) Review: https://reviewboard.asterisk.org/r/1703/
+
+2012-02-09 02:23 +0000 [r354492] Russell Bryant <russell@russellbryant.com>
+
+ * main/channel.c: Remove some unnecessary locking from
+ ast_hangup(). This patch removes some unnecessary locking of the
+ channels container in ast_hangup(). The reason this came up is
+ that this lock can very quickly block the entire system. If any
+ of the channel cleanup code decides to block, it causes a problem
+ for the whole system. For example, when audiohooks get destroyed,
+ if that blocks for a while waiting on the mixmonitor thread to
+ exit because it's busy blocking on some I/O, it causes a problem
+ for many other threads in the meantime. Review:
+ https://reviewboard.asterisk.org/r/1712/
+
+2012-02-09 02:52 +0000 [r354495] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_parkandannounce.c: Fix crash in ParkAndAnnounce. Well,
+ thats embarrasing. I forgot to initialize the caller_id storage.
+ (closes issue ASTERISK-19311) Reported by: tootai Tested by:
+ rmudgett
+
+2012-02-09 16:30 +0000 [r354542] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_sip.c: Fix SIP INFO DTMF handling for non-numeric
+ codes In ASTERISK-18924, SIP INFO DTMF handlingw as changed to
+ account for both lowercase alphatbetic DTMF events, as well as
+ uppercase alphabetic DTMF events. When this occurred, the
+ comparison of the character buffer containing the event code was
+ changed such that the buffer was first compared again '0' and '9'
+ to determine if it was numeric. Unfortunately, since the first
+ character in the buffer will typically be '1' in the case of
+ non-numeric event codes (10-16), this caused those codes to be
+ converted to a DTMF event of '1'. This patch fixes that, and
+ cleans up handling of both application/dtmf-relay and
+ application/dtmf content types. Review:
+ https://reviewboard.asterisk.org/r/1722/ (closes issue
+ ASTERISK-19290) Reported by: Ira Emus Tested by: mjordan
+
+2012-02-09 16:56 +0000 [r354545] Mark Michelson <mmichelson@digium.com>
+
+ * CHANGES, res/res_fax.c: Adding reload support to res_fax.so
+ (closes issue ASTERISK-16712) reported by Frank DiGennaro Review:
+ https://reviewboard.asterisk.org/r/1713
+
+2012-02-09 17:07 +0000 [r354547] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_sip.c: Clean-up of minor formatting issues in
+ r354542/3/4 rmudgett pointed out some formatting issues in the
+ check-in for ASTERISK-19290. This cleans those up. Review:
+ https://reviewboards.asterisk.org/r/1722/
+
+2012-02-09 17:32 +0000 [r354640-354594] Mark Michelson <mmichelson@digium.com>
+
+ * main/translate.c: Fix translation path choices. This change makes
+ it so computational cost is not taken into account when deciding
+ if a multistep path is better than a single-step path. This means
+ that the only time a multistep path will be chosen is if no
+ single-step path exists. This ensures a better quality
+ translation even if it turns out to be slightly slower. (closes
+ issue ASTERISK-16821) reported by Andrew Lindh Review:
+ https://reviewboard.asterisk.org/r/1715
+
+ * main/translate.c: Remove outdated comment.
+
+2012-02-09 19:52 +0000 [r354702-354655] Kinsey Moore <kmoore@digium.com>
+
+ * main/config.c: Make the config parser remove escaping backslashes
+ The config parser in Asterisk does not currently remove a
+ backslash that is used to escape a semicolon which would
+ otherwise be interpreted as the start of a comment. The change
+ here causes that backslash to be removed, but does not create a
+ real escape system in the config parser. The biggest complication
+ with a real escape system would be breaking existing configs
+ everywhere (parsing \\ as \ and breaking on escaped non-semicolon
+ characters) even though it would be the "right" way to do things.
+ (closes issue ASTERISK-17121) Review:
+ https://reviewboard.asterisk.org/r/1724/
+
+ * channels/chan_sip.c: Fix parsing of SIP headers where compact and
+ non-compact headers are mixed Change parsing of SIP headers so
+ that compactness of the header no longer influences which header
+ will be chosen. Previously, a non-compact header would be chosen
+ instead of a preceeding compact-form header. (closes issue
+ ASTERISK-17192) Review: https://reviewboard.asterisk.org/r/1728/
+
+2012-02-09 22:01 +0000 [r354749] Terry Wilson <twilson@digium.com>
+
+ * funcs/func_cdr.c: Note that CDRs are immutable once a bridge is
+ torn down CDRs cannot be modified after a bridge is torn down,
+ (e.g. after Dial() returns) even though the CDR() function may be
+ called. Since modifying the CDR code to change this behavior
+ could very easily break all kinds of things, this patch just
+ documents this limitation. (closes issues ASTERISK-16923) Review:
+ https://reviewboard.asterisk.org/r/1720/
+
+2012-02-10 18:03 +0000 [r354835] Richard Mudgett <rmudgett@digium.com>
+
+ * main/manager.c: Fix AMI Redirect ExtraChannel not redirecting to
+ the same exten and context. The astman_get_header() never returns
+ NULL so the check by the code for NULL would never fail. (closes
+ issue ASTERISK-16974) Reported by: Nuno Borges Patches:
+ 0018325.patch (license #6116) patch uploaded by Nuno Borges
+ (modified)
+
+2012-02-10 21:45 +0000 [r354889] Jason Parker <jparker@digium.com>
+
+ * apps/app_voicemail.c: Fix a voicemail memory leak with
+ heard/deleted messages. open_mailbox() was changed quite a long
+ time ago to allocate this memory. close_mailbox() should have
+ been changed to be responsible for freeing it.
+
+2012-02-13 17:22 +0000 [r354953] Richard Mudgett <rmudgett@digium.com>
+
+ * res/res_config_pgsql.c, configs/extconfig.conf.sample: Fix
+ reconnecting to pgsql database after connection loss. There can
+ only be one database connection in res_config_pgsql just like
+ res_config_sqlite. If the connection is lost, the connection may
+ not get reestablished to the same database if the res_pgsql.conf
+ and extconfig.conf files are inconsistent. * Made only use the
+ configured database from res_pgsql.conf. * Fixed potential buffer
+ overwrite of last[] in config_pgsql(). (closes issue
+ ASTERISK-16982) Reported by: german aracil boned Review:
+ https://reviewboard.asterisk.org/r/1731/
+
+2012-02-13 19:49 +0000 [r355009] Joshua Colp <jcolp@digium.com>
+
+ * pbx/pbx_config.c: Only allow one 'dialplan reload' to execute at
+ a time as otherwise they would share the same common local
+ context list. (closes issue AST-758)
+
+2012-02-13 22:02 +0000 [r355056] Richard Mudgett <rmudgett@digium.com>
+
+ * pbx/pbx_spool.c: Fix occasional incorrectly delayed call-file
+ execution. Since the dir timestamp is available at one second
+ resolution, we cannot know if it was updated within the same
+ second after we scanned it. Therefore, we will force another scan
+ if the dir was just modified. * Changed to force another scan if
+ the directory was just modified. (closes issue ASTERISK-19081)
+ Reported by: Knut Bakke Review:
+ https://reviewboard.asterisk.org/r/1688/
+
+2012-02-14 09:41 +0000 [r355136] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c: call manager_event only if there is not
+ null channel structure (Closes issue ASTERISK-19298) Reported by:
+ robinfood Patches: issue19298.patch uploaded by may213 (License
+ #5415)
+
+2012-02-14 13:33 +0000 [r355182] Sean Bright <sean@malleable.com>
+
+ * channels/chan_iax2.c: Clear the high order bit from the
+ destination call number before sending. send_apathetic_reply
+ takes the incoming frame's source call number as the destination
+ call number for the outgoing frame. If the incoming frame was a
+ full frame, then the high order bit of the source call number is
+ set and will be interpreted as a retransmit when sent back out as
+ the destination call number.
+
+2012-02-14 15:50 +0000 [r355228] Jason Parker <jparker@digium.com>
+
+ * configs/cdr_sqlite3_custom.conf.sample: Don't enable sqlite3 CDRs
+ by default in sample configs.
+
+2012-02-14 16:26 +0000 [r355268] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Properly invert the return of a strncmp
+ call. This was causing identification that should have been made
+ private to be public. (closes issue AST-814) reported by Patrick
+ Anderson Patches: chan_sip.c.diff uploaded by Patrick Anderson
+ (license 5430)
+
+2012-02-14 18:12 +0000 [r355365-355319] Richard Mudgett <rmudgett@digium.com>
+
+ * cel/cel_sqlite3_custom.c: Fix lock typo that should be unlock in
+ cel_sqlite_custom reload. (closes issue ASTERISK-19356) Reported
+ by: Alex Villacis Lasso Patches:
+ asterisk-1.8.9.2-cel_sqlite3_custom-fix-reload-locking-typo.patch
+ (license #5617) patch uploaded by Alex Villacis Lasso Review:
+ https://reviewboard.asterisk.org/r/1740/
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ formats/format_ogg_vorbis.c: Fix voicemail problems when using
+ ogg/vorbis. Ogg/vorbis was fairly useless as a voicemail file
+ format because it did not implement the seek and tell format
+ callbacks among other problems. Since we were already using the
+ libvorbis and libvorbisenc libraries we can use libvorbisfile as
+ it is also part of the vorbis library package. * Made use the
+ libvorbisfile to handle the ogg/vorbis file stream. The
+ format_ogg_vorbis.c is now mostly a wrapper around libvorbisfile.
+ (closes issue ASTERISK-16926) Reported by: sque Patches:
+ ogg_vorbis_use_libvorbisfile.patch (license #6108) patch uploaded
+ by sque
+
+2012-02-15 17:24 +0000 [r355529-355448] Sean Bright <sean@malleable.com>
+
+ * channels/chan_iax2.c: Use TRUNK_CALL_START as originally
+ intended. Back in r646, TRUNK_CALL_START was added and defined as
+ 0x4000. That same value was also hard-coded in one part of the
+ IAX2 code instead of using the #define. TRUNK_CALL_START has
+ changed over the years (for dealing with LOW_MEMORY), but the
+ hard-coded usage was never updated to match. This patch fixes
+ that.
+
+ * channels/chan_iax2.c: Only use maxtrunkcall and maxnontrunkcall
+ in chan_iax2 if IAX_OLD_FIND is specified. These variables are
+ only accessed from the IAX_OLD_FIND path, so there is no reason
+ to keep them updated otherwise.
+
+ * channels/chan_iax2.c: When IAX2 debugging is enabled, make sure
+ to log 'apathetic' messages too.
+
+2012-02-16 18:26 +0000 [r355608-355574] Richard Mudgett <rmudgett@digium.com>
+
+ * res/res_monitor.c: Fix AMI Monitor action without File header
+ converting channel name into filename. * Fix potential Solaris
+ crash if Monitor application has a urlbase and no fname_base
+ option.
+
+ * configure, include/asterisk/autoconfig.h.in,
+ autoconf/ast_c_declare_check.m4 (added), configure.ac,
+ formats/format_ogg_vorbis.c: Fix compile problem when old version
+ of libvorbisfile v1.1.2 is used. The principle difference between
+ libvorbisfile v1.1.2 and newer (at least v1.2.0) is the addition
+ of the predefined callbacks OV_CALLBACKS_xxx in
+ vorbis/vorbisfile.h used for ov_open_callbacks(). * Updated the
+ configure script to detect if libvorbisfile.h declares
+ OV_CALLBACKS_NOCLOSE. * Copied the declaration of
+ OV_CALLBACKS_NOCLOSE from v1.2.0 to allow v1.1.2 to compile.
+ (closes issue ASTERISK-19370) Reported by: Jonn Taylor
+
+2012-02-16 20:01 +0000 [r355622] Sean Bright <sean@malleable.com>
+
+ * main/audiohook.c: Revert a change to audio_audiohook_write_list
+ that had no affect. When I made this change initially, I was
+ under the false impression that the audiohooks structure remained
+ on the channel after all of the hooks had been detached. This is
+ not the case, ast ast_read takes care of removing the audiohooks
+ structure if the lists are empty.
+
+2012-02-16 23:53 +0000 [r355711-355700] Paul Belanger <pabelanger@digium.com>
+
+ * addons/ooh323cDriver.c, addons/ooh323c/src/ooSocket.c,
+ addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooTimer.c,
+ addons/ooh323c/src/ooCapability.c, addons/ooh323c/src/perutil.c:
+ Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2)
+
+ * addons/ooh323c/src/ooSocket.c: Missed a variable
+
+ * addons/ooh323cDriver.c, addons/ooh323c/src/ooSocket.c,
+ addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooTimer.c,
+ addons/ooh323c/src/ooCapability.c, addons/ooh323c/src/perutil.c:
+ Revert 355700 and 355701
+
+2012-02-17 16:04 +0000 [r355732-355721] Mark Michelson <mmichelson@digium.com>
+
+ * main/translate.c: Revert change to translate.c as it has caused
+ an infinite loop to occur in circumstances.
+
+ * channels/chan_sip.c: Fix regressions with regards to route-set
+ creation on early dialogs. This fixes two main issues: 1.
+ Asterisk would send a CANCEL to the route created by the
+ provisional response instead of using the same destination it did
+ in the initial INVITE. 2. If a new route set arrives in a 200 OK
+ than was in the 1XX response (perfectly possible if our outbound
+ INVITE gets forked), then the route set in the 200 OK needs to
+ overwrite the route set in the 1XX response. (closes issue
+ ASTERISK-19358) Reported by: Karsten Wemheuer Tested by: Karsten
+ Wemheuer patches: ASTERISK-19358.patch uploaded by Mark Michelson
+ (license 5049) ASTERISK-19358.patch uploaded by Stefan Schmidt
+ (license 6034) Review: https://reviewboard.asterisk.org/r/1749
+
+2012-02-17 19:32 +0000 [r355793-355746] Sean Bright <sean@malleable.com>
+
+ * channels/chan_iax2.c: Pass the correct value to
+ ast_timer_set_rate() for IAX2 trunking. IAX2 uses the trunkfreq
+ variable to determine how often to send trunk packets, but this
+ value is in milliseconds while ast_timer_set_rate() expects the
+ rate argument to be ticks per second. So we divide 1000 by
+ trunkfreq and pass that in instead. With a default of 20ms, this
+ change makes IAX2 send trunk packets every 20ms instead of every
+ 50ms. Tracked down by myself and Bob Wienholt.
+
+ * channels/chan_iax2.c, configs/iax.conf.sample: Don't allow
+ trunkfreq to be greater than 1000ms.
+
+2012-02-18 03:59 +0000 [r355839] Paul Belanger <pabelanger@digium.com>
+
+ * res/res_pktccops.c: Fix -Werror=unused-but-set-variable compiler
+ error (gcc 4.6.2)
+
+2012-02-18 07:55 +0000 [r355850] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/sig_pri.c, channels/sig_ss7.c, channels/sig_pri.h,
+ channels/chan_dahdi.c, channels/sig_analog.c, channels/sig_ss7.h,
+ channels/sig_analog.h: push 'outgoing' flag from sig_XXX up to
+ chan_dahdi 'p->outgoing' in chan_dahdi and sig_analog wern't kept
+ in sync, particulary FXS ast_hangup didn't clear the 'outgoing'
+ flag. sig_pri and sig_ss7 were keeping 'outgoing' flag insync.
+ Now provides a callback for all the low level sig_XXX modules.
+ (issue ASTERISK-19316) alecdavis (license 585) Reported by:
+ Jeremy Pepper Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1747/
+
+2012-02-19 17:49 +0000 [r356107-355901] Sean Bright <sean@malleable.com>
+
+ * channels/chan_iax2.c: Set the length of the ast_sockaddr, so that
+ we can set it's port later. Without this, the call to
+ ast_sockaddr_set_port a few lines later is a noop.
+
+ * channels/chan_iax2.c: Add some boilerplate documentation for
+ IAXVAR and IAXPEER.
+
+ * channels/chan_dahdi.c: Change some debug messages from LOG_DEBUG
+ to ast_debug.
+
+ * channels/chan_dahdi.c: This was a LOG_NOTICE, so roll it back.
+
+ * channels/chan_iax2.c: Remove spurious warning when
+ 'qualifyfreqnotok' is set successfully. (closes issue
+ ASTERISK-17176) Reported by: John Covert Tested by: Sean Bright
+ Patches: chan_iax2.c.qualifyfreqnotok.patch uploaded by John
+ Covert (license 5512)
+
+ * channels/chan_iax2.c: Make 'iax2 show callnumber usage' output
+ make sense when an IP is passed in.
+
+2012-02-22 14:50 +0000 [r356214] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_sip.c: Fix potential buffer overrun and memory leak
+ when executing "sip show peers" The "sip show peers" command uses
+ a fix sized array to sort the current peers in the peers
+ ao2_container. The size of the array is based on the current
+ number of peers in the container. However, once the size of the
+ array is determined, the number of peers in the container can
+ change, as the peers container is not locked. This could cause a
+ buffer overrun when populating the array, if peers were added to
+ the container after the array was created. Additionally, a memory
+ leak of the allocated array would occur if a user caused the
+ _show_peers method to return CLI_SHOWUSAGE. We now create a
+ snapshot of the current peers using an ao2_callback with the
+ OBJ_MULTIPLE flag. This size of the array is set to the number of
+ peers that the iterator will iterate over; hence, if peers are
+ added or removed from the peers container it will not affect the
+ execution of the "sip show peers" command. Review:
+ https://reviewboard.asterisk.org/r/1738/ (closes issue
+ ASTERISK-19231) (closes issue ASTERISK-19361) Reported by: Thomas
+ Arimont, Jamuel Starkey Tested by: Thomas Arimont, Jamuel Starkey
+ Patches: sip_show_peers_2012_02_16.diff uploaded by mjordan
+ (license 6283)
+
+2012-02-22 20:20 +0000 [r356290] Paul Belanger <pabelanger@digium.com>
+
+ * apps/app_rpt.c: Fix -Werror=unused-but-set-variable compiler
+ error (gcc 4.6.2) Review:
+ https://reviewboard.asterisk.org/r/1763/
+
+2012-02-22 21:08 +0000 [r356291] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/calendar.h, main/loader.c, res/res_calendar.c:
+ Track module use count for res_calendar If the res_calendar
+ module was followed immediately by one of the calendar tech
+ modules and "core stop gracefully" was run, Asterisk would crash.
+ This patch adds use count tracking for res_calendar so that it is
+ unloaded after the tech modules when shutting down gracefully. It
+ is now not possible to unload all the of the calendar modules via
+ "module unload res_calednar.so", but it is still possible to
+ unload them all via "module unload -h res_calendar.so". Review:
+ https://reviewboard.asterisk.org/r/1752/
+
+2012-02-22 21:29 +0000 [r356430-356335] Paul Belanger <pabelanger@digium.com>
+
+ * apps/app_rpt.c: Add back strsep() function for previous commit
+
+ * apps/app_rpt.c: Missed one strsep() function
+
+ * addons/chan_ooh323.c: Fix -Werror=unused-but-set-variable
+ compiler error (gcc 4.6.2)
+
+2012-02-23 15:37 +0000 [r356475] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix ACK routing for non-2xx responses. When
+ we send an ACK for a 2xx response to an INVITE, we are supposed
+ to use the learned route set. However, when we receive a non-2xx
+ final response to an INVITE, we are supposed to send the ACK to
+ the same place we initially sent the INVITE. We had been doing
+ this up until the changes went in that would build a route set
+ from provisional responses. That introduced a regression where we
+ would use the learned route set under all circumstances. With
+ this change, we now will set the destination of our ACK based on
+ the invitestate. If it is INV_COMPLETED then that means that we
+ have received a non-2xx final response (INV_TERMINATED indicates
+ a 2xx response was received). If it is INV_CANCELLED, then that
+ means the call is being canceled, which means that we should be
+ ACKing a 487 response. The other change introduced here is
+ setting the invitestate to INV_CONFIRMED when we send an ACK
+ *after* the reqprep instead of before. This way, we can tell in
+ reqprep more easily what the invitestate is prior to sending the
+ ACK. (closes issue ASTERISK-19389) reported by Karsten Wemheuer
+ patches: ASTERISK-19389v2.patch uploaded by Mark Michelson
+ (license #5049) (with some slight modifications prior to commit)
+
+2012-02-23 19:49 +0000 [r356521] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c, main/features.c: Fix blind transfer parking
+ issues if the dialed extension is not recognized as a parking
+ extension. Custom parking extensions may not be coded such that
+ the first and only extension priority is the Park application.
+ These custom parking extensions will not be recognized as parking
+ extensions. When a call is blind transferred to an extension that
+ is not recognized as a parking extension, the normal blind
+ transfer code causes the transferred channel to start executing
+ dialplan. Calls that get parked in this manner do not know the
+ original channel name that parked the call so the original parker
+ could never be called back if the parked call is not retrieved
+ before the timeout time. The parking space is also announced to
+ the call being parked as a side effect of not knowing the
+ original parking channel. * Fix handling of BLINDTRANSFER channel
+ variable for call parking. * Fixed SIP blind transfer using the
+ wrong dialplan context variable to check for the parking
+ extension. (closes issue ASTERISK-19322) Reported by: aragon
+ Tested by: rmudgett, jparker Review:
+ https://reviewboard.asterisk.org/r/1730/ JIRA AST-766
+
+2012-02-24 15:07 +0000 [r356650-356604] Matthew Jordan <mjordan@digium.com>
+
+ * include/asterisk/rtp_engine.h, res/res_srtp.c,
+ channels/sip/sdp_crypto.c, include/asterisk/res_srtp.h,
+ main/rtp_engine.c: Allow SRTP policies to be reloaded Currently,
+ when using res_srtp, once the SRTP policy has been added to the
+ current session the policy is locked into place. Any attempt to
+ replace an existing policy, which would be needed if the remote
+ endpoint negotiated a new cryptographic key, is instead rejected
+ in res_srtp. This happens in particular in transfer scenarios,
+ where the endpoint that Asterisk is communicating with changes
+ but uses the same RTP session. This patch modifies res_srtp to
+ allow remote and local policies to be reloaded in the underlying
+ SRTP library. From the perspective of users of the SRTP API, the
+ only change is that the adding of remote and local policies are
+ now added in a single method call, whereas they previously were
+ added separately. This was changed to account for the differences
+ in handling remote and local policies in libsrtp. Review:
+ https://reviewboard.asterisk.org/r/1741/ (closes issue
+ ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas
+ Arimont Patches: srtp_renew_keys_2012_02_22.diff uploaded by Matt
+ Jordan (license 6283) (with some small modifications for this
+ check-in)
+
+ * res/res_srtp.c: Remove srtp_shutdown from res_srtp The patch for
+ ASTERISK-19253 included properly shutting down the libsrtp
+ library in the case of module unload. Unfortunately, not all
+ distributions have the srtp_shutdown call. As such, this patch
+ removes calling srtp_shutdown.
+
+2012-02-24 18:23 +0000 [r356677] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/tcptls.h, channels/sip/include/sip.h,
+ channels/chan_sip.c: Fix worker thread resource leak in SIP
+ TCP/TLS. The SIP TCP/TLS worker threads were created joinable but
+ noone could join them if they died on their own. * Fix the SIP
+ TCP/TLS worker threads to not be created joinable. *
+ _sip_tcp_helper_thread() only needs one parameter since the pvt
+ parameter is only passed in as NULL and never used. (closes issue
+ ASTERISK-19203) Reported by: Steve Davies Review:
+ https://reviewboard.asterisk.org/r/1714/
+
+2012-02-25 17:21 +0000 [r356797] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_voicemail.c: Fix crash in app_voicemail during
+ close_mailbox In r354890, a memory leak in app_voicemail was
+ fixed by properly disposing of the allocated heard/deleted
+ pointers. However, there are situations, particularly when no
+ messages are found in a folder, where these pointers are not
+ allocated and not NULL. In that case, an invalid free would be
+ attempted, which could crash app_voicemail. As there are a number
+ of code paths where this could occur, this patch uses the number
+ of messages detected in the folder before it attempts to free the
+ pointers. This resolves the crash detected in the Asterisk Test
+ Suite's check_voicemail_nominal test.
+
+2012-02-27 15:14 +0000 [r356917] Jonathan Rose <jrose@digium.com>
+
+ * res/res_odbc.c: Remove possible segfaults from res_odbc by adding
+ locks around usage of odbc handle (closes issue ASTERISK-19011)
+ Reported by: Walter Doekes Patches:
+ issueA19011_combine_read_and_write_locks_WORK_IN_PROGRESS.patch
+ uploaded by Walter Doekes (license 5674) review:
+ https://reviewboard.asterisk.org/r/1719/ review:
+ https://reviewboard.asterisk.org/r/1622/
+
+2012-02-27 16:03 +0000 [r356963] Terry Wilson <twilson@digium.com>
+
+ * main/features.c: Copy CDR variables when set during a bridge This
+ patch makes sure amaflags, accountcode, and userfield get copied
+ to the bridge CDR when set during a bridge (like via a custom
+ feature). (closes issue ASTERISK-16990) Review:
+ https://reviewboard.asterisk.org/r/1721/
+
+2012-02-27 23:34 +0000 [r357093] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Fix callerid of Originated calls. Thanks to Matt
+ Riddell for tracking this down. (closes issue ASTERISK-19385)
+ Reported by: ornix
+
+2012-03-06 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.11.0-rc2 Released.
+
+2012-03-05 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.10.0 Released.
+
+2012-03-01 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.10.0-rc4 Released.
+
+ * main/acl.c: Prevent outbound SIP NOTIFY packets from displaying
+ a port of 0.
+
+ In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which
+ changed the behavior of ast_find_ourip such that port number was
+ wiped out. This caused the port in internip (which is used for
+ Contact and Call-ID on NOTIFYs) to be 0. This change causes
+ ast_find_ourip to be port-preserving again.
+
+2012-02-28 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.10.0-rc3 Released.
+
+ * main/channel.c: Fix callerid of Originated calls.
+
+ The callerid of originated calls (independent of mechanism) was not
+ being passed to the outbound channel. This patch fixes that. Thanks
+ to Matt Riddell for tracking this down.
+ (closes issue ASTERISK-19385)
+ Reported by: ornix
+
+ * channels/chan_sip.c: Fix ACK routing for non-2xx responses.
+
+ When we send an ACK for a 2xx response to an INVITE, we are supposed
+ to use the learned route set. However, when we receive a non-2xx
+ final response to an INVITE, we are supposed to send the ACK to the
+ same place we initially sent the INVITE.
+
+ We had been doing this up until the changes went in that would build
+ a route set from provisional responses. That introduced a regression
+ where we would use the learned route set under all circumstances.
+
+ With this change, we now will set the destination of our ACK based on
+ the invitestate. If it is INV_COMPLETED then that means that we have
+ received a non-2xx final response (INV_TERMINATED indicates a 2xx
+ response was received). If it is INV_CANCELLED, then that means the
+ call is being canceled, which means that we should be ACKing a 487
+ response.
+
+ The other change introduced here is setting the invitestate to
+ INV_CONFIRMED when we send an ACK *after* the reqprep instead of
+ before. This way, we can tell in reqprep more easily what the
+ invitestate is prior to sending the ACK.
+
+ (closes issue ASTERISK-19389)
+ reported by Karsten Wemheuer
+ patches:
+ ASTERISK-19389v2.patch uploaded by Mark Michelson (license #5049)
+
+ * channels/chan_sip.c: Fix regressions with regards to route-set
+ creation on early dialogs.
+
+ This fixes two main issues:
+ 1. Asterisk would send a CANCEL to the route created by the provisional
+ response instead of using the same destination it did in the initial
+ INVITE.
+ 2. If a new route set arrives in a 200 OK than was in the 1XX response
+ (perfectly possible if our outbound INVITE gets forked), then the
+ route set in the 200 OK needs to overwrite the route set in the 1XX
+ response.
+ (closes issue ASTERISK-19358)
+ Reported by: Karsten Wemheuer
+ Tested by: Karsten Wemheuer
+ patches:
+ ASTERISK-19358.patch uploaded by Mark Michelson (license 5049)
+ ASTERISK-19358.patch uploaded by Stefan Schmidt (license 6034)
+
+ Review: https://reviewboard.asterisk.org/r/1749
+
+2012-02-10 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.10.0-rc2 Released.
+
+ * channels/chan_sip.c: Fix SIP INFO DTMF handling for non-numeric
+ codes. In ASTERISK-18924, SIP INFO DTMF handling was changed to
+ account for both lowercase alphatbetic DTMF events, as well as
+ uppercase alphabetic DTMF events. When this occurred, the comparison
+ of the character buffer containing the event code was changed such
+ that the buffer was first compared against '0' and '9' to determine if
+ it was numeric. Unfortunately, since the first character in the
+ buffer will typically be '1' in the case of non-numeric event codes
+ (10-16), this caused those codes to be converted to a DTMF event of
+ '1'. This patch fixes that, and cleans up handling of both
+ application/dtmf-relay and application/dtmf content types.
+ Review: https://reviewboard.asterisk.org/r/1722/
+ (closes issue ASTERISK-19290) Reported by: Ira Emus
+ Tested by: mjordan
+
+ * apps/app_parkandannounce.c: Fix crash in ParkAndAnnounce from
+ uninitialized caller_id storage (closes issue ASTERISK-19311)
+ Reported by: tootai
+ Tested by: rmudgett
+
+ * channels/chan_agent.c: Fixes deadlocks occuring in chan_agent due to
+ r335976. Bad locking order was added to chan_agent to prevent
+ segfaults from having no locking in a patch by irroot. This patch
+ addresses the bad locking order by releasing locks before getting the
+ right locking order to stop deadlocks from occuring when doing
+ multiple interactions with agents. (closes issue ASTERISK-19285)
+ Reported by: Alex Villacis Lasso
+ Review: https://reviewboard.asterisk.org/r/1708/
+
+ * channels/chan_sip.c: Ensure entering T.38 passthrough does not cause
+ an infinite loop. After R340970 Asterisk was still polling the RTCP
+ file descriptor after RTCP is shut down and removed. If the
+ descriptor happened to have data ready when the removal occured then
+ Asterisk would go into an infinite loop trying to read data that it
+ can never actually access. This change disables the audio RTCP file
+ descriptor for the duration of the T.38 transaction. (closes issue
+ ASTERISK-18951) Reported-by: Kristijan Vrban
+
+ * channels/chan_sip.c,include/asterisk/dnsmgr.h,main/dnsmgr.c: Re-link
+ peers by IP when dnsmgr changes the IP Asterisk's dnsmgr currently
+ takes a pointer to an ast_sockaddr and updates it anytime an address
+ resolves to something different. There are a couple of issues with
+ this. First, the ast_sockaddr is usually the address of an ast_sockaddr
+ inside a refcounted struct and we never bump the refcount of those
+ structs when using dnsmgr. This makes it possible that a refresh could
+ happen after the destructor for that object is called (despite
+ ast_dnsmgr_release being called in that destructor). Second, the
+ module using dnsmgr cannot be aware of an address changing without
+ polling for it in the code. If an action needs to be taken on address
+ update (like re-linking a SIP peer in the peers_by_ip table), then
+ polling for this change negates many of the benefits of having dnsmgr
+ in the first place.
+
+2012-02-01 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.10.0-rc1 Released.
+
+ * Test results:
+ http://bamboo.asterisk.org/browse/TESTING-ASTERISK18100RCS-2
+
+2012-01-30 12:42 +0000 [r353260] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: Clarify log WARNING message when port-zero
+ SDP 'm' lines received. Previously, if an m-line in an SDP offer
+ or answer had a port number of zero, that line was skipped, and
+ resulted in an 'Unsupported SDP media type...' warning message.
+ This was misleading, as the media type was not unsupported, but
+ was ignored because the m-line indicated that the media stream
+ had been rejected (in an answer) or was not going to be used (in
+ an offer).
+
+2012-01-29 02:42 +0000 [r353175] Russell Bryant <russell@russellbryant.com>
+
+ * main/netsock.c: Find even more network interfaces. The previous
+ change made the code look for emN and pciN in addition to what it
+ did originally, which was search for ethN. However, it needed to
+ be looking for pciN#N, so that's what it does now. This also
+ moves the memset() to be before every ioctl().
+
+2012-01-28 14:49 +0000 [r353126] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/rtp_engine.c: Add 'L16-256' MIME subtype alias for slin16.
+ Asterisk has supported the 'L16' MIME subtype for 16kHz signed
+ linear (PCM) audio for quite some time, but some endpoints refer
+ to it as 'L16-256'. This commit adds this as an alias for the
+ existing format.
+
+2012-01-28 04:25 +0000 [r353077] Russell Bryant <russell@russellbryant.com>
+
+ * main/netsock.c: Update ast_set_default_eid() to find more network
+ interfaces. As of Fedora 15, ethN is not the name of ethernet
+ interfaces. The names are emN or pciN. Update some code that
+ searched for interfaces named ethN to look for the new names, as
+ well. For more information about why this change was made, see
+ this page: http://domsch.com/blog/?p=455
+
+2012-01-27 19:12 +0000 [r352959] Jonathan Rose <jrose@digium.com>
+
+ * res/res_monitor.c: Make failed PauseMonitor and UnpauseMonitor
+ with no valid channel not close AMI session. I also went ahead
+ and took a little time to make sure that the manager value
+ AMI_SUCCESS was used instead of just return 0 being thrown around
+ everywhere since that's how we handle this stuff these days.
+ (closes issue ASTERISK-19249) Reporter: Jamuel Starkey Patches:
+ res_monitor.c-ASTERISK-19249.diff uploaded by Jamuel Starkey
+ (license 5766)
+
+2012-01-27 18:22 +0000 [r352955] Richard Mudgett <rmudgett@digium.com>
+
+ * res/snmp/agent.c, main/taskprocessor.c, apps/app_queue.c,
+ channels/chan_iax2.c, apps/app_chanspy.c, main/indications.c,
+ res/res_odbc.c, res/res_srtp.c, main/pbx.c, channels/chan_sip.c,
+ include/asterisk/indications.h: Audit of ao2_iterator_init()
+ usage for v1.8. Fixes numerous reference leaks and missing
+ ao2_iterator_destroy() calls as a result. Review:
+ https://reviewboard.asterisk.org/r/1697/
+
+2012-01-27 00:05 +0000 [r352862] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: rfc4235 -
+ Section 4.1: Versions MUST be representable using a non-negative
+ 32 bit integer. If a BLF subscription exists for long enough,
+ using %d may print negative version numbers. Unlikely, as 2^32 at
+ 1 update per second is ~137 years, or half that before the
+ versions number started going negative. Tested with Asterisk
+ 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested
+ by: alecdavis Review: https://reviewboard.asterisk.org/r/1694/
+
+2012-01-26 20:14 +0000 [r352807] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c: Fix outbound DTMF for inband mode (tell
+ asterisk core to generate DTMF sounds). (Closes issue
+ ASTERISK-19233) Reported by: Matt Behrens Patches:
+ chan_ooh323.c.patch uploaded by Matt Behrens (License #6346)
+
+2012-01-26 19:06 +0000 [r352755] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c: Copy amaflags to sip_pvt from peer during
+ create_addr_from_peer For whatever reason, we don't have a single
+ function for copying data like this from SIP peers to the SIP
+ pvt. This patch adds the copying of amaflags to the sip_pvt, but
+ it would probably be worth discussing this function along with
+ the others that essentially just copy some amount of data from a
+ peer to a private. (Closes issue ASTERISK-19029) Reported by:
+ Matt Lehner
+
+2012-01-26 06:27 +0000 [r352704] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/chan_sip.c: Cleanup dialog-info+xml Notify dialog Make
+ similar to other Notify messages. sample output: <?xml
+ version="1.0"?> <dialog-info
+ xmlns="urn:ietf:params:xml:ns:dialog-info" version="715"
+ state="full" entity="sip:8523@192.168.x.xx"> <dialog id="8523">
+ <state>terminated</state> </dialog> </dialog-info> Tested with
+ Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585)
+ Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1693/
+
+2012-01-25 22:21 +0000 [r352643] Paul Belanger <pabelanger@digium.com>
+
+ * apps/app_voicemail.c: Fix -Werror=unused-but-set-variable
+ compiler error (gcc 4.6.2)
+
+2012-01-25 21:16 +0000 [r352612] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/test.c: Avoid unnecessary rebuilds of main/test.c.
+ main/test.c includes "asterisk/version.h", when it should include
+ "asterisk/ast_version.h" instead (and it should use the
+ ast_get_version() and ast_get_version_num() functions). This
+ commit modifies it to extract the Asterisk version information
+ using the proper APIs, and as a result means that main/test.c no
+ longer needs to be rebuilt when a Subversion checkout is updated
+ or modified.
+
+2012-01-25 17:28 +0000 [r352514-352551] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Remove some extraneous debugging from
+ registry memleak fix
+
+ * channels/chan_sip.c: Clean up some SIP registry-related memory
+ leaks 1) Be sure and free at unload the epa_backend we allocate
+ at startup 2) Do the same sip_registry cleanup at unload we do at
+ reload Review: https://reviewboard.asterisk.org/r/1689/
+
+2012-01-25 16:39 +0000 [r352511] Jonathan Rose <jrose@digium.com>
+
+ * configs/sip.conf.sample: Redocuments sip types peer, user, friend
+ in sip.conf.sample There was faulty information in the sample
+ config describing user as a synonym for friend so it has been
+ changed to better elaborate on the differences between the three
+ entity types. (closes issue ASTERISK-15537) Reported by: yarique
+
+2012-01-24 22:17 +0000 [r352424] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Don't do a DNS lookup on an outbound
+ REGISTER host if there is an outbound proxy configured. (closes
+ issue ASTERISK-16550) reported by: Olle Johansson
+
+2012-01-24 20:33 +0000 [r352367] Jonathan Rose <jrose@digium.com>
+
+ * sounds/Makefile: Set core sounds version to 1.4.22. Now that we
+ have the right license for the Russian 1.4.22 sounds as well as
+ the sounds for the Australian English 1.4.22 sounds, we can
+ finally set the sounds to use 1.4.22! (closes issue
+ ASTERISK-18978) Reported by: Cameron Twomey Patches:
+ confbridge.tar.001 uploaded by Cameron Twomey confbridge.tar.002
+ uploaded by Cameron Twomey
+
+2012-01-24 16:59 +0000 [r352291] Richard Mudgett <rmudgett@digium.com>
+
+ * funcs/func_odbc.c: Fix locking issues with channel datastores in
+ func_odbc.c. * Fixed a potential memory leak when an existing
+ datastore is manually destroyed by inline code instead of calling
+ ast_datastore_free(). (closes issue ASTERISK-17948) Reported by:
+ Archie Cobbs Review: https://reviewboard.asterisk.org/r/1687/
+
+2012-01-24 16:30 +0000 [r352287] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Move RTP timeout check to before bridged
+ channel check so it is actually executed. (issue ASTERISK-19179)
+ Reported by: TSAREGORODTSEV Yury (closes issue ASTERISK-14534)
+ Reported by: kriborgen Patches: chan_sip.patch uploaded by
+ kriborgen (license 6138)
+
+2012-01-23 20:30 +0000 [r352199-352230] Mark Michelson <mmichelson@digium.com>
+
+ * main/features.c: Fix grammar of comment.
+
+ * main/features.c: Fix blind transfers from failing if an 'h'
+ extension is present. This prevents the 'h' extension from being
+ run on the transferee channel when it is transferred via a native
+ transfer mechanism such as SIP REFER. (closes ASTERISK-19173)
+ Reported by: Ross Beer Tested by: Kristjan Vrban Patches:
+ ASTERISK-19173 by Mark Michelson (license 5049) Review:
+ https://reviewboard.asterisk.org/r/1685
+
+2012-01-23 19:12 +0000 [r352144] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_fax_spandsp.c: Correctly apply FAXOPT settings (V17, V27,
+ V29) before starting spandsp layer While the FAXOPT function
+ could be used to set the modem capabilities, the input to that
+ function was not being applied correctly to the spandsp layer.
+ This patch applies the current model capabilities before starting
+ the spandsp layer. (closes issue: ASTERISK-16409) Reported by:
+ Kristijan Vrban Tested by: Matt Jordan, Matthew Nicholson
+ Patches: spandsp-modems-1.8.diff uploaded by mnicholson (license
+ 5081) spandsp-modems-10.diff uploaded by mnicholson (license
+ 5081)
+
+2012-01-23 17:33 +0000 [r352090] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Fix sip_cfg.notifycid to be set with the
+ defined enum values. The invalid value used when notifycid was
+ enabled was benign. As far as the code was concerned -1 and 1 are
+ equivalent. (closes issue ASTERISK-19232) Reported by: Eike
+ Kuiper
+
+2012-01-21 00:20 +0000 [r352029] Richard Mudgett <rmudgett@digium.com>
+
+ * main/app.c, funcs/func_timeout.c: Fix ast_app_dtget() time unit
+ inconsistency. Note: Noone calls ast_app_dtget() with the timeout
+ parameter of zero so the bad code normally will never get
+ executed. * Fix unnecessary floating point division in
+ func_timeout.c timeout_write() when all other values are
+ integers. (closes issue ASTERISK-16817) Reported by: Dmitry
+ Andrianov
+
+2012-01-21 00:08 +0000 [r352014-352016] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Remove XXX comment that is not necessary.
+
+ * channels/chan_sip.c: Fix RTP reference leak. If a blind transfer
+ were initiated using a REFER without a prior reINVITE to place
+ the call on hold, AND if Asterisk were sending RTCP reports, then
+ there was a reference for the RTP instance of the transferer.
+ This fixes the issue by merging two similar but slightly
+ conflicting sections of code into a single area. It also adds a
+ stop_media_flows() call in the case that the transferer's UA
+ never sends a BYE to us like it is supposed to. (issue
+ ASTERISK-19192) Review: https://reviewboard.asterisk.org/r/1681/
+
+2012-01-20 19:34 +0000 [r351858-351860] Kinsey Moore <kmoore@digium.com>
+
+ * codecs/ilbc/iLBC_test.c: More corrections for the ilbc code These
+ changes are in a file that is not compiled by default, and so
+ were missed on earlier checks.
+
+ * codecs/ilbc/LPCencode.c, codecs/ilbc/iLBC_decode.c: Allow ilbc
+ code to build under dev mode GCC 4.6.3 found some set/unused
+ variables in the ILBC code.
+
+2012-01-20 16:01 +0000 [r351765] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c: Accidentally left off a semicolon only in
+ 1.8 somehow for previous patch.
+
+2012-01-20 15:48 +0000 [r351760] Matthew Jordan <mjordan@digium.com>
+
+ * codecs/ilbc/helpfun.c: Remove unused variable 'tmp' from helpfun
+ in ilbc codec gcc version 4.6.2 caught an unused variable in the
+ ilbc codec library. This would prevent compilation with
+ --enable-dev-mode; variable removed.
+
+2012-01-20 15:42 +0000 [r351759] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c: Adds setting of mwi_from field to
+ check_auth_result check_peer_ok (closes ASTERISK-19057) Reported
+ By: Yuri Patches: 348360chan_sip.diff uploaded by Yuri (license
+ 5242)
+
+2012-01-20 12:59 +0000 [r351707] Stefan Schmidt <sst@sil.at>
+
+ * contrib/asterisk-ng-doxygen: enable doxygen build for files in
+ the channels/sip folder like reqresp_parser.c
+
+2012-01-19 23:17 +0000 [r351618] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c, channels/sip/reqresp_parser.c: Misc minor
+ fixes in reqresp_parser.c and chan_sip.c. * Fix corner cases in
+ get_calleridname() parsing and ensure that the output buffer is
+ nul terminated. * Make get_calleridname() truncate the name it
+ parses if the given buffer is too small rather than abandoning
+ the parse and not returning anything for the name. Adjusted
+ get_calleridname_test() unit test to handle the truncation
+ change. * Fix get_in_brackets_test() unit test to check the
+ results of get_in_brackets() correctly. * Fix
+ parse_name_andor_addr() to not return the address of a local
+ buffer. This function is currently not used. * Fix potential NULL
+ pointer dereference in sip_sendtext(). * No need to
+ memset(calleridname) in check_user_full() or tmp_name in
+ get_name_and_number() because get_calleridname() ensures that it
+ is nul terminated. * Reply with an accurate response if
+ get_msg_text() fails in receive_message(). This is academic in
+ v1.8 because get_msg_text() can never fail.
+
+2012-01-19 22:36 +0000 [r351611] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_rtp_asterisk.c: Correct output of RTCP jitter statistics
+ in SR and RR reports Change the RTCP RR and SR generation code to
+ convert Asterisk's internal jitter statistics to be represented
+ in RTP timestamp units based on the rate of the codec in use
+ instead of in seconds. (closes issue ASTERISK-14530)
+
+2012-01-19 21:46 +0000 [r351559] Jonathan Rose <jrose@digium.com>
+
+ * include/asterisk/netsock2.h, channels/chan_sip.c: Eliminates
+ doubling the :port part of SIP Notify Message-Account headers.
+ This patch prevents the domain string from getting mangled during
+ the initreqprep step by moving the initialization to before its
+ immediate use. It also documents this pitfall for the
+ ast_sockaddr_stringify functions. (issue ASTERISK-19057) Reported
+ by: Yuri Review: https://reviewboard.asterisk.org/r/1678/
+
+2012-01-19 21:11 +0000 [r351504] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Prevent crash when an SDP offer is received
+ with an encrypted video stream when support for video is disabled
+ and res_srtp is loaded. (closes issue ASTERISK-19202) Reported
+ by: Catalin Sanda
+
+2012-01-18 20:54 +0000 [r351450] Matthew Jordan <mjordan@digium.com>
+
+ * codecs/ilbc/StateConstructW.c (added), codecs/ilbc/packing.c
+ (added), codecs/ilbc/StateConstructW.h (added),
+ codecs/ilbc/packing.h (added), codecs/ilbc/getCBvec.c (added),
+ codecs/ilbc/LPCdecode.c (added), codecs/ilbc/enhancer.c (added),
+ codecs/ilbc/lsf.c (added), codecs/ilbc/iLBC_encode.c (added),
+ codecs/ilbc/getCBvec.h (added), codecs/ilbc/LPCdecode.h (added),
+ codecs/ilbc/enhancer.h (added), codecs/ilbc/FrameClassify.c
+ (added), codecs/ilbc/iLBC_define.h (added), codecs/ilbc/lsf.h
+ (added), codecs/ilbc/extract-cfile.awk (added),
+ codecs/ilbc/iLBC_encode.h (added), codecs/ilbc/Makefile,
+ codecs/ilbc/FrameClassify.h (added), codecs/ilbc/helpfun.c
+ (added), codecs/ilbc/LICENSE_ADDENDUM (added),
+ codecs/ilbc/doCPLC.c (added), codecs/ilbc/anaFilter.c (added),
+ codecs/ilbc/helpfun.h (added), codecs/ilbc/createCB.c (added),
+ codecs/ilbc/doCPLC.h (added), codecs/ilbc/anaFilter.h (added),
+ codecs/ilbc/constants.c (added), codecs/ilbc/iLBC_decode.c
+ (added), codecs/ilbc/createCB.h (added), codecs/ilbc/constants.h
+ (added), codecs/ilbc/iLBC_decode.h (added),
+ codecs/ilbc/iCBSearch.c (added), codecs/ilbc/filter.c (added),
+ codecs/ilbc/hpInput.c (added), codecs/ilbc/gainquant.c (added),
+ codecs/ilbc/iCBSearch.h (added), codecs/ilbc/hpOutput.c (added),
+ codecs/ilbc/rfc3951.txt (added), codecs/ilbc/filter.h (added),
+ codecs/ilbc/gainquant.h (added), codecs/ilbc/LPCencode.c (added),
+ codecs/ilbc/hpInput.h (added), codecs/codec_ilbc.c,
+ codecs/ilbc/PATENTS (added), codecs/ilbc/StateSearchW.c (added),
+ codecs/ilbc/hpOutput.h (added),
+ contrib/scripts/get_ilbc_source.sh, codecs/ilbc/LICENSE (added),
+ codecs/ilbc/LPCencode.h (added), codecs/ilbc/StateSearchW.h
+ (added), codecs/ilbc/iCBConstruct.c (added),
+ codecs/ilbc/syntFilter.c (added), codecs/ilbc/iCBConstruct.h
+ (added), codecs/ilbc/iLBC_test.c (added),
+ codecs/ilbc/syntFilter.h (added): Include iLBC source code for
+ distribution with Asterisk This patch includes the iLBC source
+ code for distribution with Asterisk. Clarification regarding the
+ iLBC source code was provided by Google, and the appropriate
+ licenses have been included in the codecs/ilbc folder. Review:
+ https://reviewboard.asterisk.org/r/1675 Review:
+ https://reviewboard.asterisk.org/r/1649 (closes issue:
+ ASTERISK-18943) Reporter: Leif Madsen Tested by: Matt Jordan
+
+2012-01-18 14:57 +0000 [r351396] Stefan Schmidt <sst@sil.at>
+
+ * channels/chan_sip.c: The get_pai function in chan_sip.c didn't
+ recognized a proper callerid name and number from a
+ P-Asserted-Identity cause the header parsing logic was wrong.
+ Changing the parsing functions to the sip header parsing APIs in
+ reqresp_parser.h solves this problem. Review:
+ https://reviewboard.asterisk.org/r/1673 Reviewed by: wdoekes2 and
+ Mark Michelson
+
+2012-01-17 17:22 +0000 [r351306] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_rtp_asterisk.c: Eliminate odd initialization of probation
+ variable.
+
+2012-01-17 16:55 +0000 [r351287] Jonathan Rose <jrose@digium.com>
+
+ * CHANGES, res/res_rtp_asterisk.c, configs/rtp.conf.sample: Adds
+ pjmedia probation concepts to res_rtp_asterisk's learning mode.
+ In order to better handle RTP sources with strictrtp enabled
+ (which is now default in 10) using the learning mode to figure
+ out new sources when they change is handled by checking for a
+ number of consecutive (by sequence number) packets received to an
+ rtp struct based on a new configurable value called 'probation'.
+ Also, during learning mode instead of liberally accepting all
+ packets received, we now reject packets until a clear source has
+ been determined. Review: https://reviewboard.asterisk.org/r/1663/
+
+2012-01-17 16:41 +0000 [r351284] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Use built-in parsing functions for Contact
+ and Record-Route headers. If a Contact or a Record-Route header
+ had a quoted string with an item in angle brackets, then we would
+ mis-parse it. For instance, "Bob <1234>" <1234@example.org> would
+ be misparsed as having the URI "1234" The fix for this is to use
+ parsing functions from reqresp_parser.h since they are heavily
+ tested and are awesome. (issue ASTERISK-18990)
+
+2012-01-17 16:06 +0000 [r351233] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_sip.c: Fix udptl issue with initial INVITE
+ introduced by r351027 When an inital INVITE occurs that contains
+ image media, a channel is not yet associated with the SIP dialog.
+ The file descriptor associated with the udptl session needs to be
+ set in initialize_udptl or in sip_new to account for this
+ scenario.
+
+2012-01-17 01:37 +0000 [r351182] Russell Bryant <russell@russellbryant.com>
+
+ * channels/chan_sip.c: Add some missing locking in chan_sip. This
+ patch adds some missing locking to the function
+ send_provisional_keepalive_full(). This function is called from
+ the scheduler, which is processed in the SIP monitor thread. The
+ associated channel (or pbx) thread will also be using the same
+ sip_pvt and ast_channel so locking must be used. The
+ sip_pvt_lock_full() function is used to ensure proper locking
+ order in a safe manner. In passing, document a suspected
+ reference counting error in this function. The "fix" is left
+ commented out because when the "fix" is present, crashes occur.
+ My theory is that fixing it is exposing a reference counting
+ error elsewhere, but I don't know where. (Or my analysis of this
+ being a problem could have been completely wrong in the first
+ place). Leave the comment in the code for so that someone may
+ investigate it again in the future. Also add a bit of doxygen to
+ transmit_provisional_response(). (closes issue ASTERISK-18979)
+ Review: https://reviewboard.asterisk.org/r/1648
+
+2012-01-16 21:12 +0000 [r351080-351130] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Ensure ACK retransmit & hangup on non-200
+ response to INVITE When handling a non-2xx final response on an
+ INVITE transaction, we have to keep the transaction around after
+ we send an ACK in case we receive a retransmission of the
+ response so we can re-transmit the ACK, but also tear down the
+ ast_channel as soon as we transmit the ACK. Before this patch, we
+ could fail at both of these things. Calling
+ sip_alreadygone/needdestroy prevented us from keeping the
+ transaction up and retransmitting the ACK, and queueing
+ CONGESTION was not sufficient to cause the channel to be torn
+ down when originating calls via the CLI, for example. This patch
+ queues a hangup with CONGESTION instead of just queueing
+ CONGESTION for these responses and removes the sip_alreadygone
+ and sip_needdestroy calls from handle_response_invite on non-2xx
+ responses. It relies on the hangup calling sip_scheddestroy. For
+ more information, see section 17.1.1.1 of RFC 3261. (closes issue
+ ASTERISK-17717) Review: https://reviewboard.asterisk.org/r/1672/
+
+ * channels/chan_sip.c: Don't prematurely stop SIP session timer
+ When Asterisk is the UAS (incoming call, endpoint is re-inviting)
+ the SIP session timer expires after half the time the sip
+ endpoint indicates in the Session-expires header in
+ proc_session_timer(). The session timer was being stopped totally
+ and being handled as an error case instead of running again until
+ the second expiry. This patch treats the half-time expiry as a
+ non-error case and continues the timer until the true expiry.
+ (closes issue ASTERISK-18996) Reported by: Thomas Arimont Tested
+ by: Thomas Arimont Patches: session_timer_fix.diff by Terry
+ Wilson (License #5357) based on session_timer.patch by Thomas
+ Arimont (License #5525)
+
+2012-01-16 19:09 +0000 [r351027] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_sip.c: Create and initialize udptl only when dialog
+ negotiates for image media Prior to this patch, the udptl struct
+ was allocated and initialized when a dialog was associated with a
+ peer that supported T.38, when a new SIP channel was allocated,
+ or what an INVITE request was received. This resulted in any
+ dialog associated with a peer that supported T.38 having udptl
+ support assigned to it, including the UDP ports needed for
+ communication. This occurred even in non-INVITE dialogs that
+ would never send image media. This patch creates and initializes
+ the udptl structure only when the SDP for a dialog specifies that
+ image media is supported, or when Asterisk indicates through the
+ appropriate control frame that a dialog is to support T.38.
+ (closes issue ASTERISK-16698) Reported by: under Tested by:
+ Stefan Schmidt Patches: udptl_20120113.diff uploaded by mjordan
+ (License #6283) (closes issue ASTERISK-16794) Reported by: Elazar
+ Broad Tested by: Stefan Schmidt review:
+ https://reviewboard.asterisk.org/r/1668/
+
+2012-01-16 17:04 +0000 [r350975] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp_engine.c: Add missing code to set direct RTP setup
+ information during dialing.
+
+2012-01-15 20:07 +0000 [r350885-350888] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * main/asterisk.c: Allow only one thread at a time to do asterisk
+ cleanup/shutdown. Add locking around the really-really-quit part
+ of the core stop/restart part. Previously more than one thread
+ could be called to do cleanup, causing atexit handlers to be run
+ multiple times, in turn causing segfaults. (issue ASTERISK-18883)
+ Reviewed by: Terry Wilson Review:
+ https://reviewboard.asterisk.org/r/1662/ Review:
+ https://reviewboard.asterisk.org/r/1658/
+
+ * utils/extconf.c: Fix -Werror=unused-but-set-variable compile
+ error in utils/extconf.c. Note that I'm not confirming legitimacy
+ of having that file in tree at all. Is anyone using
+ aelparse/conf2ael? (issue ASTERISK-15350)
+
+2012-01-14 16:40 +0000 [r350788-350837] Kevin P. Fleming <kpfleming@digium.com>
+
+ * autoconf/libcurl.m4, configure, autoconf/ast_gcc_attribute.m4,
+ configure.ac: Ensure that all AC_LANG_PROGRAM calls in the
+ configure script are properly quoted. Recent versions of autoconf
+ (2.68 on my system) won't properly process the configure script
+ unless every call to AC_LANG_PROGRAM is m4-quoted. Many calls in
+ the script were, but many were not. This patch corrects the
+ unquoted calls.
+
+ * addons/chan_mobile.c, channels/chan_h323.c: Correct some
+ 'set-but-not-used' variable warnings.
+
+ * contrib/scripts/install_prereq: Ensure that two prerequisites are
+ properly installed on Debian-style distributions. * Don't specify
+ a specific version of libgmime; newer versions are available now
+ and acceptable. * Install libsrtp so that res_srtp can be built.
+
+2012-01-13 22:05 +0000 [r350736] Kinsey Moore <kmoore@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in: Run bootstrap.sh for
+ the for the ASTERISK-18929 fix configure and autoconfig.h.in were
+ not regenerated when the fix was committed.
+
+2012-01-13 21:51 +0000 [r350733] Richard Mudgett <rmudgett@digium.com>
+
+ * configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample:
+ Correct eventtype names in cel_odbc and cel_pgsql sample files
+
+2012-01-13 21:40 +0000 [r350730] Kinsey Moore <kmoore@digium.com>
+
+ * bootstrap.sh, main/asterisk.c, configure.ac: Make sure asterisk
+ builds on OpenBSD OpenBSD defines SO_PEERCRED, but it returns a
+ 'struct sockpeercred', not 'struct ucred', which causes
+ compilation of main/asterisk.c to fail in read_credentials().
+ This allows configure to check for sockpeercred and asterisk to
+ deal with it properly. (closes issue ASTERISK-18929) Reported-by:
+ Barry Miller Patch-by: Barry Miller
+
+2012-01-13 20:29 +0000 [r350679] Mark Michelson <mmichelson@digium.com>
+
+ * channels/sip/config_parser.c: Set port to a default sane value if
+ a bogus one is provided when parsing hostnames.
+
+2012-01-13 17:23 +0000 [r350555-350571] Richard Mudgett <rmudgett@digium.com>
+
+ * configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample,
+ cel/cel_pgsql.c, cel/cel_odbc.c, cel/cel_manager.c: Use
+ compatible names for event extra data for various CEL backends. *
+ Change eventextra to extra in cel_psql.c and cel_odbc.c. * Change
+ EventExtra to Extra in cel_manager.c. (issue ASTERISK-17190)
+
+ * configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample,
+ main/cel.c, configs/cel_custom.conf.sample, cel/cel_pgsql.c,
+ configs/cel_sqlite3_custom.conf.sample, cel/cel_odbc.c,
+ configs/cel.conf.sample, cel/cel_manager.c: Add missing CEL
+ logging fields to various CEL backends. * Add missing eventextra
+ to cel_psql.c and cel_odbc.c. * Add missing PeerAccount and
+ EventExtra to cel_manager.c. * Add missing userdeftype support
+ for cel_custom.conf.sample and cel_sqlite3_custom.conf.sample.
+ (closes issue ASTERISK-17190) Reported by: Bryant Zimmerman
+
+2012-01-13 16:57 +0000 [r350552] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_queue.c: Realtime queues failed to load queue
+ information without queue member table Previously, realtime
+ queues could be loaded without defining the queue member table.
+ This allowed for queue members to be dynamic, while the realtime
+ queue definitions could exist in some backing storage. Revision
+ 342223 broke this when it changed the return value for
+ realtime_multientry to return NULL when no results are returned.
+ Previously, an empty ast_config object was expected. (closes
+ issue ASTERISK-19170) Reported by: Rene Mendoza Tested by: Rene
+ Mendoza Patches: rt_queue_member_patch.diff uploaded by Matt
+ Jordan (license 6283)
+
+2012-01-12 15:57 +0000 [r350501] Jonathan Rose <jrose@digium.com>
+
+ * main/features.c: Adds peer to CEL report on CEL_BRIDGE_START and
+ CEL_BRIDGE_END (closes issue ASTERISK-17940) Reporter: Nic
+ Colledge Patches: features_18.patch uploaded by Nic Colledge
+ (license 6245)
+
+2012-01-11 22:50 +0000 [r350311-350452] Richard Mudgett <rmudgett@digium.com>
+
+ * main/cel.c: Remove extraneous BRIDGEPEER AMI VarSet event on a
+ CEL dummy channel. (closes issue ASTERISK-19180) Reported by:
+ Corey Farrell Patches: asterisk_cel_noevent_varset.diff (license
+ #5909) patch uploaded by Corey Farrell
+
+ * CHANGES, apps/app_followme.c, apps/app_dial.c: Make FollowMe
+ optionally update connected line information when the accepting
+ endpoint is bridged. Like Dial and Queue, FollowMe needs to deal
+ with AST_CONTROL_CONNECTED_LINE information so when the parties
+ are initially bridged, the connected line information will be
+ correct. * Added the 'I' option just like the app_dial and
+ app_queue 'I' option. (closes issue ASTERISK-18969) Reported by:
+ rmudgett Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1656/
+
+ * funcs/func_lock.c: Fix absolute/relative time mismatch in LOCK
+ function. The time passed by the LOCK function to an internal
+ function was relative time when the function expected absolute
+ time. * Don't use C++ keywords in get_lock(). (closes issue
+ ASTERISK-16868) Reported by: Andrey Solovyev Patches:
+ 20101102__issue18207.diff.txt (license #5003) patch uploaded by
+ Andrey Solovyev (modified)
+
+2012-01-09 21:54 +0000 [r350075-350220] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_iax2.c: Fix joinable thread terminating without
+ joiner memory leak in chan_iax.c. The iax2_process_thread() can
+ exit without anyone waiting to join the thread. If noone is
+ waiting to join the thread then a large memory leak occurs. *
+ Made iax2_process_thread() deatach itself if nobody is waiting to
+ join the thread. (closes issue ASTERISK-17339) Reported by:
+ Tzafrir Cohen Patches:
+ asterisk-1.8.4.4-chan_iax2-detach-thread-on-non-stop-exit.patch
+ (license #5617) patch uploaded by Alex Villacis Lasso (modified)
+ (closes issue ASTERISK-17825) Reported by: wangjin
+
+ * contrib/scripts/live_ast: live_ast: valgrind: run asterisk under
+ valgrind Adds a new sub-command, "valgrind" to live_ast. It runs
+ asterisk under valgrind. The extra command-line parameters are
+ passed to Asterisk as usual, and parameters to valgrind are
+ passed through LIVE_AST_VALGRIND_ARGS in live.conf . Review:
+ https://reviewboard.asterisk.org/r/1109/ Merged revisions 326636
+ from http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * contrib/scripts/live_ast, contrib/scripts/valgrind_compare
+ (added): Update contrib script live_ast to invoke Asterisk with
+ valgrind and suppression file. * Added valgrind_compare script to
+ compare two valgrind log files for differences. (issue
+ ASTERISK-17339) Reported by: Tzafrir Cohen Patches:
+ valgrind_compare (license #5035) script uploaded by Tzafrir Cohen
+ live_ast_valgrind.diff (license #5035) patch uploaded by Tzafrir
+ Cohen live_ast_valgrind_v2.diff (license #5185) patch uploaded by
+ Paul Belanger
+
+ * main/asterisk.c: Make Asterisk -x command line parameter imply -r
+ parameter presence. The Asterisk -x command line parameter is
+ documented inconsistently. * Made the -x documentation and
+ behavior consistent. * Since this is also a new year, updated the
+ copyright notices while here. (closes issue ASTERISK-19094)
+ Reported by: Eugene Patches:
+ issueA19094_correct_asterisk_option_x.patch (license #5674) patch
+ uploaded by Walter Doekes (modified) Tested by: Eugene
+
+2012-01-09 15:37 +0000 [r350023] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_meetme.c: Prevent SLA settings from getting wiped out on
+ reload If SLA was reloaded without the config file being changed,
+ current settings got wiped out before the SLA reload code decided
+ it wasn't going to reload the file since nothing was changed.
+ Moving the settings reset later in the reload process fixes this.
+ (closes issue AST-744)
+
+2012-01-06 23:17 +0000 [r349968] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Don't leak CID in From header when
+ presentation=unavailable When someone does
+ Set(CALLERPRES()=unavailable) (or
+ Set(CALLERID(pres)=unavailable)) when sendrpid=no, the From
+ header shows "Anonymous" <anonymous@anonymous.invalid>. When
+ sendrpid=yes/pai, the From header will still display the callerid
+ info, even though we supply an rpid header with the anonymous
+ info. It seems like we shouldn't leak that info in any case.
+ Skimming http://tools.ietf.org/html/draft-ietf-sip-privacy-04
+ seems to indicate that one shouldn't send identifying info in the
+ From in this case. This patch anonymizes the From header as well
+ even when sendrpid=yes/pai. (closes issue ASTERISK-16538) Review:
+ https://reviewboard.asterisk.org/r/1649/
+
+2012-01-06 16:46 +0000 [r349819-349872] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_followme.c: Fix memory leaks in app_followme
+ find_realtime(). (closes issue ASTERISK-19055) Reported by: Matt
+ Jordan
+
+ * cel/cel_sqlite3_custom.c: Make not assume that the
+ cel_sqlite3_custom SQL table primary key is AcctId. If a table is
+ created by some other application and the primary key is not
+ named "AcctId", cel/cel_sqlite3_custom.c will always try to
+ create the table and fail because it already exists. * Change the
+ SQL table query to not require AcctId as the primary key. (closes
+ issue ASTERISK-18963) Reported by: socketpair Patches: fix.patch
+ (license #6337) patch uploaded by socketpair
+
+2012-01-05 22:06 +0000 [r349731] Kinsey Moore <kmoore@digium.com>
+
+ * main/file.c: Allow playback of formats that don't support seeking
+ ast_streamfile previously did unconditional seeking on files that
+ broke playback of formats that don't support that functionality.
+ This patch avoids the seek that was causing the problem. This
+ regression was introduced in r158062. (closes issue
+ ASTERISK-18994) Patch-by: Timo Teras
+
+2012-01-05 21:46 +0000 [r349672-349728] Jonathan Rose <jrose@digium.com>
+
+ * main/dsp.c: Fix an issue where dsp.c would interpret multiple
+ dtmf events from a single key press. When receiving calls from a
+ mobile phone into a DISA system on a connection with significant
+ interference, the reporter's Asterisk system would interpret DTMF
+ incorrectly and replicate digits received. This patch resolves
+ that by increasing the number of frames a mismatch has to be
+ detected before assuming the DTMF is over by 1 frame and adjusts
+ dtmf_detect function to reset hits and misses only when an edge
+ is detected. (closes issue ASTERISK-17493) Reported by: Alec
+ Davis Patches: bug18904-refactor.diff.txt uploaded by Alec Davis
+ (license 5546) Review: https://reviewboard.asterisk.org/r/1130/
+
+ * main/asterisk.c: Ensures Asterisk closes when receiving terminal
+ signals in 'no fork' mode. When catching a signal, in no fork
+ mode the console thread is identical to the thread responsible
+ for catching the signal and closing Asterisk, which requires it
+ to first dispense with the console thread. Prior to this patch,
+ if these threads were identical, upon receiving a killing signal,
+ the thread will send an URG signal to itself, which we also catch
+ and then promptly do nothing with. Obviously this isn't useful
+ behavior. (closes issue ASTERISK-19127) Reported By: Bryon Clark
+ Patches: quit_on_signals.patch uploaded by Bryon Clark (license
+ 6157)
+
+2012-01-04 20:46 +0000 [r349558] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Fix segfault in chan_dahdi for
+ CHANNEL(dahdi_span) evaluation on hangup. * Added NULL private
+ pointer checks in the following chan_dahdi channel callbacks:
+ dahdi_func_read(), dahdi_func_write(), dahdi_setoption(), and
+ dahdi_queryoption(). (closes issue ASTERISK-19142) Reported by:
+ Diego Aguirre Tested by: rmudgett
+
+2012-01-04 20:23 +0000 [r349504-349529] Kinsey Moore <kmoore@digium.com>
+
+ * contrib/init.d/rc.debian.asterisk: Make debian init script
+ conform to the LSB standard Previously, this init script would
+ return 1 if Asterisk was already running. This is incorrect
+ behavior according to the LSB standard and has been fixed by
+ returning 0 instead. (closes issue ASTERISK-17958) Reported-by:
+ johnc
+
+ * contrib/scripts/autosupport, contrib/scripts/autosupport.8:
+ Update autosupport script and man page Added information
+ collection from the output of the utilities: top, free, uptime,
+ ifconfig Added information collection from the output of the
+ Asterisk command 'dahdi show status' Added option / flag '-n,
+ --non-interactive' Updated man page to reflect new option / flag
+ '-n, --non-interactive' Patch-by: John Bigelow (itzanger) (closes
+ issue AST-749)
+
+2012-01-04 19:27 +0000 [r349450-349482] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c: Adds Subscription-State header to notify
+ with call completion. per RFC3265 (Closes issue ASTERISK-17953)
+ Reported by: George Konopacki Patches: 19400.patch uploaded by
+ mmichelson (license 5049)
+
+ * main/pbx.c: Fix documentation for SayNumber to reflect the fact
+ that language is changed in CHANNEL() (closes issue
+ ASTERISK-18962) reported by: Nir Simionovich
+
+2012-01-27 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.9.0 Released.
+
+ * Test results:
+ http://bamboo.asterisk.org/browse/TESTING-ASTERISK1890RCS-6
+
+2012-01-24 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.9.0-rc3 Released.
+
+ * Test results:
+ http://bamboo.asterisk.org/browse/TESTING-ASTERISK1890RCS-4
+
+ * main/file.c: Allow playback of formats that don't support
+ seeking. ast_streamfile previously did unconditional seeking
+ on files that broke playback of formats that don't support that
+ functionality. This patch avoids the seek that was causing the
+ problem. (closes issue ASTERISK-18994) Patch-by: Timo Teras
+
+ * channels/chan_sip.c: AST-2012-001: prevent crash when an SDP offer
+ is received with an encrypted video stream when support for video
+ is disabled and res_srtp is loaded. (closes issue ASTERISK-19202)
+ Reported by: Catalin Sanda
+
+ * channels/chan_sip.c: Fix RTP reference leak. If a blind transfer
+ were initiated using a REFER without a prior reINVITE to place the
+ call on hold, AND if Asterisk were sending RTCP reports, then there
+ was a reference leak for the RTP instance of the transferer.
+ (closes issue ASERISK-19192) Reported by: Tyuta Vitali
+
+ * main/features.c: Fix blind transfers from failing if an 'h' extension
+ is present. This prevents the 'h' extension from being run on the
+ transferee channel when it is transferred via a native transfer
+ mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported
+ by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by
+ Mark Michelson (license 5049)
+
+2012-01-13 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.9.0-rc2 Released.
+
+ * Test results:
+ http://bamboo.asterisk.org/browse/TESTING-ASTERISK1890RCS-3
+
+ * apps/app_queue.c: Realtime queues failed to load queue
+ information without queue member table. Revision 342223
+ broke this when it changed the return value for
+ realtime_multientry to return NULL when no results are
+ returned. (closes issue ASTERISK-19170) Reported by: Rene
+ Mendoza Tested by: Rene Mendoza
+
+
+2011-12-30 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.9.0-rc1 Released.
+
+ * Test results:
+ http://bamboo.asterisk.org/browse/TESTING-ASTERISK1890RCS-2
+
+2011-12-29 15:13 +0000 [r349339] Matthew Jordan <mjordan@digium.com>
+
+ * main/rtp_engine.c: Handle AST_CONTROL_UPDATE_RTP_PEER frames in
+ local bridge loop Failing to handle AST_CONTROL_UPDATE_RTP_PEER
+ frames in the local bridge loop causes the loop to exit
+ prematurely. This causes a variety of negative side effects,
+ depending on when the loop exits. This patch handles the frame by
+ essentially swallowing the frame in the local loop, as the
+ current channel drivers expect the RTP bridge to handle the
+ frame, and, in the case of the local bridge loop, no additional
+ action is necessary. (issue ASTERISK-19040) (issue
+ ASTERISK-19128) (issue ASTERISK-17725) (issue ASTERISK-18340)
+ (closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested
+ by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1640/
+
+2011-12-28 21:30 +0000 [r349289] Sean Bright <sean@malleable.com>
+
+ * main/audiohook.c: Use ast_audiohook_write_list_empty to determine
+ if our lists are empty instead of duplicating that logic.
+
+2011-12-27 20:48 +0000 [r349194] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_musiconhold.c, res/res_timing_pthread.c,
+ include/asterisk/module.h, res/res_timing_dahdi.c,
+ res/res_timing_timerfd.c: Fix timing source dependency issues
+ with MOH Prior to this patch, res_musiconhold existed at the same
+ module priority level as the timing sources that it depends on.
+ This would cause a problem when music on hold was reloaded, as
+ the timing source could be changed after res_musiconhold was
+ processed. This patch adds a new module priority level,
+ AST_MODPRI_TIMING, that the various timing modules are now loaded
+ at. This now occurs before loading other resource modules, such
+ that the timing source is guaranteed to be set prior to resolving
+ the timing source dependencies. (closes issue ASTERISK-17474)
+ Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf,
+ Wes Van Tlghem, elguero, Thomas Arimont Patches:
+ asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff
+ uploaded by elguero (License #5026)
+ asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff
+ uploaded by elguero (License #5026)
+ asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by
+ elguero (License #5026) Review:
+ https://reviewboard.asterisk.org/r/1578/
+
+2011-12-27 17:09 +0000 [r349144] Sean Bright <sean@malleable.com>
+
+ * main/audiohook.c: Once an audiohook is attached to a channel, we
+ continue to transcode all of the frames, even after all of the
+ hooks are detached. This patch short-cicuits us out before we
+ transcode unnecessarily.
+
+2011-12-23 17:25 +0000 [r349044] Sean Bright <sean@malleable.com>
+
+ * apps/app_chanspy.c: In ChanSpy, don't create audiohooks that will
+ never be used. When ChanSpy is initialized it creates and
+ attaches 3 audiohooks: 1) Read audio off of the channel that we
+ are spying on 2) Write audio to the channel that we are spying on
+ 3) Write audio to the channel that is bridged to the channel that
+ we are spying on. The first is always necessary, but the others
+ are used only when specific options are passed to the ChanSpy
+ application (B, d, w, and W to be specific). When those flags are
+ not passed, neither of those audiohooks are ever sent frames, but
+ we still try to process the hooks for each voice frame that we
+ recieve on the channel. So in short - only create and attach
+ audiohooks that we actually need.
+
+2011-12-23 15:24 +0000 [r348992] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_dial.c: Fix missing doc tags found while fixing
+ ASTERISK-18689 Add missing <variable></variable> tags in app_dial
+ documentation.
+
+2011-12-23 02:09 +0000 [r348940] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/pbx.h, main/pbx.c, channels/chan_sip.c: Fix
+ extension state callback references in chan_sip. Chan_sip gives a
+ dialog reference to the extension state callback and assumes that
+ when ast_extension_state_del() returns, the callback cannot
+ happen anymore. Chan_sip then reduces the dialog reference count
+ associated with the callback. Recent changes (ASTERISK-17760)
+ have resulted in the potential for the callback to happen after
+ ast_extension_state_del() has returned. For chan_sip, this could
+ be very bad because the dialog pointer could have already been
+ destroyed. * Added ast_extension_state_add_destroy() so chan_sip
+ can account for the sip_pvt reference given to the extension
+ state callback when the extension state callback is deleted. *
+ Fix pbx.c awkward statecbs handling in
+ ast_extension_state_add_destroy() and handle_statechange() now
+ that the struct ast_state_cb has a destructor to call. * Ensure
+ that ast_extension_state_add_destroy() will never return -1 or 0
+ for a successful registration. * Fixed pbx.c statecbs_cmp() to
+ compare the correct information. The passed in value to compare
+ is a change_cb function pointer not an object pointer. * Make
+ pbx.c ast_merge_contexts_and_delete() not perform callbacks with
+ AST_EXTENSION_REMOVED with locks held. Chan_sip is notorious for
+ deadlocking when those locks are held during the callback. *
+ Removed unused lock declaration for the pbx.c store_hints list.
+ (closes issue ASTERISK-18844) Reported by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1635/
+
+2011-12-22 22:31 +0000 [r348888] Matthew Jordan <mjordan@digium.com>
+
+ * cel/cel_pgsql.c: Fix for memory leaks / cleanup in cel_pgsql
+ There were a number of issues in cel_pgsql's pgsql_log method: *
+ If either sql or sql2 could not be allocated, the method would
+ return while the pgsql_lock was still locked * If the execution
+ of the log statement succeeded, the sql and sql2 structs were
+ never free'd * Reconnection successes were logged as ERRORs. In
+ general, the severity of several logging statements was reduced
+ (closes issue ASTERISK-18879) Reported by: Niolas Bouliane Tested
+ by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1624/
+
+2011-12-22 18:38 +0000 [r348833] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/frame.h: Allow packetization vaules > 127
+ According to the RTP packetization documentation, and the maximum
+ values listed in AST_FORMAT_LIST, we should support values > that
+ the signed char array that ast_codec_pref makes available to
+ store the value. All places in the code treat the framing field
+ as though it were an int array instaead of a char array anyway,
+ so this just fixes the type of the array. (closes issue
+ ASTERISK-18876) Review: https://reviewboard.asterisk.org/r/1639/
+
+2011-12-20 23:08 +0000 [r348735] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_iax2.c: Fix chan_iax2 to not report an RDNIS number
+ if it is blank. Some ISDN switches complain or block the call if
+ the RDNIS number is empty. * Made chan_iax2 not save a RDNIS
+ number into the ast_channel if the string is blank. This is what
+ other channel drivers do. (closes issue ASTERISK-17152) Reported
+ by: rmudgett
+
+2011-12-19 21:31 +0000 [r348647] Richard Mudgett <rmudgett@digium.com>
+
+ * configure, configure.ac: Fix crashes on other platforms caused by
+ interference from Darwin weak symbol support. Support weak
+ symbols on a platform specific basis. The Mac OS X (Darwin)
+ support must be isolated from the other platforms because it has
+ caused other platforms to crash. Several other platforms
+ including Linux have GCC versions that define the weak attribute.
+ However, this attribute is only setup for use in the code by
+ Darwin. (closes issue ASTERISK-18728) Reported by: Ben Klang
+ Review: https://reviewboard.asterisk.org/r/1617/
+
+2011-12-18 18:27 +0000 [r348516] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configs/sip.conf.sample, /: Correct two flaws in sip.conf.sample
+ related to AST-2011-013. * The sample file listed *two* values
+ for the 'nat' option as being the default. Only 'force_rport' is
+ the default. * The warning about having differing 'nat' settings
+ confusingly referred to both peers and users. ........ Merged
+ revisions 348515 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2
+
+2011-12-16 23:51 +0000 [r348310-348464] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, main/features.c: Clean-up on isle five for
+ __ast_request_and_dial() and ast_call_forward(). * Add locking
+ when a channel inherits variables and datastores in
+ __ast_request_and_dial() and ast_call_forward(). Note: The
+ involved channels are not active so there was minimal potential
+ for problems. * Remove calls to ast_set_callerid() in
+ __ast_request_and_dial() and ast_call_forward() because the set
+ information is for the wrong direction. * Don't use C++ keywords
+ for variable names in ast_call_forward(). * Run the redirecting
+ interception macro if defined when forwarding a call in
+ ast_call_forward(). Note: Currently will never execute because
+ the only callers that supply a calling channel supply a hungup or
+ zombie channel. * Make feature_request_and_dial() put the
+ transferee into autoservice when it calls ast_call_forward() in
+ case a redirection interception macro is run. Note: Currently
+ will never happen because the caller channel (Party B) is always
+ hungup at this time. * Make feature_request_and_dial() ignore the
+ AST_CONTROL_PROCEEDING frame to silence a log message.
+
+ * main/channel.c: Fix cut and past error in ast_call_forward().
+ (issue ASTERISK-18836)
+
+ * include/asterisk/cdr.h, apps/app_followme.c, apps/app_queue.c,
+ res/res_monitor.c, main/channel.c, main/pbx.c,
+ apps/app_authenticate.c, funcs/func_cdr.c, main/features.c: Fix
+ crash during CDR update. The ast_cdr_setcid() and
+ ast_cdr_update() were shown in ASTERISK-18836 to be called by
+ different threads for the same channel. The channel driver thread
+ and the PBX thread running dialplan. * Add lock protection around
+ CDR API calls that access an ast_channel pointer. (closes issue
+ ASTERISK-18836) Reported by: gpluser Review:
+ https://reviewboard.asterisk.org/r/1628/
+
+ * apps/app_parkandannounce.c: Fix ParkAndAnnounce to pass the
+ CallerID to the announcing channel. ParkAndAnnounce tried to pass
+ the CallerID to the announcing channel but the ID was wiped out
+ by the channel masquerade done when parking the call. * Save the
+ CallerID before parking the channel to pass it to the announcing
+ channel. * Fixed a minor memory leak in ParkAndAnnounce. *
+ Updated some ParkAndAnnounce log messages.
+
+2011-12-14 22:01 +0000 [r348212] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: Don't clear LOCALSTATIONID before sending or
+ receiving. The user may set that variable. ASTERISK-18921
+
+2011-12-14 20:34 +0000 [r348154-348157] Jonathan Rose <jrose@digium.com>
+
+ * configs/features.conf.sample: Fix accidental use of tabs instead
+ of spaces from previous features.conf.sample change
+
+ * configs/features.conf.sample: Document PARKINGSLOT variable in
+ features.conf.sample (issue ASTERISK-16239)
+
+2011-12-13 23:00 +0000 [r348101] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_followme.c, bridges/bridge_builtin_features.c: Fix
+ FollowMe CallerID on outgoing calls. The addition of the
+ Connected Line support changed how CallerID is passed to outgoing
+ calls. The FollowMe application was not updated to pass CallerID
+ to the outgoing calls. * Fix FollowMe CallerID on outgoing calls.
+ * Restructured findmeexec() to fix several memory leaks and
+ eliminate some duplicated code. * Made check the return value of
+ create_followme_number(). Putting a NULL into the numbers list is
+ bad if create_followme_number() fails. * Fixed a couple uses of
+ ast_strdupa() inside loops. * The changes to
+ bridge_builtin_features.c fix a similar CallerID issue with the
+ bridging API attended and blind transfers. (Not used at this
+ time.) (closes issue ASTERISK-17557) Reported by: hamlet505a
+ Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1612/
+
+2011-12-13 15:16 +0000 [r348048] Stefan Schmidt <sst@sil.at>
+
+ * channels/chan_sip.c: Fix possible misshandling of an incoming SIP
+ response as a peer poke response. Also make sure peer has even
+ qualify enabled when handle a peer poke response. (closes issue
+ ASTERISK-18940) Reported by: Vitaliy Tested by: Vitaliy and
+ UnixDev Review: https://reviewboard.asterisk.org/r/1620 Reviewed
+ by: David Vossel
+
+2011-12-12 19:22 +0000 [r347995] Terry Wilson <twilson@digium.com>
+
+ * res/res_srtp.c: Add a separate buffer for SRTCP packets The
+ function ast_srtp_protect used a common buffer for both SRTP and
+ SRTCP packets. Since this function can be called from multiple
+ threads for the same SRTP session (scheduler for SRTCP and
+ channel for SRTP) it was possible for the packets to become
+ corrupted as the buffer was used by both threads simultaneously.
+ This patch adds a separate buffer for SRTCP packets to avoid the
+ problem. (closes issue ASTERISK-18889, Reported/patch by Daniel
+ Collins)
+
+2011-12-09 01:19 +0000 [r347811] Richard Mudgett <rmudgett@digium.com>
+
+ * main/pbx.c: Fix some parsing issues in
+ add_exten_to_pattern_tree(). * Simplify compare_char() and avoid
+ potential sign extension issue. * Fix infinite loop in
+ add_exten_to_pattern_tree() handling of character set escape
+ handling. * Added buffer overflow checks in
+ add_exten_to_pattern_tree() character set collection. * Made
+ ignore empty character sets. * Added escape character handling to
+ end-of-range character in character sets. This has a slight
+ change in behavior if the end-of-range character is an escape
+ character. You must now escape it. * Fix potential sign extension
+ issue when expanding character set ranges. * Made remove
+ duplicated characters from character sets. The duplicate
+ characters lower extension matching priority and prevent
+ duplicate extension detection. * Fix escape character handling
+ when the escape character is trying to escape the end-of-string.
+ We could have continued processing characters after the end of
+ the exten string. We could have added the previous character to
+ the pattern matching tree incorrectly. (closes issue
+ ASTERISK-18909) Reported by: Luke-Jr
+
+2011-12-08 21:28 +0000 [r347718] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * channels/chan_sip.c: Fix regression when using tcpenable=no and
+ tlsenable=yes. The tlsenable settings are tucked away in
+ main/tcptls.c, so I missed them when resolving ASTERISK-18837.
+ This should resolve the test suite breakage of the sip tls tests.
+ Review: https://reviewboard.asterisk.org/r/1615 Reviewed by: Matt
+ Jordan
+
+2011-12-08 17:50 +0000 [r347595] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: Mark channel running the h exten with the
+ soft-hangup flag. When a bridge is broken, ast_bridge_call()
+ might execute the h exten on the calling channel. However, that
+ channel may not have been the channel that broke the bridge by
+ hanging up. The channel executing the h exten must be in a hung
+ up state so things like AGI run in the correct mode. * Make sure
+ ast_bridge_call() marks the channel it is executing the h exten
+ on as hung up. (The AST_SOFTHANGUP_APPUNLOAD flag is used so as
+ to match the pbx.c main dialplan execution loop when it executes
+ the h exten.) (closes issue ASTERISK-18811) Reported by: David
+ Hajek Patches: jira_asterisk_18811_v1.8.patch (license #5621)
+ patch uploaded by rmudgett Tested by: David Hajek, rmudgett
+
+2011-12-08 16:19 +0000 [r347531] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Don't crash on INFO automon request with
+ no channel AST-2011-014. When automon was enabled in
+ features.conf, it was possible to crash Asterisk by sending an
+ INFO request if no channel had been created yet. (closes issue
+ ASTERISK-18805) ........ Merged revisions 347530 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2
+
+2011-12-07 21:36 +0000 [r347438] Richard Mudgett <rmudgett@digium.com>
+
+ * main/manager.c: Update AMI Getvar and Setvar documentation about
+ supplying a channel name. (closes issue ASTERISK-18958) Reported
+ by: Red Patches: jira_asterisk_18958_v1.8.patch (license #5621)
+ patch uploaded by rmudgett
+
+2011-12-07 20:23 +0000 [r347369] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_meetme.c: Fix: Meetme recording variables from realtime
+ DB use null entries over channel variables Meetme would attempt
+ to substitute the realtime values of RECORDING_FILE and
+ RECORDING_FORMAT from the meetme db entry instead of using the
+ channel variable set for those variables in spite of those
+ database entries being NULL or even lacking a column to represent
+ them. (closes issue ASTERISK-18873) Reported by: Byron Clark
+ Patches: ASTERISK-18873-1.patch uploaded by Byron Clark (license
+ 6157)
+
+2011-12-06 23:47 +0000 [r347292] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Make SIP INFO messages for dtmf-relay
+ signals case insensitive. (closes issue ASTERISK-18924) Reported
+ by: Kevin Taylor
+
+2011-12-06 21:44 +0000 [r347239] Jonathan Rose <jrose@digium.com>
+
+ * main/pbx.c: Documents CHANNEL(musicclass) taking priority over
+ m([x]) in waitExten If waitExten specifies a music class to use
+ with its music on hold option, it will use CHANNEL(musicclass)
+ instead if that channel variable has been set on the initiating
+ channel. This documents that behavior in the waitExten app so
+ that this can be known without checking the documentation of the
+ code in function local_ast_moh_start. (closes issue
+ ASTERISK-18804)
+
+2011-12-06 19:39 +0000 [r347111-347166] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * channels/chan_sip.c: Don't allow transport=tcp when tcpenable=no.
+ When tcpenable=no, sending to transport=tcp hosts was still
+ allowed. Resolving the source address wasn't possible and yielded
+ the string "(null)" in SIP messages. Fixed that and a couple of
+ not-so-correct log messages. (closes issue ASTERISK-18837)
+ Reported by: Andreas Topp Review:
+ https://reviewboard.asterisk.org/r/1585 Reviewed by: Matt Jordan
+
+ * apps/app_voicemail.c: Add regression tests for issue
+ ASTERISK-18838. Review: https://reviewboard.asterisk.org/r/1572
+ Reviewed by: Matt Jordan
+
+ * apps/app_voicemail.c: Move setting of voicemail zonetag and
+ locale up a bit. The voicemail [general] zonetag and locale
+ variables weren't loaded until after the mailboxes were
+ initialized. This caused the settings to be unset for those
+ mailboxes until a reload was performed. (closes issue
+ ASTERISK-18838) Review: https://reviewboard.asterisk.org/r/1570
+ Reviewed by: Matt Jordan
+
+2011-12-06 17:05 +0000 [r347058] Matthew Jordan <mjordan@digium.com>
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: Fixed crash from
+ orphaned MWI subscriptions in chan_sip This patch resolves the
+ issue where MWI subscriptions are orphaned by subsequent SIP
+ SUBSCRIBE messages. When a peer is removed, either by pruning
+ realtime SIP peers or by unloading / loading chan_sip, the MWI
+ subscriptions that were orphaned would still be on the event
+ engine list of valid subscriptions but have a pointer to a peer
+ that no longer was valid. When an MWI event would occur, this
+ would cause a seg fault. (closes issue ASTERISK-18663) Reported
+ by: Ross Beer Tested by: Ross Beer, Matt Jordan Patches:
+ blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283)
+ Review: https://reviewboard.asterisk.org/r/1610/
+
+2011-12-05 17:39 +0000 [r347006] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h: Restore call progress code for analog
+ ports. Extracting sig_analog from chan_dahdi lost call progress
+ detection functionality. * Fix analog ports from considering a
+ call answered immediately after dialing has completed if the
+ callprogress option is enabled. (closes issue ASTERISK-18841)
+ Reported by: Richard Miller Patches: chan_dahdi.diff (license
+ #5685) patch uploaded by Richard Miller (Modified by me)
+ sig_analog.c.diff (license #5685) patch uploaded by Richard
+ Miller (Modified by me) sig_analog.h.diff (license #5685) patch
+ uploaded by Richard Miller
+
+2011-12-05 14:56 +0000 [r346954] Jonathan Rose <jrose@digium.com>
+
+ * main/pbx.c: Resolve duplicate label used in multiple priorities
+ for the same extension. Prior to this patch, if labels with the
+ same name were used for different priorities in the same
+ extension, the new label would be accepted, but it would be
+ unusable since attempts to reach that label would just go to the
+ first one. Now pbx.c detects this, generates a warning in logs,
+ and culls the label before adding it to the dialplan. (closes
+ issue ASTERISK-18807) Reported by: Kenneth Shumard Patches:
+ pbx.c.patch uploaded by Kenneth Shumard (License 5077)
+
+2011-12-05 14:45 +0000 [r346951] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_jabber.exports.in: Fix chan_jingle/gtalk load regression
+ introduced in r346087 Add missing symbol exports for
+ ast_aji_client_destroy and ast_aji_buddy_destroy for usage
+ outside res_jabber. Testing of these changes focused on
+ res_jabber itself, so this problem was missed. Reported-by:
+ Michael Spiceland
+
+2011-12-04 09:57 +0000 [r346899] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * channels/chan_sip.c: For SIP REGISTER fix domain-only URIs and
+ domain ACL bypass. The code that allowed admins to create users
+ with domain-only uri's had stopped to work in 1.8 because of the
+ reqresp parser rewrites. This is fixed now: if you have a
+ [mydomain.com] sip user, you can register with useraddr
+ sip:mydomain.com. Note that in that case -- if you're using
+ domain ACLs (a configured domain list) -- mydomain.com must be in
+ the allow list as well. Reviewboard r1606 shows a list of
+ registration combinations and which SIP response codes are
+ returned. Review: https://reviewboard.asterisk.org/r/1533/
+ Reviewed by: Terry Wilson (closes issue ASTERISK-18389) (closes
+ issue ASTERISK-18741)
+
+2011-12-02 16:19 +0000 [r346762] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c, channels/chan_h323.c: process null frame
+ pointer returned by ast_rtp_instance_read correctly (closes issue
+ ASTERISK-16697) Reported by: under Patches: segfault.diff
+ (License #5871) patch uploaded by under
+
+2011-12-01 21:11 +0000 [r346700] Richard Mudgett <rmudgett@digium.com>
+
+ * configs/res_stun_monitor.conf.sample, include/asterisk/stun.h,
+ main/stun.c, res/res_stun_monitor.c: Re-resolve the STUN address
+ if a STUN poll fails for res_stun_monitor. The STUN socket must
+ remain open between polls or the external address seen by the
+ STUN server is likely to change. However, if the STUN request
+ poll fails then the STUN server address needs to be re-resolved
+ and the STUN socket needs to be closed and reopened. * Re-resolve
+ the STUN server address and create a new socket if the STUN
+ request poll fails. * Fix ast_stun_request() return value
+ consistency. * Fix ast_stun_request() to check the received
+ packet for expected message type and transaction ID. * Fix
+ ast_stun_request() to read packets until timeout or an associated
+ response packet is found. The stun_purge_socket() hack is no
+ longer required. * Reduce ast_stun_request() error messages to
+ debug output. * No longer pass in the destination address to
+ ast_stun_request() if the socket is already bound or connected to
+ the destination. (closes issue ASTERISK-18327) Reported by:
+ Wolfram Joost Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1595/
+
+2011-12-01 20:36 +0000 [r346564-346697] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c: Change 183 Ringing in sipfrag body to 180
+ ringing. 183 Ringing isn't even a thing. 183 is actually a
+ session progress message. (closes issue ASTERISK-18925) Reported
+ by: Sebastian Denz Tested by: jrose Patches:
+ asterisk18-use_180_instead_of_183_in_sipfrag.diff by Sebastian
+ Denz (License #6139)
+
+ * include/asterisk/tcptls.h, main/tcptls.c, channels/chan_sip.c:
+ r346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) |
+ 18 lines Cleaning up chan_sip/tcptls file descriptor closing.
+ This patch attempts to eliminate various possible instances of
+ undefined behavior caused by invoking close/fclose in situations
+ where fclose may have already been issued on a
+ tcptls_session_instance and/or closing file descriptors that
+ don't have a valid index for fd (-1). Thanks for more than a
+ little help from wdoekes. (closes issue ASTERISK-18700) Reported
+ by: Erik Wallin (issue ASTERISK-18345) Reported by: Stephane
+ Cazelas (issue ASTERISK-18342) Reported by: Stephane Chazelas
+ Review: https://reviewboard.asterisk.org/r/1576/
+
+2011-11-30 19:36 +0000 [r346472] Leif Madsen <lmadsen@digium.com>
+
+ * configs/queues.conf.sample: Update queues.conf.sample
+ documentation. Update the documentation surrounding the use of
+ MONITOR_EXEC to make it more clear that it can be used for both
+ Monitor() and MixMonitor() usage. (closes issue ASTERISK-17413)
+ Reported by: David Woolley Patches:
+ issue18817_mixmonitor_queues_doc.diff by Michael L. Young
+ (License #5026)
+
+2011-11-28 14:30 +0000 [r346292] Stefan Schmidt <sst@sil.at>
+
+ * res/res_rtp_asterisk.c: Fix regression that 'rtp/rtcp set debup
+ ip' only works when also a port was specified. (closes issue
+ ASTERISK-18693) Reported by: Davide Dal Fra Review:
+ https://reviewboard.asterisk.org/r/1600/ Reviewed by: Walter
+ Doekes
+
+2011-11-23 22:52 +0000 [r346239] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_iax2.c, include/asterisk/acl.h,
+ channels/chan_skinny.c, channels/chan_h323.c, main/acl.c: Fix
+ calls to ast_get_ip() not initializing the address family.
+
+2011-11-23 20:15 +0000 [r346144-346147] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * channels/chan_sip.c: Minor cleanup in chan_sip get_msg_text()
+ function. In r116240, get_msg_text() got an extra parameter to
+ fix the unwanted addition of trailing newlines to SIP MESSAGE
+ bodies. This caused all linefeeds to be trimmed, which isn't
+ right either. This is a stop-gap; the right fix is to return the
+ original SIP request body. Review:
+ https://reviewboard.asterisk.org/r/1586 Reviewed by: Matt Jordan
+
+ * include/asterisk/strings.h: Fix ast_str_truncate signedness
+ warning and documentation. Review:
+ https://reviewboard.asterisk.org/r/1594
+
+2011-11-23 17:12 +0000 [r346086] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_gtalk.c, res/res_jabber.c, channels/chan_jingle.c,
+ include/asterisk/jabber.h: Fix res_jabber resource leaks This
+ should fix almost all resource leaks in res_jabber that involve
+ ASTOBJ_CONTAINER_FIND and resolves an ambiguous situation where
+ ast_aji_get_client would sometimes bump an object's refcount and
+ sometimes not. Review: https://reviewboard.asterisk.org/r/1553
+
+2011-11-23 16:09 +0000 [r346030] Terry Wilson <twilson@digium.com>
+
+ * res/res_musiconhold.c: Resume playing existing hold music for
+ cached realtime MOH As a result of the fix for ASTERISK-18039,
+ realtime caching MOH no longer properly resumes playing back a
+ file between different holds in the same call. This is because
+ scanning for new files causes the existing file array to be
+ emptied and we were just comparing that the saved pointer to the
+ filename matched the pointer to the filename in a particular
+ position in the array. An easy fix is to save the filename
+ instead of a pointer to it and then do a strcmp instead of
+ comparing the addresses. (closes issue ASTERISK-18912) Review:
+ https://reviewboard.asterisk.org/r/1596/
+
+2011-11-22 22:55 +0000 [r345976] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/dnsmgr.h, main/dnsmgr.c: Fix dnsmgr entries to
+ ask for the same address family each time. The dnsmgr refresh
+ would always get the first address found regardless of the
+ original address family requested. So if you asked for only IPv4
+ addresses originally, you might get an IPv6 address on refresh. *
+ Saved the original address family requested by
+ ast_dnsmgr_lookup() to be used when the address is refreshed.
+
+2011-11-22 20:29 +0000 [r345923] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * include/asterisk/logger.h: Clarify why the AST_LOG_* macros exist
+ next to the LOG_* macros. (issue ASTERISK-17973)
+
+2011-11-21 21:03 +0000 [r345828-345829] Terry Wilson <twilson@digium.com>
+
+ * CHANGES: Change nat=yes to nat=force_rport in CHANGES Fix a small
+ documentation merge issue ASTERISK-18862
+
+ * configs/sip.conf.sample, CHANGES, /, channels/chan_sip.c: Default
+ to nat=yes; warn when nat in general and peer differ It is
+ possible to enumerate SIP usernames when the general and
+ user/peer nat settings differ in whether to respond to the port a
+ request is sent from or the port listed for responses in the Via
+ header. In 1.4 and 1.6.2, this would mean if one setting was
+ nat=yes or nat=route and the other was either nat=no or
+ nat=never. In 1.8 and 10, this would mean when one was
+ nat=force_rport and the other was nat=no. In order to address
+ this problem, it was decided to switch the default behavior to
+ nat=yes/force_rport as it is the most commonly used option and to
+ strongly discourage setting nat per-peer/user when at all
+ possible. For more discussion of the issue, please see:
+ http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html
+ (closes issue ASTERISK-18862) Review:
+ https://reviewboard.asterisk.org/r/1591/ ........ Merged
+ revisions 345776 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.4 ........ Merged
+ revisions 345800 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2
+
+2011-11-19 15:08 +0000 [r345682] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/db.c: Update the documentation to better clarify how the
+ existing commands work. Review:
+ https://reviewboard.asterisk.org/r/1593/
+
+2011-11-17 17:06 +0000 [r345546] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Remove dead code since pri_grab() can never
+ fail. Dead code makes programmers sick. I am sick of looking at
+ it.
+
+2011-11-17 17:04 +0000 [r345545] Jason Parker <jparker@digium.com>
+
+ * apps/app_confbridge.c: Fix documentation of 's' option. The menu
+ key is #, not *. Reported by p3nguin on #asterisk.
+
+2011-11-16 14:42 +0000 [r345487] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_voicemail.c: Guarantee messages go into the right
+ folders with multiple recipients Before, using the U flag in
+ Voicemail with multiple recipients would put urgent messages in
+ the INBOX folder for all users past the first thanks to a bug
+ with the message copying function. This would also cause messages
+ to fail to be sent if the INBOX directory hadn't been created for
+ that mailbox yet. (closes issue ASTERISK-18245) Reported by: Matt
+ Jordan (closes issue ASTERISK-18246) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/1589/
+
+2011-11-15 20:09 +0000 [r345219-345431] Richard Mudgett <rmudgett@digium.com>
+
+ * res/res_agi.c: Make FastAGI HANGUP show up in AGI debug output. *
+ Change from using send() to ast_agi_send() so the HANGUP shows up
+ in the AGI debug output. (closes issue ASTERISK-18723) Reported
+ by: James Van Vleet Patches: jira_asterisk_18723_v1.8.patch
+ (license #5621) patch uploaded by rmudgett
+
+ * channels/sig_pri.c: Fix typo in sig_pri using wrong structure
+ name. It is fortunate that the typo does not alter generated code
+ since the e->restart.channel and e->ring.channel members are in
+ the same position. (closes issue ASTERISK-18868) Reported by:
+ zvision Patches: sig_pri.c.diff (License #5755) patch uploaded by
+ zvision
+
+ * apps/app_queue.c: Make queue log indicate if ADDMEMBER is paused
+ for AMI and realtime. * Add parameter to queue log ADDMEMBER to
+ indicate if the member is paused. (closes issue ASTERISK-18645)
+ Reported by: garlew Patches: paused.diff (License #5337) patch
+ uploaded by garlew Tested by: rmudgett, garlew Review:
+ https://reviewboard.asterisk.org/r/1469/
+
+ * UPGRADE.txt, configs/sip.conf.sample, channels/sip/include/sip.h,
+ channels/chan_sip.c: Restore SIP DTMF overlap dialing method. The
+ recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support
+ working correctly removed a long standing ability to do overlap
+ dialing using DTMF in the early media phase of a call. See
+ ASTERISK-18702 it has a very good description of the issue. I
+ started with Pavel Troller's chan_sip.diff patch on issue
+ ASTERISK-18702. * Added 'dtmf' enum value to sip.conf
+ allowoverlap config option. The new option value causes the
+ Incomplte application to not send anything with chan_sip so the
+ caller can supply more digits via DTMF. * Renames
+ SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE
+ since that is what it really means. * Fixed get_destination()
+ inconsistency with the pickup extension matching. * Fixed
+ initialization of PAGE3 of global_flags in reload_config().
+ (closes issue ASTERISK-18702) Reported by: Pavel Troller Review:
+ https://reviewboard.asterisk.org/r/1517/ Review:
+ https://reviewboard.asterisk.org/r/1582/
+
+ * main/pbx.c: Fix Progress spelling error in main/pbx.c. (closes
+ issue ASTERISK-18857) Reported by: David M Patches:
+ mainpbx-trivial.patch (License #6326) patch uploaded by David M
+
+2011-11-14 19:05 +0000 [r345163] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c: Don't read past end of input when calling write()
+ int blah = 1; ... write(chan->alertpipe[1], &blah, new_frames *
+ sizeof(blah)) != (new_frames * sizeof(blah))) is only valid when
+ new_frames == 1. Otherwise we start reading into adjacent
+ variables declared on the stack. The read end discards what is
+ read, so the values don't matter but it's not a good idea to read
+ past where we want even though new_frames is almost always 1 and
+ should never be large. This patch is basically taken out of
+ kpfleming's eventfd branch, as he mentioned that he remembered
+ fixing it there when I talked to him about this issue. Review:
+ https://reviewboard.asterisk.org/r/1583/
+
+2011-11-14 19:00 +0000 [r345160] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * channels/sip/include/reqresp_parser.h: Update reqresp_parser
+ parse_uri doxygen comments. The issue mentioned in the bug report
+ had been fixed recently by twilson. The reporter included this
+ documentation fix. (closes issue ASTERISK-18572) Reported by:
+ Richard Miller Patch by: Richard Miller (modified)
+
+2011-11-14 15:08 +0000 [r345063] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c: Ensure that a null vmexten does not cause a
+ segfault When sip_send_mwi_to_peer was modified recently to avoid
+ deadlocks, vmexten was not expected to be null. This change
+ handles that situation to avoid a segfault.
+
+2011-11-14 15:00 +0000 [r345062] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_voicemail.c: Moves voicemail setup password entry to the
+ end of the setup process. This change was made because
+ forcegreeting and forcename settings in voicemail could be
+ circumvented by hanging up after entering a password, because the
+ only way voicemail currently observes whether a mailbox is new or
+ not is by checking to see if the password is the same as the
+ mailbox number or not. (closes issue ASTERISK-18282) Reported by:
+ Matt Jordan Review: https://reviewboard.asterisk.org/r/1581/
+
+2011-11-12 16:05 +0000 [r344965] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * channels/chan_misdn.c: mISDN Round Robin break when no channel is
+ available Prevent channels been parsed repetitively.
+
+2011-11-12 00:24 +0000 [r344899] Terry Wilson <twilson@digium.com>
+
+ * res/res_musiconhold.c: Don't forget to rescan MOH files for
+ cached realtime classes Realtime MOH class caching was
+ implemented because without it, you would build a completely new
+ MOH class and would start the music over at the beginning each
+ time hold was pressed in a conversation. Unfortunately, this
+ broke re-scanning for file changes for realtime MOH classes. This
+ patch corrects that issue. (closes issue ASTERISK-18039) Review:
+ https://reviewboard.asterisk.org/r/1579/
+
+2011-11-11 21:54 +0000 [r344835-344843] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * main/utils.c, include/asterisk/stringfields.h,
+ include/asterisk/utils.h: Use __alignof__ instead of sizeof for
+ stringfield length storage. Kevin P Fleming suggested that
+ r343157 should use __alignof__ instead of sizeof. For most
+ systems this won't be an issue, but better fix it now while it's
+ still fresh. Review: https://reviewboard.asterisk.org/r/1573
+
+ * channels/sip/reqresp_parser.c: Remove unneeded if(params) checks
+ in reqresp_parser. Nick Lewis added them in
+ https://reviewboard.asterisk.org/r/549/diff/1-2/ for no apparent
+ reason. There is no way that params could become NULL in that
+ piece of code, so I removed these excess checks again.
+
+ * main/manager.c: Fix bad quoting of multiline mxml opaque_data
+ that caused invalid xml. The opaque_data was added and enclosed
+ in single quotes, assuming it would be only a single line. The
+ rest of the lines were appended after the closing quote. (closes
+ issue ASTERISK-18852) Reported by: peep_ on IRC Review:
+ https://reviewboard.asterisk.org/r/1577
+
+2011-11-11 20:42 +0000 [r344823] Matthew Jordan <mjordan@digium.com>
+
+ * main/file.c: Video format was treated as audio when removed from
+ the file playback scheduler This patch fixes the format type
+ check in ast_closestream and filestream_destructor. Previously a
+ comparison operator was used, but since audio formats are no
+ longer contiguous (and AST_FORMAT_AUDIO_MASK includes formats
+ that have a value greater than the video formats), a bitwise AND
+ operation is used instead. Duplicated code was also moved to
+ filestream_close. (closes issue ASTERISK-18682) Reported by: Aldo
+ Bedrij Tested by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1580/
+
+2011-11-11 20:10 +0000 [r344769] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c: Fix regression introduced by SDP fixups If
+ capability is adjusted when switching to UDPTL during fax
+ transmission, fax teardown fails. Make sure capability is only
+ touched if RTP is active. This regression was introduced in
+ R344385.
+
+2011-11-11 18:35 +0000 [r344661-344715] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Check sip.conf maxforwards parameter for
+ range 1 <= x <= 255. JIRA AST-710
+
+ * main/cli.c: Make CLI "core show channel" not hold the channel
+ lock during console output. Holding the channel lock while the
+ CLI "core show channel" command is executing can slow down the
+ system. It could block the system if the console output is halted
+ or paused. * Made capture the CLI "core show channel" output into
+ a buffer to be output after the channel is unlocked. * Removed
+ use of C++ keyword as a variable name. out renamed to obuf. *
+ Checked allocation of obuf for failure so will not crash. (closes
+ issue ASTERISK-18571) Reported by: Pavel Troller Tested by:
+ rmudgett
+
+2011-11-11 15:21 +0000 [r344608] Jonathan Rose <jrose@digium.com>
+
+ * main/pbx.c: Fix a segmentation fault when using an extension with
+ CID matching and no CID. Attempting to call an extension which
+ used Caller ID matching with a channel that has an empty caller
+ id string would result in a segmentation fault. (closes issue
+ ASTERISK-18392 Reported By: Ales Zelenik
+
+2011-11-10 22:59 +0000 [r344536-344539] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_queue.c: Fix potential deadlock calling ast_call() with
+ channel locks held. Fixed app_queue.c:ring_entry() calling
+ ast_call() with the channel locks held. Chan_local attempts to do
+ deadlock avoidance in its ast_call() callback and could deadlock
+ if a channel lock is already held.
+
+ * apps/app_queue.c: Make AMI event AgentCalled get
+ CallerID/ConnectedLine info from the incoming channel. It was
+ strange that the AgentCalled AMI event would get most of its
+ information from the incoming channel but then get the CallerID
+ information from the outgoing channel. Before connected line
+ support was added, this information was always the same at this
+ point. (closes issue ASTERISK-18152) Reported by: Thomas Farnham
+ Tested by: rmudgett
+
+2011-11-10 21:14 +0000 [r344385-344439] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_meetme.c: Fix another incorrect case with meetme's PIN
+ logic and add documentation This fixes an issue where a user of a
+ dynamic conference was asked for a PIN twice. This also adds
+ documentation to assist in future modifications to the piece of
+ code responsible for PIN checking. (closes issue AST-670)
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: Fix several bugs
+ with SDP parsing and well-formedness of responses Fix bug
+ ASTERISK-16558 which dealt with the order of responses to
+ incoming streams defined by SDP. Fix unreported bug where
+ offering multiple same-type streams would cause Asterisk to reply
+ with an incorrect SDP response missing one or more streams
+ without a proper declination. Fix bugs related to a single
+ non-audio stream being offered with responses requesting codecs
+ that were not offered in the initial invite along with an
+ additional audio stream that was not in the initial invite.
+ Review: https://reviewboard.asterisk.org/r/1516/
+
+2011-11-10 16:18 +0000 [r344330] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_rtp_asterisk.c: only attempt to do stun handling on ipv4
+ or ipv4 mapped to ipv6 addresses Patch by: jkonieczny (modified)
+ ASTERISK-18490
+
+2011-11-09 20:37 +0000 [r344268] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Fix deadlock during dialplan reload. Another
+ deadlock between the conlock/hints and channels/channel locking
+ orders. * Don't hold the channel and private lock in sip_new()
+ when calling ast_exists_extension(). (closes issue
+ ASTERISK-18740) Reported by: Byron Clark Patches:
+ sip_exists_exten_dlock_3.diff (license #5041) patch uploaded by
+ Gregory Hinton Nietsky ASTERISK-18740.patch (license #6157) patch
+ uploaded by Byron Clark Tested by: Byron Clark
+
+2011-11-09 19:57 +0000 [r344215] Terry Wilson <twilson@digium.com>
+
+ * channels/sip/include/sip.h,
+ channels/sip/include/reqresp_parser.h, channels/chan_sip.c,
+ channels/sip/reqresp_parser.c: Don't treat a host:port string as
+ a domain The domain matching code prior to 1.8 used to manually
+ remove the port from the host:port string when determining if an
+ incoming request matched the list of domains. When switching to
+ the new parsing functions, the documentation implied that the
+ "domain" was being returned by these functions, when instead it
+ was returning the "hostport" as defined by RFC 3261. This led to
+ confusion and resulted in 1.8+ rejecting an incoming request from
+ x.x.x.x:xxxxx when domain=x.x.x.x was set in sip.conf. This patch
+ renames the "domain" variables in the parsing functions to
+ "hostport" to more accurately describe what it is that they are
+ returning and also properly truncates the resulting hostport
+ strings when dealing with domain matching. Review:
+ https://reviewboard.asterisk.org/r/1574/
+
+2011-11-09 18:42 +0000 [r344158] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ootypes.h, addons/ooh323c/src/oochannels.c,
+ addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooh323.c,
+ addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooq931.h: (closes
+ issue ASTERISK-18748) Reported by: Fabrizio Lazzaretti Patches:
+ ASTERISK-18748-5.patch (License #5415) patch uploaded by may213
+ Tested by: Fabrizio Lazzaretti
+
+2011-11-09 18:38 +0000 [r344157] Terry Wilson <twilson@digium.com>
+
+ * tests/test_netsock2.c: Add a unit test for
+ ast_sockaddr_split_hostport Review:
+ https://reviewboard.asterisk.org/r/1575/
+
+2011-11-09 17:13 +0000 [r344102] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_meetme.c: Fix pin parameter behavior regression in
+ MeetMe The last time this code was touched (by me), a subtlety
+ was missed based on the difference between needing to check a
+ pin's validity and the need to prompt for a pin. (closes issue
+ ASTERISK-18488)
+
+2011-11-09 15:25 +0000 [r344048] Matthew Nicholson <mnicholson@digium.com>
+
+ * formats/format_wav.c: don't call ltohl() twice on the same value
+ ASTERISK-18739 Patch by: pawel (modified)
+
+2011-11-08 19:25 +0000 [r343936] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * pbx/pbx_config.c: Fix crash when dialplan remove include is
+ called with too few arguments. "dialplan remove include x from y"
+ crashed when the amount of arguments was less than 6. (closes
+ issue ASTERISK-18762) Reported by: Andrey Solovyev Tested by:
+ Andrey Solovyev
+
+2011-11-08 17:58 +0000 [r343851] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c, main/acl.c: Fixed reference to incorrect
+ variable if unknown host configured crash. * Fixed a LOG_ERROR
+ message referencing the config variable list v that had
+ previously been processed and became NULL. * Added error return
+ value set that was missing in an ast_append_ha() error return
+ path. (closes issue ASTERISK-18743) Reported by: Michele Patches:
+ issueA18743-fix_dynamic_exclude_static_bad_host_log.patch
+ (license #5674) patch uploaded by Walter Doekes Tested by:
+ Michele
+
+2011-11-08 13:26 +0000 [r343791] Leif Madsen <lmadsen@digium.com>
+
+ * build_tools/prep_tarball: Fix boo-boo in prep_tarball script. A
+ hardcoded a branch number was in the prep_tarball which could not
+ work. Changed it to the variable.
+
+2011-11-07 21:40 +0000 [r343690] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: respect case changes in peer names on sip
+ reload ASTERISK-18669
+
+2011-11-07 21:13 +0000 [r343637] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Fix __sip_subscribe_mwi_do() incorectly
+ changing dialogs hash key callid. Changing an object value used
+ as a container key requires removing the object from the
+ container and reinserting it. * Created change_callid_pvt() to
+ call instead of build_callid_pvt(). The change_callid_pvt() will
+ correctly change the dialog callid so the ao2 conainter can
+ explicitly unlink it.
+
+2011-11-07 20:27 +0000 [r343621] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c: Prevent BLF subscriptions from causing
+ deadlocks Fix a locking inversion in sip_send_mwi_to_peer that
+ was causing deadlocks. This function now requires that both the
+ peer and associated pvt be unlocked before it is called for cases
+ where peer and peer->mwipvt form a circular reference. (closes
+ issue ASTERISK-18663) Review:
+ https://reviewboard.asterisk.org/r/1563/
+
+2011-11-07 19:36 +0000 [r343577] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Fix deadlock if peer is destroyed while
+ sending MWI notice. A dialog cannot be destroyed by the
+ ao2_callback dialog_needdestroy because of a deadlock between the
+ dialogs container lock and the RWLOCK of the events subscription
+ list. * Create dialogs_to_destroy container to hold dialogs that
+ will be destroyed. * Ensure that the event subscription callback
+ will never happen with an invalid peer pointer by making the
+ event callback removal the first thing in the peer destructor
+ callback. (closes issue ASTERISK-18747) Reported by: Gregory
+ Hinton Nietsky Review: https://reviewboard.asterisk.org/r/1564/
+
+2011-11-03 20:26 +0000 [r343375] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * res/res_config_sqlite.c: Fix sqlite config driver segfault and
+ broken queries The sqlite realtime handler assumed you had a
+ static config configured as well. The realtime multientry handler
+ assumed that you weren't using dynamic realtime. (closes issue
+ ASTERISK-18354) (closes issue ASTERISK-18355) Review:
+ https://reviewboard.asterisk.org/r/1561
+
+2011-11-03 19:56 +0000 [r343336] Richard Mudgett <rmudgett@digium.com>
+
+ * funcs/func_dialgroup.c: Remove invalid flag given to iterator in
+ func_dialgroup.c
+
+2011-11-03 16:15 +0000 [r343281] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/printHandler.c, addons/ooh323c/src/ooq931.c,
+ addons/ooh323c/src/ooGkClient.c, addons/ooh323c/src/ooTimer.c,
+ addons/ooh323c/src/dlist.c, addons/ooh323c/src/dlist.h: Final fix
+ memleaks in GkClient codes, same for Timer codes. (these memleaks
+ stop development of gk codes, now i can continue) Fix
+ printHandler 'Unbalanced Structure' issues with locking
+ printHandler data for single thread.
+
+2011-11-03 15:33 +0000 [r343220-343276] Terry Wilson <twilson@digium.com>
+
+ * channels/sip/include/sip.h: Make room for the fax detect flags
+ The original REGISTERTRYING flag, in addition to being impossible
+ to check, also encroached on the space for the flag above it.
+ This patch moves the flags that were below REGISTERTRYING back to
+ where they were as though we had just removed the REGISTERTRYING
+ option.
+
+ * channels/sip/include/sip.h, contrib/realtime/mysql/sippeers.sql,
+ channels/chan_sip.c: Remove registertrying option in chan_sip
+ This option is not only useless, but has been broken since
+ inception since the flag was never copied from the peer where it
+ is set to the pvt where it was checked. RFC 3261 specificially
+ states that you should not send a provisional response to a
+ non-INVITE request, and if we did fix the code so that it worked,
+ it would cause the same kind of user enumeration vulnerability
+ that we've discussed with the nat= setting. This patch removes
+ registertrying option and any code that would have sent a 100
+ response to a register. Review:
+ https://reviewboard.asterisk.org/r/1562/
+
+2011-11-02 22:21 +0000 [r343157-343181] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * channels/chan_sip.c: Fix improper warning introduced by r342927
+ and more tweaks Changeset r342927 introduced a warning which was
+ only supposed to be emitted when a found realtime peer had an
+ empty (or no) name. It turned out that there were some
+ inconsistencies left. Now found peers with an empty name are
+ explicitly ignored like before r342927 but better. Reviewed by:
+ Stefan Schmidts, Terry Wilson Review:
+ https://reviewboard.asterisk.org/r/1560
+
+ * main/utils.c, include/asterisk/stringfields.h,
+ include/asterisk/utils.h: Ensure that string field lengths are
+ properly aligned Integers should always be aligned. For some
+ platforms (ARM, SPARC) this is more important than for others.
+ This changeset ensures that the string field string lengths are
+ aligned on *all* platforms, not just on the SPARC for which there
+ was a workaround. It also fixes that the length integer can be
+ resized to 32 bits without problems if needed. (closes issue
+ ASTERISK-17310) Reported by: radael, S Adrian Reviewed by:
+ Tzafrir Cohen, Terry Wilson Tested by: S Adrian Review:
+ https://reviewboard.asterisk.org/r/1549
+
+2011-11-02 19:32 +0000 [r343047-343102] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_authenticate.c: Add note about how Authenticate()
+ application with option 'd' works. (closes issue ASTERISK-17422)
+ Reported by: Leif Madsen
+
+ * configs/queues.conf.sample: Update documentation for leastrecent
+ strategy. In queues.conf.sample the leastrecent strategy was
+ incorrectly described. Now updated to reflect how the strategy
+ actually checks peers. (closes issue ASTERISK-17854) Reported by:
+ Sebastian Denz Patches: queues.conf-doc_issue.patch (License
+ #6139)
+
+2011-11-02 13:44 +0000 [r342990] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_meetme.c: Modify comments in MeetMe application
+ documentation about DAHDI. The MeetMe application documentation
+ has some comments about usage of DAHDI, and they were a bit
+ outdated relative to modern DAHDI releases. This patch changes
+ the comment to just tell the user that a functional DAHDI timing
+ source is required, and no longer mention 'dahdi_dummy', since
+ that module does not exist in current DAHDI releases.
+
+2011-11-01 20:53 +0000 [r342869-342927] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * main/config.c, channels/chan_sip.c,
+ configs/extconfig.conf.sample, include/asterisk/config.h: Several
+ fixes to the chan_sip dynamic realtime peer/user lookup There
+ were several problems with the dynamic realtime peer/user lookup
+ code. The lookup logic had become rather hard to read due to lots
+ of incremental changes to the realtime_peer function. And, during
+ the addition of the sipregs functionality, several possibilities
+ for memory leaks had been introduced. The insecure=port matching
+ has always been broken for anyone using the sipregs family. And,
+ related, the broken implementation forced those using sipregs to
+ *still* have an ipaddr column on their sippeers table. Thanks
+ Terry Wilson for comprehensive testing and finding and fixing
+ unexpected behaviour from the multientry realtime call which
+ caused the realtime_peer to have a completely unused code path.
+ This changeset fixes the leaks, the lookup inconsistenties and
+ that you won't need an ipaddr column on your sippeers table
+ anymore (when you're using sipregs). Beware that when you're
+ using sipregs, peers with insecure=port will now start matching!
+ (closes issue ASTERISK-17792) (closes issue ASTERISK-18356)
+ Reported by: marcelloceschia, Walter Doekes Reviewed by: Terry
+ Wilson Review: https://reviewboard.asterisk.org/r/1395
+
+ * UPGRADE.txt, configs/res_ldap.conf.sample, res/res_realtime.c,
+ configs/dbsep.conf.sample, main/config.c,
+ contrib/realtime/mysql/sipfriends.sql (removed),
+ contrib/realtime/mysql/sippeers.sql (added),
+ configs/res_config_mysql.conf.sample,
+ configs/extconfig.conf.sample: Cleanup references to sipusers and
+ sipfriends dynamic realtime families Somewhere between 1.4 and
+ 1.8 the sipusers family has become completely unused. Before
+ that, the sipfriends family had been obsoleted in favor of
+ separate sipusers and sippeers families. Apparently, they have
+ been merged back again into a single family which is now called
+ "sippeers". Reviewed by: irroot, oej, pabelanger Review:
+ https://reviewboard.asterisk.org/r/1523
+
+2011-10-31 15:58 +0000 [r342769] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_iax2.c, main/pbx.c: Fixed invalid memory access
+ when adding extension to pattern match tree When an extension is
+ removed from a context, its entry in the pattern match tree is
+ not deleted. Instead, the extension is marked as deleted. When an
+ extension is removed and re-added, if that extension is also a
+ prefix of another extension, several log messages would report an
+ error and did not check whether or not the extension was deleted
+ before accessing the memory. Additionally, if the extension was
+ already in the tree but previously deleted, and the pattern was
+ at the end of a match, the findonly flag was not honored and the
+ extension would be erroneously undeleted. Additionaly, it was
+ discovered that an IAX2 peer could be unregistered via the CLI,
+ while at the same time it could be scheduled for unregistration
+ by Asterisk. The unregistration method now checks to see if the
+ peer was already unregistered before continuing with an
+ unregistration. (closes issue ASTERISK-18135) Reported by: Jaco
+ Kroon, Henry Fernandes, Kristijan Vrban Tested by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/1526
+
+2011-10-29 04:19 +0000 [r342661] Richard Mudgett <rmudgett@digium.com>
+
+ * tests/test_linkedlists.c, include/asterisk/linkedlists.h: Fix
+ AST_LIST_INSERT_BEFORE_CURRENT() updating the wrong variable.
+ AST_LIST_INSERT_BEFORE_CURRENT() could not be used twice in an
+ iteration or before AST_LIST_REMOVE_CURRENT() without corrupting
+ the list. AST_LIST_INSERT_BEFORE_CURRENT() could also corrupt the
+ list if AST_LIST_INSERT_BEFORE_CURRENT() or
+ AST_LIST_REMOVE_CURRENT() is used on the next iteration. * Fixed
+ cut and paste error using the wrong variable in
+ AST_LIST_INSERT_BEFORE_CURRENT(). * Added linked list unit tests
+ for AST_LIST_INSERT_BEFORE_CURRENT(), AST_LIST_APPEND_LIST(), and
+ AST_LIST_INSERT_LIST_AFTER().
+
+2011-10-27 19:34 +0000 [r342545-342602] Jonathan Rose <jrose@digium.com>
+
+ * res/res_rtp_multicast.c: Fix sequence number overflow over 16
+ bits causing codec change in RTP packets. Sequence number was
+ handled as an unsigned integer (usually 32 bits I think, more
+ depending on the architecture) and was put into the rtp packet
+ which is basically just a bunch of bits using an or operation.
+ Sequence number only has 16 bits allocated to it in an RTP packet
+ anyway, so it would add to the next field which just happened to
+ be the codec. This makes sure the sequence number is set to be a
+ 16 bit integer regardless of architecture (hopefully) and also
+ makes it so the incrementing of the sequence number does bitwise
+ or at the peak of a 16 bit number so that the value will be set
+ back to 0 when going beyond 65535 anyway. (closes issue
+ ASTERISK-18291) Reported by: Will Schick Review:
+ https://reviewboard.asterisk.org/r/1542/
+
+ * res/res_jabber.c: Cleanup reference leaks in res_jabber
+ res_jabber.c had a number of places where astobjs would be
+ referenced and have their reference counts bumped without having
+ a dereference made before the object lost scope. This patch adds
+ a number of ASTOBJ_UNREFs to resolve that. Review:
+ https://reviewboard.asterisk.org/r/1478/
+
+2011-10-25 22:04 +0000 [r342484-342487] Richard Mudgett <rmudgett@digium.com>
+
+ * main/astobj2.c: Check fopen return value for ao2 reference debug
+ output. Reported by: wdoekes Patched by: wdoekes Review:
+ https://reviewboard.asterisk.org/r/1539/
+
+ * channels/sig_pri.c: Change D-channel warning to be less confusing
+ on non-NFAS setups. The "No D-channels available! Using Primary
+ channel as D-channel anyway!" WARNING message has been confusing
+ on non-NFAS setups. The message refers to things that are NFAS
+ specific. * Changed the warning to several different warnings to
+ be more accurate for the situation and less confusing as a
+ result: "No D-channels up! Switching selected D-channel from X to
+ Y.", "No D-channels up!", and "D-channel is down!".
+
+2011-10-25 21:08 +0000 [r342380-342435] Terry Wilson <twilson@digium.com>
+
+ * apps/app_queue.c: Use int for storing ao2_container_count instad
+ of size_t AST-676
+
+ * apps/app_queue.c: Simplify queue membercount code Despite an
+ ominous sounding comment stating that membercount was for "logged
+ in" members only and thus we couldn't use ao2_container_count(),
+ I could not find a single place in the code where that seemed to
+ be accurate. The only time we decremented membercount was when we
+ were marking something dead or actually removing it. The only
+ places we incremented it were either after ao2_link(), or trying
+ to correct for having set it to 0 during a reload. In every case
+ where we were correcting the value, it seemed that we were trying
+ to make the count actually match what ao2_container_count() would
+ return. The only place I could find where we made a determination
+ about something being "logged in" or not, we didn't trust the
+ membercount, but instead looked at devicestate, paused, etc. This
+ patch removes membercount, replaces its use with
+ ao2_container_count, and manually adds the results of
+ ao2_container_count to a "membercount" field for ast_data queue
+ query results. This patch also would fix AST-676, but as it is
+ slightly riskier than the previously committed fix, the two
+ commits have been made separately. Reivew:
+ https://reviewboard.asterisk.org/r/1541/
+
+ * apps/app_queue.c: Properly update membercount for reloaded
+ members Since q->membercount is set to 0 before reloading, it is
+ important to increment it again for reloaded members as well as
+ added. (closes issue AST-676) Review:
+ https://reviewboard.asterisk.org/r/1541/
+
+2011-10-25 19:08 +0000 [r342276-342328] Kinsey Moore <kmoore@digium.com>
+
+ * pbx/pbx_spool.c: Fix compilation on Snow Leopard/FreeBSD for
+ pbx_spool.c One of the changes in the recent spool handling of
+ hardlinks patch was just outside a HAVE_INOTIFY block and caused
+ compilation to fail in some build environments. This has been
+ corrected.
+
+ * pbx/pbx_spool.c: Fix spool handling to allow call files to be
+ hardlinked into place This fixes the inotify code to handle call
+ files being hardlinked into the spool directory. The smsq utility
+ does this, instead of rename(), to ensure that it cannot
+ accidentally overwrite an existing spool file. A rename() might
+ do that, but link() will definitely not. The inotify code had
+ broken this, because it would wait for an IN_CLOSE_WRITE event on
+ the file... which was never forthcoming, since it was never
+ opened. Now we look for IN_OPEN events following the IN_CREATE
+ event, and only wait for an IN_CLOSE_WRITE if the file was
+ actually opened. Patch-by: dwmw2 (closes issue ASTERISK-18331)
+ Review: https://reviewboard.asterisk.org/r/1391/
+
+2011-10-25 01:23 +0000 [r342223] Terry Wilson <twilson@digium.com>
+
+ * main/config.c, include/asterisk/config.h: Return NULL when no
+ results returned for realtime_multientry It was not documented
+ what the return value should be when no entries were returned
+ with the multientry realtime callback. This change forces
+ consistent behavior even if the backends return an empty
+ ast_config. Review: https://reviewboard.asterisk.org/r/1521/
+
+2011-10-24 19:49 +0000 [r342061] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c: Outbound SIP OPTIONS messages will now
+ include fromuser of related peer. This behavior matches up more
+ closely with the way invite/register/etc are handled. This patch
+ also modifies some adjacent code for code style compliance.
+ Pretty minor. (closes issue ASTERISK-17616) Reported by: Jeremy
+ Kister Patches: chan_sip.c-options-fromuser-fix-v1.patch uploaded
+ by Jeremy Kister (license #6232)
+
+2011-10-23 11:36 +0000 [r341906-341921] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * apps/app_queue.c: Revert Janitor patch 341906 For now
+
+ * apps/app_queue.c: Whitespace Fixups / Add Braces This janitorial
+ patch is related to work on RB1538
+
+2011-10-21 16:41 +0000 [r341806-341809] Matthew Nicholson <mnicholson@digium.com>
+
+ * pbx/pbx_lua.c: only process args that exist ASTERISK-18395
+
+ * pbx/pbx_lua.c: don't limit the length of app and function
+ arguments ASTERISK-18395
+
+2011-10-20 21:54 +0000 [r341717] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/features.h, main/features.c, res/res_agi.c: Fix
+ AGI exec Park to honor the Park application parameters. The fix
+ for ASTERISK-12715 and ASTERISK-12685 added a check for the Park
+ application because the channel needed to be masqueraded to
+ prevent a crash. Since the Park application now always
+ masquerades the channel into the parking lot, the special check
+ is no longer needed. The fix also resulted in AGI exec Park
+ attempting to double park the call and not honor the Park
+ application parameters. * Removed no longer necessary call to
+ ast_masq_park_call() by AGI exec for the Park application.
+ (Reverts -r146923) * Fix Park application to only return 0 or -1.
+ The AGI exec Park was causing broken pipe error messages because
+ the Park application returned 1 on successful park. (closes issue
+ ASTERISK-18737)
+
+2011-10-20 21:26 +0000 [r341664-341704] Paul Belanger <pabelanger@digium.com>
+
+ * funcs/func_callerid.c: Fixed typo from previous commit
+
+ * funcs/func_callerid.c: Updated documentation for the optional CID
+ parameter with CALLERID
+
+2011-10-20 15:11 +0000 [r341529] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/strings.h: Clean up ast_check_digits The code
+ was originally copied from the is_int() function in the AEL code.
+ wdoekes pointed out that the function should take a const char*
+ and that their was an unneeded variable. This is now fixed.
+
+2011-10-19 18:59 +0000 [r341435] Paul Belanger <pabelanger@digium.com>
+
+ * channels/chan_gtalk.c: Outgoing calls with Google Voice Google
+ has recently make some changes (again) to their protocol. Rather
+ then patching asterisk to flip between the two different methods,
+ we now allow both. Lets hope this keeps Google Voice happy for a
+ while. (closes issue ASTERISK-18714) Reported by: Iordan Iordanov
+ Patches: chan_gtalk.patch uploaded by Iordan Iordanov (licenses
+ 6311)
+
+2011-10-19 07:38 +0000 [r341379] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/strings.h, channels/chan_sip.c: Don't use
+ is_int() since it doesn't link well on all platforms Just create
+ an normal API function in strings.h that does the same thing just
+ to be safe. ASTERISK-17146
+
+2011-10-19 07:15 +0000 [r341366] Stefan Schmidt <sst@sil.at>
+
+ * channels/chan_sip.c: Don't sent in-dialog requests like UPDATE
+ when Asterisk has not yet received a Contact URI from a UAS
+
+2011-10-18 23:37 +0000 [r341314] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Don't resolve numeric hosts or contact
+ unresolved hosts If a SIP dial string contains a numeric hostname
+ that is not a peer name, don't try to resolve it as it is
+ unlikely that someone really means Dial(SIP/0.0.4.26) when
+ Dial(SIP/1050) is called. Also, make sure that create_addr
+ returns -1 if an address isn't resolved so that we don't attempt
+ to send SIP requests to an address that doesn't resolve. (closes
+ issue ASTERISK-17146, ASTERISK-17716) Review:
+ https://reviewboard.asterisk.org/r/1532/
+
+2011-10-18 23:20 +0000 [r341312] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c: fix issue on channel numbering (calls could
+ have same channel number on heavy loaded system)
+
+2011-10-18 21:03 +0000 [r341254] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_iax2.c, channels/sip/include/sip.h,
+ channels/chan_mgcp.c, include/asterisk/features.h,
+ channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/chan_sip.c, main/features.c: More parking issues. * Fix
+ potential deadlocks in SIP and IAX blind transfer to parking. *
+ Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect
+ the parkext_exclusive option with transfers
+ (Park(,,,,,exclusive_lot) parameter). Created
+ ast_park_call_exten() and ast_masq_park_call_exten() to maintian
+ API compatibility. * Made masq_park_call() handle a failed
+ ast_channel_masquerade() setup. * Reduced excessive struct
+ parkeduser.peername[] size.
+
+2011-10-17 17:35 +0000 [r341189] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Initialize variables before calling
+ parse_uri If parse_uri was called with an empty URI, some
+ pointers would be modified and an invalid read could result. This
+ patch avoids calling parse_uri with an empty contact uri when
+ parsing REGISTER requests. AST-2011-012 (closes issue
+ ASTERISK-18668)
+
+2011-10-17 16:23 +0000 [r341108-341112] Paul Belanger <pabelanger@digium.com>
+
+ * apps/app_voicemail.c: Fix previous commit
+
+ * apps/app_voicemail.c: Voicemail compiler flags are 'core' support
+
+2011-10-17 15:35 +0000 [r341088] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Don't try to remove peers without IPs from
+ peers_by_ip (closes issue ASTERISK-18696)
+
+2011-10-17 15:08 +0000 [r341074] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * pbx/pbx_realtime.c: Remove an unused include of md5.h Unused
+ include of asterisk/md5.h in pbx_realtime.c . A commit needed to
+ test the commit message.
+
+2011-10-14 21:36 +0000 [r341022] Kevin P. Fleming <kpfleming@digium.com>
+
+ * build_tools/embed_modules.xml, Makefile.moddir_rules: Change the
+ internal name of the menuselect options that are used to control
+ whether modules are embedded or not; using just the bare category
+ name led to accidentally enabling these options when users used
+ the wrong "--enable" operation on the menuselect command line.
+ Now the internal option names are prefixed with "EMBED_", so they
+ won't be the same as the name of the category containing the
+ modules they control the embedding of.
+
+2011-10-14 20:49 +0000 [r340970] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_rtp_asterisk.c, channels/chan_sip.c: Quiet RTCP Receiver
+ Reports during fax transmission RTCP is now disabled for
+ "inactive" RTP audio streams during SIP T.38 sessions. The
+ ability to disable RTCP streams in res_rtp_asterisk was missing,
+ so this code was added to support the bug fix. (closes issue
+ ASTERISK-18400)
+
+2011-10-14 16:33 +0000 [r340878] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c: Avoid unnecessary WARNING message Add
+ AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid
+ displaying a WARNING message. (closes issue ASTERISK-18610) Patch
+ by: Kristijan_Vrban
+
+2011-10-14 15:58 +0000 [r340863] Jonathan Rose <jrose@digium.com>
+
+ * codecs/codec_dahdi.c, apps/app_system.c, res/res_curl.c,
+ funcs/func_realtime.c, build_tools/cflags.xml, utils/utils.xml,
+ res/res_fax.c, apps/app_celgenuserevent.c: Fixes some support
+ level info so that it can be read by menuselect. (issue
+ ASTERISK-18268) Review: https://reviewboard.asterisk.org/r/1525/
+
+2011-10-13 22:48 +0000 [r340809] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: Fix DTMF blind transfer continuing to execute
+ dialplan after transfer. Party A calls Party B. Party A DTMF
+ blind transfers Party B to Party C. Party A channel continues to
+ execute dialplan. * Fixed the return value of
+ builtin_blindtransfer() to return the correct value after a
+ transfer so the dialplan will not keep executing. * Removed
+ unnecessary connected line update that did not really do
+ anything. * Made access to GOTO_ON_BLINDXFR thread safe in
+ check_goto_on_transfer(). * Fixed leak of xferchan for failure
+ cases in check_goto_on_transfer(). * Updated debug messages in
+ builtin_blindtransfer() and check_goto_on_transfer(). (closes
+ issue ASTERISK-18275) Reported by: rmudgett Tested by: rmudgett
+
+2011-10-13 06:58 +0000 [r340717] Stefan Schmidt <sst@sil.at>
+
+ * channels/chan_sip.c: storing the route-set also on a 181 response
+ not only on 180,182 or 183.
+
+2011-10-13 06:52 +0000 [r340662-340715] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Initialize ast_sockaddr before calling
+ ast_sockaddr_resolve Avoid possible jump based on unitialized
+ value
+
+ * res/res_config_sqlite.c: Don't skip the query field on a realtime
+ multi query There is no documented reason to not add the query
+ field to the varlist returned by a realtime multi query, despite
+ the config category being set to its value. Of course, there is
+ no documentation that the category should be set to the value
+ either. There is lots of no documentation when it comes to
+ realtime. But, other engines do not skip this field so I am
+ forcing this backend to follow the convention, because not doing
+ so is very silly.
+
+2011-10-12 20:30 +0000 [r340576] Stefan Schmidt <sst@sil.at>
+
+ * channels/chan_sip.c: Store route-set from provisional SIP
+ responses so early-dialog requests can be routed properly
+
+2011-10-12 20:19 +0000 [r340534] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Update SIP realtime fullcontact regardless
+ of caching We should update the fullcontact field in the realtime
+ table whether or not rtcachefriends is set. There is no reason to
+ treat a non-cached realtime entity differently than a cached in
+ this regard. (closes issue ASTERISK-18446) Reported by: wdoekes
+
+2011-10-12 20:07 +0000 [r340470-340522] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Initialize the PRI channel alarms properly
+ on startup. The PRI channel alarms were initialized with an
+ inverted sense. (closes issue ASTERISK-18710) Reported by:
+ Tzafrir Cohen
+
+ * apps/app_meetme.c: Update MeetMe p and X option documentation
+ when interacting with the s option. ASTERISK-12175 changed the p
+ and X options to not interfere with the s option when they are
+ used together. It makes more sense for the s option to have
+ priority for the DTMF '*' key since it cannot change its
+ activation code. Otherwise, you could not use option s with the p
+ or X options. JIRA AST-671
+
+2011-10-12 16:27 +0000 [r340418] Paul Belanger <pabelanger@digium.com>
+
+ * channels/chan_sip.c: Fix verbose messages when IPv6 logic was
+ added (closes issue ASTERISK-18612) Reported by: Tim Osman
+
+2011-10-11 21:03 +0000 [r340279-340365] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_ss7.c, channels/chan_dahdi.c, channels/sig_ss7.h:
+ Add protection for SS7 channel allocation and better glare
+ handling. * Added a CLI "ss7 show channels" command that might
+ prove useful for future debugging. * Made the incoming SS7
+ channel event check and gripe message uniform. * Made sure that
+ the DNID string for an incoming call is always initialized.
+ (issue ASTERISK-17966) Reported by: Kenneth Van Velthoven
+ Patches: jira_asterisk_17966_v1.8_glare.patch (license #5621)
+ patch uploaded by rmudgett
+
+ * channels/sip/include/dialog.h, channels/chan_sip.c: Fix some
+ potential deadlocks pointed out by helgrind. * Fixed deadlock
+ potential calling dialog_unlink_all() in __sip_autodestruct().
+ Found by helgrind. * Fixed deadlock potential in
+ handle_request_invite() after calling sip_new(). Found by
+ helgrind. * The sip_new() function now returns with the created
+ channel already locked. * Removed the dead code that starts a PBX
+ in in sip_new(). No sip_new() callers caused that code to be
+ executed and it was a bad thing to do anyway. * Removed unused
+ parameters and return value from dialog_unlink_all(). * Made
+ dialog_unlink_all() and __sip_autodestruct() safely obtain the
+ owner and private channel locks without a deadlock avoidance
+ loop.
+
+ * include/asterisk/manager.h, main/manager.c: Convert registered
+ AMI actions to ao2 objects. * Fixed race between calling an AMI
+ action callback and unregistering that action. Refixes
+ ASTERISK-13784 broken by ASTERISK-17785 change. * Fixed potential
+ memory leak if an AMI action failed to get registered because is
+ already was registered. Part of the ao2 conversion. * Fixed AMI
+ ListCommands action not walking the actions list with a lock
+ held. * Fix usage of ast_strdupa() and alloca() in loops. Excess
+ stack usage. * Fix AMI Originate action Variable header requiring
+ a space after the header colon. Reported by Yaroslav Panych on
+ the asterisk-dev list. * Increased the number of listed variables
+ allowed per AMI Originate action Variable header to 64. * Fixed
+ AMI GetConfigJSON action output format. * Fixed usage of res
+ contents outside of scope in append_channel_vars(). * Fixed
+ inconsistency of config file channelvars option. The values no
+ longer accumulate with every channelvars option in the config
+ file. Only the last value is kept to be consistent with the CLI
+ "manager show settings" command. (closes issue ASTERISK-18479)
+ Reported by: Jaco Kroon
+
+2011-10-11 00:43 +0000 [r340263] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * include/asterisk/sha1.h, main/channel.c, main/sha1.c: Update SHA1
+ code to RFC 6234 RFC 6234 is an update to RFC 3174 from which the
+ code was originally taken. It has a slightly better code, and a
+ better phrased license (simple 3-clause BSD). * main/sha1.c is
+ sha1.c from RFC 6234 with formatting changes only. *
+ include/asterisk/sha1.h merges sha.h and sha-private.h from RFC
+ 6234. * Removed unused include of asterisk/sha1.h from
+ main/channels.c Review: https://reviewboard.asterisk.org/r/1503/
+
+2011-10-10 20:23 +0000 [r340164] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_sip.c: Updated chan_sip to place calls on hold if
+ SDP address in INVITE is ANY This patch fixes the case where an
+ INVITE is received with c=0.0.0.0 or ::. In this case, the call
+ should be placed on hold. Previously, we checked for the address
+ being null; this patch keeps that behavior but also checks for
+ the ANY IP addresses. Review:
+ https://reviewboard.asterisk.org/r/1504/ (closes issue
+ ASTERISK-18086) Reported by: James Bottomley Tested by: Matt
+ Jordan
+
+2011-10-10 14:14 +0000 [r340108] Matthew Nicholson <mnicholson@digium.com>
+
+ * doc/appdocsxml.dtd, main/loader.c, main/xmldoc.c, main/pbx.c,
+ main/manager.c, res/res_fax.c, apps/app_fax.c,
+ include/asterisk/module.h, res/res_agi.c,
+ include/asterisk/xmldoc.h: Load the proper XML documentation when
+ multiple modules document the same application. This patch adds
+ an optional "module" attribute to the XML documentation spec that
+ allows the documentation processor to match apps with identical
+ names from different modules to their documentation. This patch
+ also fixes a number of bugs with the documentation processor and
+ should make it a little more efficient. Support for multiple
+ languages has also been properly implemented. ASTERISK-18130
+ Review: https://reviewboard.asterisk.org/r/1485/
+
+2011-10-09 01:16 +0000 [r339830-339938] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * channels/chan_unistim.c: Fix compilation issue, caused by missed
+ session structure (closes issue ASTERISK-18694) Reported by:
+ alex70
+
+ * channels/chan_unistim.c: Fix segfault in Unistim channel (closes
+ issue ASTERISK-18638) Reported by: jonnt
+
+ * channels/chan_unistim.c: Fix char array cast as short array in
+ send_client() function (for ARM platform) (closes issue
+ ASTERISK-17314) Reported by: jjoshua
+
+2011-10-07 19:34 +0000 [r339625-339776] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_url.c: Initialize option flags for SendURL application.
+ (closes issue ASTERISK-18574) Reported by: marcelloceschia
+
+ * autoconf/ast_ext_lib.m4, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Fix regression in
+ configure script for libpri capability checks. JIRA AST-598 added
+ the PRI_L2_PERSISTENCE option to fix BRI PTMP TE layer 2
+ persistence issues with some telcos. ASTERISK-18535 attempted to
+ fix the unexpected requirement that libpri *must* have that
+ feature to work with Asterisk. The AST_EXT_LIB_SETUP_DEPENDENT
+ lines made the PRI optional features required. Unfortunately, I
+ thought AST_EXT_LIB_SETUP_DEPENDENT didn't do anything useful for
+ libpri and deleted those lines for libpri. The result was the
+ HAVE_PRI_xxx defines that control the ability to use optional
+ libpri features were also deleted. * Created
+ AST_EXT_LIB_SETUP_OPTIONAL configuration macro to allow optional
+ features in a library that the source code could take advantage
+ of if the code supports the feature. (closes issue
+ ASTERISK-18687) Reported by: Norbert Tested by: rmudgett
+
+ * main/udptl.c, channels/chan_sip.c: Fix debugging messages
+ generated by 'udptl debug'. * Makes chan_sip set the tag to the
+ channel name. * Fixes received debug message sequence number. *
+ Removed tx/rx debug message type since it was hard coded to 0. *
+ Made udptl.c logged message header consistent if possible: "UDPTL
+ (%s): ". * Removed unused rx_expected_seq_no from struct
+ ast_udptl. (closes issue ASTERISK-18401) Reported by: Kevin P.
+ Fleming Patches: jira_asterisk_18401_v1.8.patch (license #5621)
+ patch uploaded by rmudgett Tested by: Matthew Nicholson
+
+2011-10-05 21:30 +0000 [r339566] Leif Madsen <lmadsen@digium.com>
+
+ * build_tools/prep_tarball: Update prep_tarball script to download
+ pre-exported documentation. I've updated the prep_tarball script
+ to now download the pre-exported documentation from the Asterisk
+ wiki. This will give us more control over what is being included
+ in the tarball releases, and will make both the PDF and HTML
+ exported documentation look much better (especially when viewing
+ from a console). (Closes issue ASTERISK-18677)
+
+2011-12-15 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.8.0 Released.
+
+2011-12-09 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.8.0-rc5 Released.
+
+ * Fixed crash from orphaned MWI subscriptions in chan_sip
+
+ This patch resolves the issue where MWI subscriptions are orphaned
+ by subsequent SIP SUBSCRIBE messages. When a peer is removed, either
+ by pruning realtime SIP peers or by unloading / loading chan_sip, the
+ MWI subscriptions that were orphaned would still be on the event engine
+ list of valid subscriptions but have a pointer to a peer that no longer
+ was valid. When an MWI event would occur, this would cause a seg fault.
+
+ (closes issue ASTERISK-18663)
+ Review: https://reviewboard.asterisk.org/r/1610/
+
+ * Don't crash on INFO automon request with no channel
+
+ AST-2011-014. When automon was enabled in features.conf, it was possible
+ to crash Asterisk by sending an INFO request if no channel had been
+ created yet.
+
+ (closes issue ASTERISK-18805)
+
+ * Default to nat=yes; warn when nat in general and peer differ
+
+ AST-2011-013. It is possible to enumerate SIP usernames when the general and
+ user/peer nat settings differ in whether to respond to the port a request is
+ sent from or the port listed for responses in the Via header. In 1.4 and
+ 1.6.2, this would mean if one setting was nat=yes or nat=route and the other
+ was either nat=no or nat=never. In 1.8 and 10, this would mean when one
+ was nat=force_rport and the other was nat=no.
+
+ In order to address this problem, it was decided to switch the default
+ behavior to nat=yes/force_rport as it is the most commonly used option
+ and to strongly discourage setting nat per-peer/user when at all
+ possible.
+
+ For more discussion of the issue, please see:
+ http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html
+
+ (closes issue ASTERISK-18862)
+ Review: https://reviewboard.asterisk.org/r/1591/
+
+2011-11-16 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.8.0-rc4 Released.
+
+ * Ensure that a null vmexten does not cause a segfault.
+
+ When sip_send_mwi_to_peer was modified recently to avoid deadlocks,
+ vmexten was not expected to be null. This change handles that
+ situation to avoid a segfault.
+
+ (closes issue ASTERISK-18663)
+
+2011-11-09 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.8.0-rc3 Released.
+
+ * Prevent BLF subscriptions from causing deadlocks
+
+ Fix a locking inversion in sip_send_mwi_to_peer that was causing
+ deadlocks.
+ This function now requires that both the peer and associated pvt be
+ unlocked
+ before it is called for cases where peer and peer->mwipvt form a
+ circular
+ reference.
+
+ (closes issue ASTERISK-18663)
+ Review: https://reviewboard.asterisk.org/r/1563/
+
+ * Fix deadlock if peer is destroyed while sending MWI notice.
+
+ A dialog cannot be destroyed by the ao2_callback dialog_needdestroy
+ because of a deadlock between the dialogs container lock and the
+ RWLOCK of the events subscription list.
+
+ * Create dialogs_to_destroy container to hold dialogs that will be
+ destroyed.
+
+ * Ensure that the event subscription callback will never happen with
+ an invalid peer pointer by making the event callback removal the first
+ thing in the peer destructor callback.
+
+ (closes issue ASTERISK-18747)
+ Reported by: Gregory Hinton Nietsky
+
+ Review: https://reviewboard.asterisk.org/r/1564/
+
+ * Fix issue with setting defaultenabled on categories that are already
+ enabled by default.
+
+ (closes issue ASTERISK-18738)
+ Reported by: Paul Belanger
+
+2011-10-18 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.8.0-rc2 Released.
+
+ * AST-2011-012
+
+ * menuselect/menuselect.c: Fix --enable/--enable-category.
+
+ ------------------------------------------------------------------------
+ r339719 | rmudgett | 2011-10-06 17:47:50 -0500 (Thu, 06 Oct 2011) | 20 lines
+ Fix regression in configure script for libpri capability checks.
+
+ JIRA AST-598 added the PRI_L2_PERSISTENCE option to fix BRI PTMP TE layer
+ 2 persistence issues with some telcos. ASTERISK-18535 attempted to fix
+ the unexpected requirement that libpri *must* have that feature to work
+ with Asterisk. The AST_EXT_LIB_SETUP_DEPENDENT lines made the PRI
+ optional features required. Unfortunately, I thought
+ AST_EXT_LIB_SETUP_DEPENDENT didn't do anything useful for libpri and
+ deleted those lines for libpri. The result was the HAVE_PRI_xxx defines
+ that control the ability to use optional libpri features were also
+ deleted.
+
+ * Created AST_EXT_LIB_SETUP_OPTIONAL configuration macro to allow optional
+ features in a library that the source code could take advantage of if the
+ code supports the feature.
+
+ (closes issue ASTERISK-18687)
+ Reported by: Norbert
+ Tested by: rmudgett
+ ------------------------------------------------------------------------
+ r340878 | twilson | 2011-10-14 11:33:28 -0500 (Fri, 14 Oct 2011) | 8 lines
+
+ Avoid unnecessary WARNING message
+
+ Add AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid
+ displaying a WARNING message.
+
+ (closes issue ASTERISK-18610)
+ Patch by: Kristijan_Vrban
+ ------------------------------------------------------------------------
+ r341088 | twilson | 2011-10-17 10:35:05 -0500 (Mon, 17 Oct 2011) | 4 lines
+
+ Don't try to remove peers without IPs from peers_by_ip
+
+ (closes issue ASTERISK-18696)
+ ------------------------------------------------------------------------
+
+2011-10-05 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.8.0-rc1 Released.
+
+2011-10-05 21:30 +0000 [r339566] Leif Madsen <lmadsen@digium.com>
+
+ * build_tools/prep_tarball: Update prep_tarball script to download
+ pre-exported documentation. I've updated the prep_tarball script
+ to now download the pre-exported documentation from the Asterisk
+ wiki. This will give us more control over what is being included
+ in the tarball releases, and will make both the PDF and HTML
+ exported documentation look much better (especially when viewing
+ from a console). (Closes issue ASTERISK-18677)
+
+2011-10-05 17:01 +0000 [r339506-339511] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c: Fix Dial F option notes formatting.
+
+ * main/manager.c: Fix XML error in AMI action Challenge.
+
+2011-10-05 16:31 +0000 [r339505] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: The app name in the documentation must match what
+ we register the application as.
+
+2011-10-05 16:26 +0000 [r339406-339504] Richard Mudgett <rmudgett@digium.com>
+
+ * main/manager.c: Add missing documentation of required AMI action
+ Challenge AuthType header. (closes issue ASTERISK-18554) Reported
+ by: Vlad Povorozniuc Patches:
+ __20110919-manager-challenge-docs.patch.txt (license #4999) patch
+ uploaded by Leif Madsen
+
+ * Makefile: Make always create the MOH directory
+ (/var/lib/asterisk/moh). (closes issue ASTERISK-18409) Reported
+ by: abelbeck Patches: asterisk-1.8-makefile-moh.patch (license
+ #5903) patch uploaded by abelbeck Tested by: abelbeck, Michael
+ Keuter
+
+2011-10-04 19:33 +0000 [r339297-339352] Jonathan Rose <jrose@digium.com>
+
+ * main/say.c: Removes improper use of sound 'and' in German
+ language mode from application saynumber Asterisk would say 'Five
+ hundert und sechs und zwanzig' instead of 'Five hundert sechs und
+ zwanzig'... which is both weird sounding and wrong. This patch
+ makes sure Asterisk will only say the 'and' word between the
+ single digit and double digit places. (closes issue
+ ASTERISK-18212) Reported By: Lionel Elie Mamane Patches:
+ upstream_germand_no_and.diff (License #5402) uploaded by Lionel
+ Elie Mamane
+
+ * res/res_jabber.c: Reverting revision 333265 due to component
+ connection problems it introduces. I'm going to attempt some
+ generic res_jabber cleanup and come up with a new fix for this
+ problem, but first it seems prudent to remove this rather broad
+ attempt to fix it and instead approach this problem either from
+ the same angle but looking only at canceling (or possibly
+ rescheduling) the send when we absolutely know it will cause a
+ segfault or, if that can't be easily accomplished, strictly from
+ the devstate side of things. Also, I'm pretty sure a lot of the
+ code in res_jabber isn't thread safe. (issue ASTERISK-18626)
+ (issue ASTERISK-18078)
+
+2011-10-04 11:44 +0000 [r339244] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/memheap.c: fix forget declaration in previous
+ change
+
+2011-10-03 20:12 +0000 [r339144-339147] Leif Madsen <lmadsen@digium.com>
+
+ * channels/chan_sip.c: Remove duplicated Maxforwards line in AMI
+ output. (Closes issue ASTERISK-18637) Reported by: Jacek
+ Konieczny Patches: asterisk-sipshowpeer.patch (License #6298)
+ uploaded by Jacek Konieczny
+
+ * apps/app_dial.c: Make documentation for Dial() options 'F' and
+ 'F()' more clear. (Closes issue ASTERISK-18646) Reported by:
+ Physis Heckman Tested by: Richard Mudgett
+
+2011-10-03 18:42 +0000 [r339087] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/memheap.c: destroy memheap mutex properly
+ before memheap deleted (fix memory leak occured after r304950
+ changes with DEBUG_THREAD compile option)
+
+2011-10-03 18:40 +0000 [r339086] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c, main/file.c: Properly ignore
+ AST_CONTROL_UPDATE_RTP_PEER in more places After the change in
+ r336294, the new AST_CONTROL_UPDATE_RTP_PEER frame is sent when a
+ re-invite happens. If we receive a re-invite from a device the
+ waitstream_core was not aware of the new control frame and would
+ drop the call. (closes issue ASTERISK-18610) Reported by:
+ Kristijan_Vrban
+
+2011-09-30 22:05 +0000 [r338800] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Fix segfault in analog_ss_thread() not
+ checking ast_read() for NULL. NOTE: The problem was reported
+ against v1.6.2. It is unlikely to ever happen on v1.8 and above
+ since chan_dahdi.c:analog_ss_thread() is unlikely to be used. The
+ version in sig_analog.c has largely replaced it. (closes issue
+ ASTERISK-18648) Reported by: Stephan Bosch Patches:
+ jira_asterisk_18648_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: Stephan Bosch
+
+2011-09-30 18:54 +0000 [r338718] Jonathan Rose <jrose@digium.com>
+
+ * configs/queues.conf.sample: Adds documentation for
+ QueueMemberStatus event generation
+
+2011-09-30 16:27 +0000 [r338663] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Fix formatting of AMI header for SIP show
+ peer. ASTERISK-17486 exposed the problem for AMI parsers. (closes
+ issue ASTERISK-18649) Reported by: Jacek Konieczny Patches:
+ asterisk-sipshowpeer_response_end.patch (license #6298) patch
+ uploaded by Jacek Konieczny
+
+2011-09-30 09:31 +0000 [r338609] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c, configure.ac: Remove r338137 and r338138.
+
+2011-09-29 21:12 +0000 [r338555] Paul Belanger <pabelanger@digium.com>
+
+ * tests/test_linkedlists.c, tests/test_amihooks.c,
+ tests/test_security_events.c, tests/test_locale.c,
+ tests/test_logger.c, tests/test_dlinklists.c: Test modules should
+ depend on the TEST_FRAMEWORK flag
+
+2011-09-29 20:54 +0000 [r338551] Jason Parker <jparker@digium.com>
+
+ * tests/test_db.c, tests/test_netsock2.c: Test modules have a
+ support level of core.
+
+2011-09-29 18:31 +0000 [r338492] Leif Madsen <lmadsen@digium.com>
+
+ * channels/chan_sip.c: Update documentation for SIP_HEADER. The
+ SIP_HEADER function only works on the the initial SIP INVITE. The
+ documentation was updated in trunk, but not in 1.8 or 10, so I'm
+ making them match. (Closes issue ASTERISK-18640)
+
+2011-09-29 12:13 +0000 [r338416] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: The rtptimeout
+ setting is ignored on a per peer basis. Not only is the
+ rtptimeout ignored in some cases but rtpkeepalive and
+ rtpholdtimeout is affected. this commit also removes
+ rtptimeout/rtpholdtimeout on text rtp. (closes issue
+ ASTERISK-18559) Review: https://reviewboard.asterisk.org/r/1452
+
+2011-09-28 22:35 +0000 [r338235-338322] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Make duplicate call ptr warning message more
+ helpful. * Adds the value of the call ptr to the duplicate call
+ ptr message to help trace why there is a duplicate call ptr.
+
+ * include/asterisk/logger.h: Fix inconsistency in
+ LOG_VERBOSE/AST_LOG_VERBOSE declaration. (closes issue
+ ASTERISK-17973) Reported by: Luke H Patches: logger_h.patch
+ (license #6278) patch uploaded by Luke H
+
+2011-09-28 20:52 +0000 [r338227] Jason Parker <jparker@digium.com>
+
+ * tests/test_db.c, tests/test_netsock2.c, build_tools/cflags.xml,
+ channels/chan_usbradio.c, build_tools/cflags-devmode.xml,
+ agi/agi.xml, utils/utils.xml, build_tools/embed_modules.xml: Add
+ support levels to non-module sections of menuselect (cflags,
+ utils, etc).
+
+2011-09-28 20:24 +0000 [r338224] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Fix chan_dahd compiling with gcc 4.6 when
+ PRI and SS7 not present. (closes issue ASTERISK-18357) Reported
+ by: Matthew Nicholson
+
+2011-09-28 07:28 +0000 [r338137-338138] TransNexus OSP Development <support@transnexus.com>
+
+ * configure.ac: Updated for checking OSP Toolkit version 4.0.0.
+
+ * apps/app_osplookup.c: Updated for OSP Toolkit 4.0.0.
+
+2011-09-27 20:10 +0000 [r338084] Paul Belanger <pabelanger@digium.com>
+
+ * apps/app_macro.c: Upgrade app_macro to core
+
+2011-09-26 19:30 +0000 [r337973] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/channel.h, main/cel.c, main/manager.c,
+ funcs/func_odbc.c, cel/cel_custom.c, apps/app_minivm.c,
+ main/logger.c, cel/cel_sqlite3_custom.c, cdr/cdr_manager.c,
+ cdr/cdr_custom.c, apps/app_voicemail.c, apps/app_dial.c,
+ main/pbx.c, cdr/cdr_sqlite3_custom.c, cdr/cdr_syslog.c,
+ tests/test_gosub.c, include/asterisk/cel.h: Fix deadlock when
+ using dummy channels. Dummy channels created by
+ ast_dummy_channel_alloc() should be destoyed by
+ ast_channel_unref(). Using ast_channel_release() needlessly grabs
+ the channel container lock and can cause a deadlock as a result.
+ * Analyzed use of ast_dummy_channel_alloc() and made use
+ ast_channel_unref() when done with the dummy channel. (Primary
+ reason for the reported deadlock.) * Made
+ app_dial.c:dial_exec_full() not call ast_call() holding any
+ channel locks. Chan_local could not perform deadlock avoidance
+ correctly. (Potential deadlock exposed by this issue. Secondary
+ reason for the reported deadlock since the held lock was part of
+ the deadlock chain.) * Fixed some uses of
+ ast_dummy_channel_alloc() not checking the returned channel
+ pointer for failure. * Fixed some potential chan=NULL pointer
+ usage in func_odbc.c. Protected by testing the bogus_chan value.
+ * Fixed needlessly clearing a 1024 char auto array when setting
+ the first char to zero is enough in manager.c:action_getvar().
+ (closes issue ASTERISK-18613) Reported by: Thomas Arimont
+ Patches: jira_asterisk_18613_v1.8.patch (license #5621) patch
+ uploaded by rmudgett Tested by: Thomas Arimont
+
+2011-09-23 19:14 +0000 [r337839-337898] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * contrib/init.d/rc.archlinux.asterisk: Spelling fix
+
+ * apps/app_queue.c: Make sure a CDR is on the stack for call in the
+ Queue. Only let update_cdr act on the last CDR in the stack. In
+ some circumstances [Attended transfer to queue] a CDR record is
+ not inserted for this call where it should. (closes issue
+ ASTERISK-18567) Review: https://reviewboard.asterisk.org/r/1266
+
+2011-09-23 00:44 +0000 [r337774] Russell Bryant <russell@digium.com>
+
+ * configs/res_pktccops.conf.sample: Comment out entries in sample
+ res_pktccops.conf. With these options enabled, they can cause
+ Asterisk to freak out by SYN flooding a network and eating the
+ CPU. Obviously it would be good to fix the code so that this
+ can't happen, but we can at least change the default
+ configuration so it doesn't happen. This was reported downstream
+ to the Fedora issue tracker:
+ https://bugzilla.redhat.com/show_bug.cgi?id=658431
+
+2011-09-22 21:29 +0000 [r337720] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Made ISDN not add numbering plan prefix
+ strings to empty numbers. When the Caller-ID is restricted, the
+ expected behavior is for the Caller-ID to be blank. In
+ chan_dahdi, the national prefix is placed onto the Caller-ID
+ number even if it is restricted (empty) causing the Caller-ID to
+ be the national prefix rather than blank. This behavior was lost
+ when sig_pri was extracted from chan_dahdi. * Made not add prefix
+ strings to empty connected line, calling, and ANI number strings.
+ (closes issue ASTERISK-18577) Reported by: Kris Shaw Patches:
+ jira_asterisk_18577_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: Kris Shaw
+
+2011-09-22 11:39 +0000 [r337430-337541] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * res/res_srtp.c: Add warned to ast_srtp to prevent errors on each
+ frame from libsrtp The first 9 frames are not reported as some
+ devices dont use srtp from first frame these are suppresed. the
+ warning is then output only once every 100 frames.
+
+ * channels/chan_h323.c: If IP address is used in chan_h323 host
+ parameter of peer configuration. module tries to resolve IP
+ address to IP address and fails. Simple fix to set family of
+ socket this is a hangover from ipv6 changes. (closes issue
+ ASTERISK-18237) (issue ASTERISK-17278) (issue ASTERISK-17500)
+
+ * main/channel.c: Its possible to loose audio on ast_write when the
+ channel is not transcoded correctly. in the case of DAHDI the
+ channel is hungup. This patch tries to "fix" the problem and make
+ the channel compatiable and warn the user of this problem. Please
+ note there is a underlying problem with codec negotion this does
+ not fix the problem it does try to rectify it and prevent loss of
+ service. Review: https://reviewboard.asterisk.org/r/1442/ (closes
+ issue ASTERISK-17541) (closes issue ASTERISK-18063) (issue
+ ASTERISK-14384) (issue ASTERISK-17502) (issue ASTERISK-18325)
+ (issue ASTERISK-18422)
+
+2011-09-21 21:18 +0000 [r337325-337353] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * apps/app_voicemail.c: More silly spacing changes
+
+ * apps/app_voicemail.c: Dumb little spacing fix.
+
+ * funcs/func_curl.c: Escape commas in keys and values, when keys
+ and values are enumerated by commas. Review:
+ https://reviewboard.asterisk.org/r/1433
+
+2011-09-20 22:38 +0000 [r337118] Matthew Jordan <mjordan@digium.com>
+
+ * main/app.c, apps/app_followme.c, apps/app_voicemail.c,
+ apps/app_dial.c, include/asterisk/app.h, apps/app_meetme.c,
+ apps/app_minivm.c: Fix for incorrect voicemail duration in
+ external notifications This patch fixes an issue where the
+ voicemail duration was being reported with a duration
+ significantly less than the actual sound file duration.
+ Voicemails that contained mostly silence were reporting the
+ duration of only the sound in the file, as opposed to the
+ duration of the file with the silence. This patch fixes this by
+ having two durations reported in the __ast_play_and_record family
+ of functions - the sound_duration and the actual duration of the
+ file. The sound_duration, which is optional, now reports the
+ duration of the sound in the file, while the actual full duration
+ of the file is reported in the duration parameter. This allows
+ the voicemail applications to use the sound_duration for minimum
+ duration checking, while reporting the full duration to external
+ parties if the voicemail is kept. (issue ASTERISK-2234) (closes
+ issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad
+ House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1443
+
+2011-09-20 22:18 +0000 [r337115] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/init.d/rc.redhat.asterisk: Update RedHat Init script to
+ work with Heartbeat. The current RedHat init script was not LSB
+ compatible. This change will make it LSB compatible so that it
+ can work correctly with Heartbeat. (Closes issue ASTERISK-18253)
+ Reported by: c0rnoTa
+
+2011-09-20 21:04 +0000 [r337061] Kinsey Moore <kmoore@digium.com>
+
+ * tests/test_pbx.c, main/pbx.c: Make CANMATCH with the new pattern
+ match engine behave more like the old one When checking an
+ extension for E_CANMATCH using the new extension matching
+ algorithm, an exact match was not returned as a possible match
+ resulting in the queue failing to allow a caller to exit on DTMF.
+ This removes the requirement that an extension be longer than
+ acquired digits for an E_CANMATCH operation to succeed. (closes
+ issue ASTERISK-18044) Review:
+ https://reviewboard.asterisk.org/r/1367/
+
+2011-09-20 19:10 +0000 [r336977-337007] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_ss7.c: Check if a channel was created before using
+ the pointer in sig_ss7_new_ast_channel(). Fixes the crash in
+ ASTERISK-17955 gdb-11918.txt backtrace. * Added some missing
+ libss7 access lock protection. * Prevent cancelling the
+ ss7_linkset() thread at inoportune times just like the
+ pri_dchannel() thread. (issue ASTERISK-17955) Reported by: Ian M
+ Sherman Patches: jira_asterisk_17955_v1.8.patch (license #5621)
+ patch uploaded by rmudgett (attached to related ASTERISK-17966)
+
+ * channels/sig_ss7.c: Fix deadlock from not releasing SS7 linkset
+ lock. sig_ss7_hangup() failed to release the SS7 linkset lock if
+ the call had the alreadyhungup flag set. * Made unlock the SS7
+ linkset lock in sig_ss7_hangup() if the alreadyhungup flag is
+ set. * Made ss7_start_call() not hold any locks while creating
+ the channel for an incoming call to prevent deadlock. * Made
+ ss7_grab() a void function, since it could never fail, to
+ simplify calling code. * Made obtain the channel lock to do
+ softhangup in some places. Patches: jira_ast_668_v1.8.patch
+ (license #5621) patch uploaded by rmudgett JIRA AST-668
+
+2011-09-20 00:56 +0000 [r336877] Russell Bryant <russell@digium.com>
+
+ * res/res_rtp_asterisk.c: Fix crashes in ast_rtcp_write(). This
+ patch addresses crashes related to RTCP handling. The backtraces
+ just show a crash in ast_rtcp_write() where it appears that the
+ RTP instance is no longer valid. There is a race condition with
+ scheduled RTCP transmissions and the destruction of the RTP
+ instance. This patch utilizes the fact that ast_rtp_instance is a
+ reference counted object and ensures that it will not get
+ destroyed while a reference is still around due to scheduled RTCP
+ transmissions. RTCP transmissions are scheduled and executed from
+ the chan_sip scheduler context. This scheduler context is
+ processed in the SIP monitor thread. The destruction of an RTP
+ instance occurs when the associated sip_pvt gets destroyed (which
+ happens when the sip_pvt reference count reaches 0). However, the
+ SIP monitor thread is not the only thread that can cause a
+ sip_pvt to get destroyed. The sip_hangup function, executed from
+ a channel thread, also decrements the reference count on a
+ sip_pvt and could cause it to get destroyed. While this is being
+ changed anyway, the patch also removes calling ast_sched_del()
+ from within the RTCP scheduler callback. It's not helpful. Simply
+ returning 0 prevents the callback from being rescheduled. (closes
+ issue ASTERISK-18570) Related issues that look like they are the
+ same problem: (issue ASTERISK-17560) (issue ASTERISK-15406)
+ (issue ASTERISK-15257) (issue ASTERISK-13334) (issue
+ ASTERISK-9977) (issue ASTERISK-9716) Review:
+ https://reviewboard.asterisk.org/r/1444/
+
+2011-09-19 22:07 +0000 [r336791] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Don't interfere with T.38 reinvites This is
+ an update to the fix for ASTERISK-18340 and ASTERISK-17725
+
+2011-09-19 20:27 +0000 [r336733] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * Makefile.rules, include/asterisk/optional_api.h, Makefile,
+ configure, include/asterisk/autoconfig.h.in, main/Makefile,
+ codecs/gsm/Makefile, configure.ac: Various changes to allow 1.8
+ to compile on Mac OS X Lion (10.7) * Makefile workaround for 10.6
+ extended to work on 10.7 and later. * Now uses the 'weak' symbol
+ for Lion systems, which no longer support 'weak_import' Closes
+ ASTERISK-17612. Closes ASTERISK-18213. Tested by: tilghman, oej.
+
+2011-09-19 20:07 +0000 [r336716] Jonathan Rose <jrose@digium.com>
+
+ * res/res_musiconhold.c, apps/app_queue.c, apps/app_mixmonitor.c,
+ apps/app_echo.c, apps/app_saycounted.c, apps/app_mp3.c,
+ apps/app_morsecode.c: Document applications that play audio and
+ do not answer unanswered calls. This patch is part of an effort
+ to document early media and its usage. If you are interested in
+ contributing to this documentation effort, there are probably
+ other applications worth documenting as well as an Asterisk wiki
+ article at
+ https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application
+
+2011-09-19 18:46 +0000 [r336658] Richard Mudgett <rmudgett@digium.com>
+
+ * UPGRADE.txt, apps/app_dial.c: Made Dial d and H options no longer
+ immediately auto-answer the calling leg. The Dial d and H options
+ break DTMF attended transfer atxferdropcall option. 1) Party A
+ calls party B. 2) Party B does a DTMF attended transfer to Party
+ C. If the dialplan uses the Dial d or H options to call Party C
+ then the Dial application answers the call immediately before
+ initiating the call leg to Party C. The premature answer causes
+ the transfer code to not invoke the atxferdropcall=no behavior
+ for a blonde transfer since Party C has "answered". The transfer
+ code thinks that Party B has "consulted" with Party C when Party
+ B hangs up and completes the transfer to Party A. Party A now
+ hears ringback until Party C actually answers. ASTERISK-13294
+ Dial d option. ASTERISK-11067 Dial H option to disconnect before
+ answer. The referenced issues made Dial answer with the d and H
+ options because many SIP and ISDN phones cannot send DTMF before
+ the call is connected. * Made require the dialplan to control
+ when or if the call needs to be answered to use the Dial
+ application d and H options. (The call is no longer surprise
+ answered when using the Dial d or H options.) Review:
+ https://reviewboard.asterisk.org/r/1381/ JIRA AST-623 JIRA
+ AST-666
+
+2011-09-19 16:21 +0000 [r336591] Jason Parker <jparker@digium.com>
+
+ * contrib/realtime/postgresql/realtime.sql,
+ configs/cel_odbc.conf.sample, sounds/Makefile,
+ contrib/realtime/mysql/sipfriends.sql,
+ contrib/realtime/mysql/voicemail.sql, cel/cel_odbc.c, /,
+ contrib/realtime/mysql/iaxfriends.sql,
+ contrib/realtime/mysql/meetme.sql: Remove weird mergeinfo props
+ that make merges annoying sometimes.
+
+2011-09-19 15:41 +0000 [r336572] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/get_ilbc_source.sh: Update get_ilbc_source.sh
+ script to work again. Recently iLBC support in Asterisk has
+ changed after the acquisition of GIPS by Google. More information
+ about how this may affect you is available in a blog post at:
+ http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/
+
+2011-09-19 15:25 +0000 [r336569] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Rework sig_pri_hangup() to be simpler and
+ clearer. JIRA AST-675
+
+2011-09-19 13:33 +0000 [r336501] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Add diversion header to a 302 redirect
+ response if we have diversion data (closes issue ASTERISK-18143)
+ patch by oej
+
+2011-09-19 13:27 +0000 [r336499] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * channels/chan_h323.c: A long time ago in a galaxy far far away a
+ IPv6 update was made, chan_h323 was not updated causeing all to
+ flee to chan_ooh323. the brave Jedi [asterisk developers]
+ pondered this miscarrige of justice and restored order to the
+ force for the sake of closing out 2 old issues. (closes issue
+ ASTERISK-17278) (closes issue ASTERISK-17500) Reported by: dread,
+ sybasesql Tested by: irroot Reviewed by: IRC (russellb,
+ kpfleming)
+
+2011-09-19 12:06 +0000 [r336378-336440] Olle Johansson <oej@edvina.net>
+
+ * main/manager.c: Make sure manager_debug option is reset at reload
+
+ * Makefile: Revert accidental change that fixes OS/X Lion support
+
+ * Makefile, channels/chan_sip.c: Add missing unlock at MWI message
+ sending time (closes issue ASTERISK-18573) Patches:
+ sip_mwi_lock.patch (license #5041) by Gregory Hinton Nietsky
+ Thanks to irrot for the reminder, to Gregory for the patch!
+
+2011-09-16 22:10 +0000 [r336312-336314] Terry Wilson <twilson@digium.com>
+
+ * funcs/func_frame_trace.c: Whitespace fix
+
+ * funcs/func_frame_trace.c: Add missing frame types to
+ func_frame_trace Also casts control frames to the proper enum so
+ that the compile will catch new additions.
+
+2011-09-16 19:53 +0000 [r336294] Jonathan Rose <jrose@digium.com>
+
+ * include/asterisk/frame.h, main/channel.c, main/rtp_engine.c,
+ channels/chan_sip.c: Fix bad RTP media bridges in directmedia
+ calls on peers separated by multiple Asterisk nodes. In a
+ situation involving devices on separate Asterisk trunks, the
+ remote RTP bridge would break when starting a call with
+ directmedia. This patch queues a new type of control frame so
+ that our RTP bridge loop can properly detect when these
+ situations occur and check to see if peers need to be updated in
+ order to send their media to the proper location. (Closes issue
+ ASTERISK-18340) Reported by: Thomas Arimont (Closes issue
+ ASTERISK-17725) Reported by: kwk Tested by: twilson, jrose
+
+2011-09-16 19:06 +0000 [r336234] Sean Bright <sean@malleable.com>
+
+ * UPGRADE.txt: Make a note that inotify won't work with an NFS
+ mounted spooler directory.
+
+2011-09-16 10:09 +0000 [r335978-336166] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * channels/chan_misdn.c: The round robin routing routine in
+ chan_misdn.c is broken. it rotates between ports but never checks
+ the channels in the ports. i have extensivly tested it and
+ verified it works on 1 upto 4 ports. before the patch only 1 out
+ of each port was used now all are used as expected. (closes issue
+ ASTERISK-18413) Reported by: irroot Tested by: irroot Reviewed
+ by: irroot Review: https://reviewboard.asterisk.org/r/1410/
+
+ * apps/app_queue.c: Locking order in app_queue.c causes deadlocks.
+ a channel lock must never be held with the queues container lock
+ held. the deadlock occured on masquerade. the queues container
+ lock is a relic of the past the old queue module lock. with ao2
+ there is no need to hold this lock when dealing with members this
+ patch removes unneeded locks. (closes issue ASTERISK-18101)
+ (closes issue ASTERISK-18487) Reported by: Paul Rolfe, Jason
+ Legault Tested by: irroot, Jason Legault, Paul Rolfe Reviewed by:
+ Matthew Nicholson Review:
+ https://reviewboard.asterisk.org/r/1402/
+
+ * channels/chan_agent.c: lock the channel before calling
+ ast_bridged_channel() to prevent a seg fault. AMI agents list
+ called on shutdown causes a segfault, introducing proper locking
+ will prevent this. (closes issue ASTERISK-18092) Reported by:
+ agustina Patches: chan_agent.patch (License #5041) patch uploaded
+ by irroot
+
+2011-09-14 18:21 +0000 [r335851-335911] Richard Mudgett <rmudgett@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac: Remove
+ unnecessary libpri dependency checks in the configure script.
+ Using the --with-pri option with the configure script generated
+ an error about not having PRI_L2_PERSISTENCE if you did not have
+ the absolute latest libpri SVN checkout installed. The
+ AST_EXT_LIB_SETUP_DEPENDENT macro in the configure.ac script
+ seems to be for libraries that are dependent upon other libraries
+ and not necessarily for optional/added features within a library.
+ (closes issue ASTERISK-18535) Reported by: Michael Keuter
+
+ * channels/chan_dahdi.c: Fixed cut-n-paste regression using the
+ wrong variable. Fixes the missing DAHDI channels when using the
+ newer chan_dahdi.conf sections for channel configuration. (closes
+ issue ASTERISK-18496) Reported by: Sean Darcy Patches:
+ jira_asterisk_18496_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: Sean Darcy, rmudgett
+
+2011-09-14 13:28 +0000 [r335790] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/manager.c: The tech and data members of
+ fast_originate_helper are not string fields. ASTERISK-17709
+
+2011-09-13 22:10 +0000 [r335720] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_directed_pickup.c: Remove obsolete todo comment about
+ PICKUPRESULT.
+
+2011-09-13 21:33 +0000 [r335716] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * main/asterisk.c: do parse defaultlanguage from asterisk.conf Do
+ parse the option "defaultlanguage" from the [options] section of
+ asterisk.conf, as in the sample config file. Otherwise the
+ build-time default language (normally "en") is always the default
+ one. Review: https://reviewboard.asterisk.org/r/1342/
+ Signed-off-by: Tzafrir Cohen (License #5035)
+ <tzafrir.cohen@xorcom.com>
+
+2011-09-13 21:30 +0000 [r335714] Paul Belanger <pabelanger@digium.com>
+
+ * apps/app_meetme.c: Meetme should have 'core' support level
+ (closes issue ASTERISK-18542)
+
+2011-09-13 18:52 +0000 [r335655] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * configure, configure.ac: Move mandatory checks closer to the
+ beginning of the file. If these are going to fail, they should
+ fail as quickly as possible.
+
+2011-09-13 18:20 +0000 [r335618] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, main/manager.c: Don't limit the size of appdata for
+ manager originate actions. ASTERISK-17709 Patch by: tilghman
+ (with modifications)
+
+2011-09-13 07:11 +0000 [r335497] Russell Bryant <russell@digium.com>
+
+ * main/event.c, include/asterisk/event.h, res/ais/evt.c: Fix a
+ crash in res_ais. This patch resolves a crash observed in a load
+ testing environment that involved the use of the res_ais module.
+ I observed some crashes where the event delivery callback would
+ get called, but the length parameter incidcating how much data
+ there was to read was 0. The code assumed (with good reason I
+ would think) that if this callback got called, there was an event
+ available to read. However, if the rare case that there's nothing
+ there, catch it and return instead of blowing up. More
+ specifically, the change always ensure that the size of the
+ received event in the cluster is always big enough to be a real
+ ast_event. Review: https://reviewboard.asterisk.org/r/1423/
+
+2011-09-12 15:54 +0000 [r335431-335433] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c: Properly set caller_warning and callee_warning
+ before we try to use them. ASTERISK-18199 Patch by: elguero
+ Testing by: rtang
+
+ * bridges/bridge_multiplexed.c: Prevent a race condition when the
+ bridge technology changes. This change was ported from asterisk
+ 10. ASTERISK-18155
+
+2011-09-12 14:21 +0000 [r335320-335341] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_dial.c: Ensure frames are not written to dialed channel
+ if ringback is requested When a single channel was dialed and
+ there was media to be forwarded to the calling channel, the media
+ was written without regard for ringback causing silence to be
+ heard in some circumstances. This regression was introduced when
+ the meaning of "single" changed to mean only the number of
+ channels dialed. (closes issue ASTERISK-18083)
+
+ * channels/chan_iax2.c: Prevent IAX2 from getting IPv6 addresses
+ via DNS IAX2 does not support IPv6 and getting such addresses
+ from DNS can cause error messages on the remote end involving bad
+ IPv4 address casts in the presence of IPv6/IPv4 tunnels. This
+ patch ensures that IAX2 will not encounter IPv6 addresses via DNS
+ queries. (closes issue ASTERISK-18090)
+
+2011-09-12 13:25 +0000 [r335319] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Lock the peer->mvipvt to avoid crashes with
+ SIP history enabled After the launch of 1.6 event-based MWI we
+ have two threads handling the peer->mwipvt, which cause issues
+ with SIP history additions in combination with the max limit for
+ number of history entries. Review:
+ https://reviewboard.asterisk.org/r/1373/ (closes issue
+ ASTERISK-18288) Thanks to irrot for peer review. Work with this
+ bug funded by IPvision AS
+
+2011-09-12 11:09 +0000 [r335259] Stefan Schmidt <sst@sil.at>
+
+ * channels/chan_sip.c: build_peer doesnt unlink a peer object from
+ peers_by_ip container which leads to a wrong refcounter value.
+ adding an ao2_unlink from the peers_by_ip container fix it.
+ Review: https://reviewboard.asterisk.org/r/1428/
+
+2011-09-09 16:09 +0000 [r335064] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_console.c, channels/sig_pri.c, channels/chan_oss.c,
+ main/channel.c, channels/chan_usbradio.c, main/dial.c,
+ channels/chan_dahdi.c, channels/chan_misdn.c,
+ channels/chan_skinny.c, funcs/func_frame_trace.c,
+ main/features.c, channels/chan_h323.c, channels/chan_alsa.c,
+ include/asterisk/frame.h, channels/sig_ss7.c,
+ channels/chan_mgcp.c, apps/app_dial.c, channels/chan_unistim.c,
+ main/pbx.c, addons/chan_ooh323.c, channels/chan_sip.c: Updated
+ SIP 484 handling; added Incomplete control frame When a SIP phone
+ uses the dial application and receives a 484 Address Incomplete
+ response, if overlapped dialing is enabled for SIP, then the 484
+ Address Incomplete is forwarded back to the SIP phone and the
+ HANGUPCAUSE channel variable is set to 28. Previously, the
+ Incomplete application dialplan logic was automatically
+ triggered; now, explicit dialplan usage of the application is
+ required. Additionally, this patch adds a new AST_CONTOL_FRAME
+ type called AST_CONTROL_INCOMPLETE. If a channel driver receives
+ this control frame, it is an indication that the dialplan expects
+ more digits back from the device. If the device supports overlap
+ dialing it should attempt to notify the device that the dialplan
+ is waiting for more digits; otherwise, it can handle the frame in
+ a manner appropriate to the channel driver. (closes issue
+ ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew
+ Jordan Review: https://reviewboard.asterisk.org/r/1416/
+
+2011-09-08 22:27 +0000 [r334953] Richard Mudgett <rmudgett@digium.com>
+
+ * main/logger.c: Fix crash with res_fax when MALLOC_DEBUG and "core
+ stop gracefully" are used. Asterisk crashes if MALLOC_DEBUG is
+ enabled when res_fax tries to unregister its logger level. * Make
+ ast_logger_unregister_level() use ast_free() instead of free().
+ When MALLOC_DEBUG is enabled, ast_free() does not degenerate into
+ a call to free(). Therefore, if you allocated memory with a form
+ of ast_malloc you must free it with ast_free.
+
+2011-09-07 19:35 +0000 [r334843] Paul Belanger <pabelanger@digium.com>
+
+ * channels/chan_iax2.c: Cleanup chan_iax2.c log messages Review:
+ https://code.asterisk.org/code/cru/CR-AST-11
+
+2011-09-07 19:31 +0000 [r334840] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: Fix AMI action Park crash. * Made AMI action
+ Park not say anything to the parker channel (AMI header Channel2)
+ since the AMI action is a third party parking the call. (This is
+ a change in behavior that cannot be preserved without a lot of
+ effort.) * Made not play pbx-parkingfailed if the Park 's' option
+ is used. JIRA AST-660
+
+2011-09-07 13:26 +0000 [r334682] Stefan Schmidt <sst@sil.at>
+
+ * main/features.c: Adding the Feature to sent a Reason Header in a
+ SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE
+ before doing a masquerade in the pickup function.
+
+2011-09-07 08:12 +0000 [r334616-334620] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * CHANGES, apps/app_queue.c: peroid typo
+
+ * main/pbx.c: Prevent segfault if call arrives before Asterisk is
+ fully booted. Prevent ast_pbx_start and ast_run_start from
+ starting a new thread unless asterisk is fully booted. alecdavis
+ (license 585) Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1407/
+
+2011-09-06 13:48 +0000 [r334453] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * apps/app_voicemail.c: Make SQL query in app_voicemail.c portable
+ LIMIT is not portable. Regression from r312212 (closes issue
+ ASTERISK-18255) Reported by: Leif Madsen Tested by: Leif Madsen
+ Review: https://reviewboard.asterisk.org/r/1415/
+
+2011-09-23 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.7.0 Released.
+
+2011-09-19 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.7.0-rc2 Released.
+
+ * r335851 | rmudgett | 2011-09-14 10:53:25 -0500 (Wed, 14 Sep 2011) |
+ 11 lines
+
+ Fixed cut-n-paste regression using the wrong variable.
+
+ Fixes the missing DAHDI channels when using the newer chan_dahdi.conf
+ sections for channel configuration.
+
+ (closes issue ASTERISK-18496)
+
+ * r335911 | rmudgett | 2011-09-14 13:21:35 -0500 (Wed, 14 Sep 2011) |
+ 13 lines
+
+ Remove unnecessary libpri dependency checks in the configure script.
+
+ Using the --with-pri option with the configure script generated an
+ error
+ about not having PRI_L2_PERSISTENCE if you did not have the absolute
+ latest libpri SVN checkout installed.
+
+ The AST_EXT_LIB_SETUP_DEPENDENT macro in the configure.ac script seems
+ to
+ be for libraries that are dependent upon other libraries and not
+ necessarily for optional/added features within a library.
+
+ (closes issue ASTERISK-18535)
+
+ * r336572 | lmadsen | 2011-09-19 10:41:16 -0500 (Mon, 19 Sep 2011) | 7
+ lines
+
+ Update get_ilbc_source.sh script to work again.
+
+ Recently iLBC support in Asterisk has changed after the acquisition of
+ GIPS
+ by Google. More information about how this may affect you is available
+ in a
+ blog post at:
+
+ http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/
+
+ * r335714 | pabelanger | 2011-09-13 16:30:18 -0500 (Tue, 13 Sep 2011)
+ | 4 lines
+
+ Meetme should have 'core' support level
+
+ (closes issue ASTERISK-18542)
+
+2011-09-07 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.7.0-rc1 Released.
+
+2011-09-06 13:48 +0000 [r334453] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * apps/app_voicemail.c: Make SQL query in app_voicemail.c portable
+ LIMIT is not portable. Regression from r312212 (closes issue
+ ASTERISK-18255) Reported by: Leif Madsen Tested by: Leif Madsen
+ Review: https://reviewboard.asterisk.org/r/1415/
+
+2011-09-02 20:59 +0000 [r334296-334355] Richard Mudgett <rmudgett@digium.com>
+
+ * res/res_musiconhold.c: MusicOnHold has extra unref which may lead
+ to memory corruption and crash. The problem happens when a call
+ is disconnected and you had started a MOH class that does not use
+ the files mode. If you define REF_DEBUG and recreate the problem,
+ it will announce itself with the following warning: Attempt to
+ unref mohclass 0xb70722e0 (default) when only 1 ref remained, and
+ class is still in a container! * Fixed moh_alloc() and
+ moh_release() functions not handling the state->class reference
+ consistently. (closes issue ASTERISK-18346) Reported by: Mark
+ Murawski Patches: jira_asterisk_18346_v1.8.patch (license #5621)
+ patch uploaded by rmudgett Tested by: rmudgett, Mark Murawski
+ Review: https://reviewboard.asterisk.org/r/1404/
+
+ * main/config.c, include/asterisk/config.h: Fix potential memory
+ allocation failure crashes in config.c. * Added required checks
+ to the returned memory allocation pointers to prevent crashes. *
+ Made ast_include_rename() create a replacement ast_variable list
+ node if the new filename is longer than the available space.
+ Fixes potential crash and memory leak. * Factored out
+ ast_variable_move() from ast_variable_update() so
+ ast_include_rename() can also use it when creating a replacement
+ ast_variable list node. * Made the filename stuffed at the end of
+ the struct a minimum allocated size in ast_variable_new() in case
+ ast_include_rename() changes the stored filename. * Constify
+ struct char pointers pointing to strings stuffed at the end of
+ the struct for: ast_variable, cache_file_mtime, and
+ ast_config_map. * Factored out cfmtime_new() to remove inlined
+ code and allow some struct pointers to become const. * Removed
+ the list lock from struct cache_file_mtime that was never used. *
+ Added doxygen comments to several structure elements and better
+ documented what strings are stuffed at the struct end char array.
+ * Reworked ast_config_text_file_save() and set_fn() to handle
+ allocation failure of the include file scratch pad object
+ tracking blank lines. * Made ast_config_text_file_save() fn[]
+ declared with PATH_MAX to ensure it is long enough for any
+ filename with path. Also reduced the number of container fileset
+ buckets from a rediculus 180,000 to 1023. JIRA AST-618 Review:
+ https://reviewboard.asterisk.org/r/1378/
+
+2011-09-01 17:38 +0000 [r334229-334234] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/pbx.c: Remove 1.6 compatibility documentation from 1.8, as
+ it no longer applies.
+
+ * res/res_config_odbc.c: Create a local alias for
+ ast_odbc_clear_cache. As a function pointer, the reference has to
+ be resolved at load time irrespective of the RTLD_LAZY flag.
+ Creating a local alias solves this problem, because the structure
+ is initialized with that local function pointer, while the actual
+ function can remain lazily linked until runtime. The reason why
+ this is important is because we lazily load function references
+ during the module loading process, in order to obtain priority
+ values for each module, ensuring that modules are loaded in the
+ correct order. Previous to this change, when this module was
+ initially loaded, the module loader would emit a symbol
+ resolution error, because of the above requirement. Closes
+ ASTERISK-18399 (reported by Mikael Carlsson, fix suggested by
+ Walter Doekes, patch by me)
+
+2011-08-31 18:50 +0000 [r334156] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Disable T.38 when we get a invite with image
+ media port set to 0 ASTERISK-17678
+
+2011-08-31 15:57 +0000 [r334009-334012] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: No DAHDI channel available for conference,
+ user introduction disabled. The following error will consistently
+ occur when trying to dial into a MeetMe conference when the
+ server does not have DAHDI hardware installed: app_meetme.c: No
+ DAHDI channel available for conference, user introduction
+ disabled (is chan_dahdi loaded?) While chan_dahdi is loaded
+ correctly during compilation and install of Asterisk/Dahdi,
+ including associated modules, etc., a chan_dahdi.conf
+ configuration file in /etc/asterisk is not created by FreePBX if
+ hardware does not exist, causing MeetMe to be unable to open a
+ DAHDI pseudo channel. * Allow chan_dahdi to create a pseudo
+ channel when there is no chan_dahdi.conf file to load. (closes
+ issue ASTERISK-17398) Reported by: Preston Edwards Patches:
+ jira_asterisk_17398_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: rmudgett
+
+ * main/channel.c, channels/chan_agent.c: Call pickup race leaves
+ orphaned channels or crashes. Multiple users attempting to pickup
+ a call that has been forked to multiple extensions either crashes
+ or fails a masquerade with a "bad things may happen" message.
+ This is the scenario that is causing all the grief: 1) Pickup
+ target is selected 2) target is marked as being picked up in
+ ast_do_pickup() 3) target is unlocked by ast_do_pickup() 4) app
+ dial or queue gets a chance to hang up losing calls and calls
+ ast_hangup() on target 5) SINCE A MASQUERADE HAS NOT BEEN SETUP
+ YET BY ast_do_pickup() with ast_channel_masquerade(),
+ ast_hangup() completes successfully and the channel is no longer
+ in the channels container. 6) ast_do_pickup() then calls
+ ast_channel_masquerade() to schedule the masquerade on the dead
+ channel. 7) ast_do_pickup() then calls ast_do_masquerade() on the
+ dead channel 8) bad things happen while doing the masquerade and
+ in the process ast_do_masquerade() puts the dead channel back
+ into the channels container 9) The "orphaned" channel is visible
+ in the channels list if a crash does not happen. This patch does
+ the following: * Made ast_hangup() set AST_FLAG_ZOMBIE on a
+ successfully hung-up channel and not release the channel lock
+ until that has happened. * Made __ast_channel_masquerade() not
+ setup a masquerade if either channel has AST_FLAG_ZOMBIE set. *
+ Fix chan_agent misuse of AST_FLAG_ZOMBIE since it would no longer
+ work. (closes issue ASTERISK-18222) Reported by: Alec Davis
+ Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer (closes
+ issue ASTERISK-18273) Reported by: Karsten Wemheuer Tested by:
+ rmudgett, Alec Davis, irroot, Karsten Wemheuer Review:
+ https://reviewboard.asterisk.org/r/1400/
+
+2011-08-31 15:18 +0000 [r334006] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c: Correct an AMI protocol violation with
+ SIPshowpeer The response of SIPshowpeer ends with "\r\n\r\n".
+ Since other commands are ended by using \r\n this confuses any
+ interfacing script. (closes issue ASTERISK-17486)
+
+2011-08-30 21:16 +0000 [r333947] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooq931.c,
+ addons/ooh323c/src/ooCalls.c, addons/ooh323c/src/ooh323.c,
+ addons/ooh323c/src/ooGkClient.c, addons/ooh323c/src/ooCalls.h:
+ cleanups in ACF/ARJ GK replies processing fixed long (24 sec)
+ pause if acf/arj proccessed before ast_cond_wait called to wait
+ this
+
+2011-08-29 21:38 +0000 [r333836] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Refresh peer address if DNS unavailable at
+ peer creation If Asterisk starts and no DNS is available,
+ outbound registrations will fail indefinitely. This patch copies
+ the address from the sip_registry struct, which will be updated,
+ to the peer->addr when necessary. If dnsmgr is enabled, the
+ registration fails without the patch because even though the
+ address on the registry is updated via dnsmgr, the address is
+ just copied on the first try. Since we use ast_sockaddr_copy,
+ dnsmgr can't update the address that is copied to the sip_pvt or
+ peers. Closes issue ASTERISK-18000 Review:
+ https://reviewboard.asterisk.org/r/1335/
+
+2011-08-29 21:06 +0000 [r333784-333785] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/channel.h: Add some do not hold locks notes to
+ channel.h
+
+ * addons/chan_mobile.c: Fix deadlock potential of
+ chan_mobile.c:mbl_ast_hangup().
+
+2011-08-29 17:11 +0000 [r333630] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_voicemail.c: Fixed improperly formatted TestEvent AMI
+ message in app_voicemail
+
+2011-08-29 15:55 +0000 [r333569] Jonathan Rose <jrose@digium.com>
+
+ * res/res_jabber.c: Accidental use of variable client->status
+ instead of client->state in from ASTERISK-18078 (issue
+ ASTERISK-18078)
+
+2011-08-28 09:49 +0000 [r333507] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_vpb.cc: chan_vpb: remove unused variables (gcc4.6)
+ GCC 4.6 detects variables that get assined to, but never used
+ later. Also removes some remmed-out lines that become invalid.
+ (closes issue ASTERISK-18336) Signed-off-by: Tzafrir Cohen
+ (License #5035) <tzafrir.cohen@xorcom.com>,
+
+2011-08-26 16:19 +0000 [r333378] Jonathan Rose <jrose@digium.com>
+
+ * res/res_jabber.c: [patch] Buddies are always auto-registered when
+ processing the roster Reporter said autoregister flag was ignored
+ for registering 'buddies' which had a subscription to us.
+ Verified that this was the case and observed how the patch
+ addressed this and made sure it didn't break anything. (closes
+ issue ASTERISK-14233) Reported by: Simon Arlott Patches:
+ asterisk-0015229.patch (license #5756) patch uploaded by Simon
+ Arlott Tested by: Jonathan Rose
+
+2011-08-26 14:36 +0000 [r333339-333354] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_voicemail.c: Fixed incorrect pointer copy to structure
+ copy in revision 333339
+
+ * apps/app_voicemail.c: Bug fixes for voicemail user emailsubject /
+ emailbody. This code change fixes a few issues with the voicemail
+ user override of emailbody and emailsubject, including escaping
+ the strings, potential memory leaks, and not overriding the
+ voicemail defaults. Revision 325877 fixed this for
+ ASTERISK-16795, but did not fix it for ASTERISK-16781. A
+ subsequent check-in prevented 325877 from being applied to 10.
+ This check-in resolves both issues, and applies the changes to
+ 1.8, 10, and trunk. (closes issue ASTERISK-16781) Reported by:
+ Sebastien Couture Tested by: mjordan (closes issue
+ ASTERISK-16795) Reported by: mdeneen Tested by: mjordan Review:
+ https://reviewboard.asterisk.org/r/1374
+
+2011-08-25 19:00 +0000 [r333267] Jason Parker <jparker@digium.com>
+
+ * Makefile: Fix for DESTDIR spaces patch.
+
+2011-08-25 18:47 +0000 [r333265] Jonathan Rose <jrose@digium.com>
+
+ * res/res_jabber.c: Segfault when publishing device states via XMPP
+ and not connected When using publishing device state with
+ res_jabber, Asterisk will attempt to send a device state using
+ the unconnected client using iks_send_raw and crash. This patch
+ checks the validity of the connection before attempting to send
+ the device state. (closes issue ASTERISK-18078) Reported by:
+ Michael L. Young Patches:
+ res_jabber-segfault-pubsub-not-connected2.patch (license #5026)
+ patch uploaded by Michael L. Young Tested by: Jonathan Rose
+
+2011-08-25 15:27 +0000 [r333201] Jason Parker <jparker@digium.com>
+
+ * makeopts.in, sounds/Makefile, Makefile, build_tools/mkpkgconfig,
+ configure, configure.ac: Fix installation into directories
+ containing spaces. This also vastly simplifies the logic in
+ sounds/Makefile (Closes issue ASTERISK-18290) Reported by: Paul
+ Belanger Review: https://reviewboard.asterisk.org/r/1379/
+
+2011-08-23 18:14 +0000 [r333010] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_queue.c: Memory Leak in app_queue The patch that was
+ committed in the 1.6.x versions of Asterisk for ASTERISK-15862
+ actually fixed two issues. One was not applicable to 1.8 but the
+ other is. queue_leak.patch fixes the portion applicable to 1.8.
+ (closes issue ASTERISK-18265) Reported by: Fred Schroeder
+ Patches: queue_leak.patch (license #5049) patch uploaded by
+ mmichelson Tested by: Thomas Arimont
+
+2011-08-23 18:11 +0000 [r333009] Matthew Nicholson <mnicholson@digium.com>
+
+ * UPGRADE.txt, configs/sip.conf.sample, CHANGES,
+ channels/sip/include/sip.h: default 'sipstorecause' to no We've
+ decided to disable this feature by default in future 1.8
+ versions. This would be an unexpected behavior change for anyone
+ depending on that SIP_CAUSE update in their dialplan. Please
+ refer to the asterisk-dev mailing list more information:
+ http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html
+ (issue AST-580)
+
+2011-08-22 22:11 +0000 [r332939-332945] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_queue.c, main/config.c, include/asterisk/config.h:
+ Revert previous commit. Not ready yet.
+
+ * apps/app_queue.c, main/config.c, include/asterisk/config.h:
+ Signed
+
+ * main/config.c: Minor code optimizations. * Simplify
+ ast_category_browse() logic for easier understanding. * Remove
+ dead code in ast_variable_delete() and simplify some of its
+ logic.
+
+2011-08-22 19:41 +0000 [r332876] Paul Belanger <pabelanger@digium.com>
+
+ * channels/chan_gtalk.c: Revert previous commit It seems google is
+ still making changes to the protocol. (issue ASTERISK-18301)
+
+2011-08-22 19:32 +0000 [r332874] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_queue.c: Reference leaks in app_queue. * Fixed
+ load_realtime_queue() leaking a queue reference when it
+ overwrites q when processing a realtime queue. (issue
+ ASTERISK-18265) * Make join_queue() unreference the queue
+ returned by load_realtime_queue() when it is done with the
+ pointer. The load_realtime_queue() returns a reference to the
+ just loaded realtime queue. * Fixed queues container reference
+ leak in queues_data_provider_get(). * queue_unref() should not
+ return q that was just unreferenced. * Made logic in
+ __queues_show() and queues_data_provider_get() when calling
+ load_realtime_queue() easier to understand.
+
+2011-08-22 18:15 +0000 [r332817] Matthew Jordan <mjordan@digium.com>
+
+ * main/app.c, configs/manager.conf.sample,
+ include/asterisk/manager.h, apps/app_voicemail.c,
+ include/asterisk/test.h, main/manager.c, main/file.c,
+ main/test.c: Review: https://reviewboard.asterisk.org/r/1364/
+ This update adds a new AMI event, TestEvent, which is enabled
+ when the TEST_FRAMEWORK compiler flag is defined. It also adds
+ initial usage of this event to app_voicemail. The TestEvent AMI
+ event is used extensively by the voicemail tests in the Asterisk
+ Test Suite.
+
+2011-08-22 18:14 +0000 [r332759-332816] Richard Mudgett <rmudgett@digium.com>
+
+ * res/res_config_pgsql.c, res/res_config_odbc.c: Memory leaks in
+ realtime_multi_xxx() when database access returns error. * Fix
+ realtime_multi_pgsql() configuration memory leak when the
+ database access returns an error. * Fix realtime_multi_odbc()
+ configuration category use after free when the database access
+ returns an error.
+
+ * main/config.c: Memory leak reading realtime database variable
+ list. Calling ast_load_realtime() can leak the last list node if
+ the read list only contains empty variable value items. * Fixed
+ list filter loop in ast_load_realtime() to delete the list node
+ immediately instead of the next time through the loop. The next
+ time through the loop may not happen if the node to delete is the
+ last in the list. (issue ASTERISK-18277) (issue ASTERISK-18265)
+ Patches: jira_asterisk_18265_v1.8_config.patch (license #5621)
+ patch uploaded by rmudgett
+
+2011-08-21 14:31 +0000 [r332699] Paul Belanger <pabelanger@digium.com>
+
+ * channels/chan_gtalk.c: Fix outgoing calls in chan_gtalk (closes
+ issue ASTERISK-18301) Reported by: az1324
+
+2011-08-18 21:26 +0000 [r332559] Terry Wilson <twilson@digium.com>
+
+ * main/netsock2.c: Fix possible error on stringification of
+ IPv4-mapped addrs The FreeBSD netsock2 test has been failing for
+ a while. We were pasing sa->len to getnameinfo instead of
+ sa_tmp->len. ASTERISK-18289
+
+2011-08-18 19:28 +0000 [r332503] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_dahdi.c: CRC4 in "dahdi show status" gives wrong
+ impression to T1 users Change CRC4 to CRC in the output of "dahdi
+ show status" so that it can apply in more situations without
+ confusing users, especially since T1 lines use CRC6 instead of
+ CRC4. (closes issue AST-471)
+
+2011-08-18 14:46 +0000 [r332355-332446] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * build_tools/cflags.xml, build_tools/cflags-devmode.xml: Move
+ BETTER_BACKTRACES out of development mode, as it's useful when
+ DEBUG_THREADS is enabled.
+
+ * makeopts.in, sounds/Makefile, Makefile, agi/Makefile,
+ utils/Makefile, configure, include/asterisk/autoconfig.h.in,
+ configure.ac, Makefile.moddir_rules: Re-add support for spaces in
+ pathnames, including now spaces in DESTDIR. This was initially
+ added to 1.8 prior to release, primarily to support the standard
+ paths on Mac OS X, but was partially reverted recently in
+ Subversion, due to the lack of support for spaces in DESTDIR.
+ This commit restores support for the standard paths on Mac OS X,
+ and also includes support for spaces in DESTDIR. (closes issue
+ ASTERISK-18290) Reported by: pabelanger Review:
+ https://reviewboard.asterisk.org/r/1326/
+
+2011-08-17 17:35 +0000 [r332320] Terry Wilson <twilson@digium.com>
+
+ * res/res_timing_timerfd.c: Don't read from a disarmed or invalid
+ timerfd Numerous isues have been reported for deadlocks that are
+ caused by a blocking read in res_timing_timerfd on a file
+ descriptor that will never be written to. This patch adds some
+ checks to make sure that the timerfd is both valid and armed
+ before calling read(). Should fix: ASTERISK-1842, ASTERISK-18197,
+ ASTERISK-18166, AST-486 AST-495, AST-507 and possibly others.
+
+2011-08-17 15:51 +0000 [r332264] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Outgoing BRI
+ calls fail when using Asterisk 1.8 with HA8, HB8, and B410P
+ cards. France Telecom brings layer 2 and layer 1 down on BRI
+ lines when the line is idle. When layer 1 goes down Asterisk
+ cannot make outgoing calls and the HA8 and HB8 cards also get IRQ
+ misses. The inability to make outgoing calls is because the line
+ is in red alarm and Asterisk will not make calls over a line it
+ considers unavailable. The IRQ misses for the HA8 and HB8 card
+ are because the hardware is switching clock sources from the line
+ which just brought layer 1 down to internal timing. There is a
+ DAHDI option for the B410P card to not tell Asterisk that layer 1
+ went down so Asterisk will allow outgoing calls: "modprobe
+ wcb4xxp teignored=1". There is a similar DAHDI option for the HA8
+ and HB8 cards: "modprobe wctdm24xxp bri_teignored=1".
+ Unfortunately that will not clear up the IRQ misses when the
+ telco brings layer 1 down. * Add layer 2 persistence option to
+ customize the layer 2 behavior on BRI PTMP lines. The new option
+ has three settings: 1) Use libpri default layer 2 setting. 2)
+ Keep layer 2 up. Bring layer 2 back up when the peer brings it
+ down. 3) Leave layer 2 down when the peer brings it down. Layer 2
+ will be brought up as needed for outgoing calls. JIRA AST-598
+
+2011-08-17 14:31 +0000 [r332234] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: print a warning instructing the user to
+ disable storesipcause if we process 100 or more scheduler entries
+ at a time AST-580
+
+2011-08-16 20:10 +0000 [r332176] Paul Belanger <pabelanger@digium.com>
+
+ * tests/test_db.c, tests/test_linkedlists.c, tests/test_sched.c,
+ tests/test_netsock2.c, tests/test_strings.c, tests/test_pbx.c,
+ tests/test_func_file.c, tests/test_security_events.c,
+ tests/test_stringfields.c, tests/test_time.c, tests/test_skel.c,
+ tests/test_locale.c, tests/test_acl.c, tests/test_devicestate.c,
+ tests/test_utils.c, tests/test_aoc.c, tests/test_astobj2.c,
+ tests/test_poll.c, tests/test_amihooks.c,
+ tests/test_substitution.c, tests/test_heap.c,
+ tests/test_ast_format_str_reduce.c, tests/test_expr.c,
+ tests/test_logger.c, tests/test_gosub.c, tests/test_app.c,
+ tests/test_dlinklists.c, tests/test_event.c: Flag test modules as
+ 'core' Review: https://reviewboard.asterisk.org/r/1369/
+
+2011-08-16 17:38 +0000 [r332118] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c: ASTERISK-18067 ASTERISK-15479 - White Space
+ affects mailbox value, multiple MWI subs Before, having multiple
+ subscriptions to mailboxes on a sip peer set via the mailbox
+ setting in sip.conf would only result in updates being sent on
+ whichever mailbox triggered the mwi event. Now all of them get
+ counted regardless. Also fixes a bug involving parsing of the
+ mailbox option in sip.conf so that trailing and leading spaces
+ before/after commas are trimmed. (closes issue ASTERISK-18067)
+ Reported by: aragon (closes issue ASTERISK-15479) Reported by:
+ Ben Winslow Patches:
+ chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288)
+ patch uploaded by Ben Winslow
+
+2011-08-16 16:31 +0000 [r332100] Richard Mudgett <rmudgett@digium.com>
+
+ * CHANGES, configs/features.conf.sample, main/asterisk.c,
+ main/features.c: Fix multiple parking issues. JIRA ASTERISK-17183
+ Multi-parkinglot directs calls to wrong parkinglot. JIRA
+ ASTERISK-17870 Cannot retrieve parked calls. JIRA ASTERISK-17430
+ ParkedCall() with no extension should pickup first available call
+ and does not. JIRA AST-576 Issues with parking lots * Removed
+ searching for parking lots by extension. Parking lots can only be
+ found by the parking lot name since parking lot access extensions
+ and spaces are not guaranteed to be unique. * Added
+ parking_lot_name option to the Park and ParkedCall applications.
+ Updated documentation for Park and ParkedCall applications. * Add
+ parkext_exclusive configuration option to make parking entry
+ extensions specify which parking lot they access. (closes issue
+ ASTERISK-17183) Reported by: David Cabrejos Tested by: rmudgett,
+ David Cabrejos (closes issue ASTERISK-17870) Reported by: Remi
+ Quezada (closes issue ASTERISK-17430) Reported by: Philippe
+ Lindheimer JIRA ASTERISK-17452 Parking_offset not used JIRA
+ AST-624 'next' setting for findslot does nothing * Reimplemented
+ since findslot feature option broken by -r114655. (closes issue
+ ASTERISK-17452) Reported by: David Woolley Tested by: rmudgett
+ JIRA ASTERISK-15792 Dialplan continues execution after transfer
+ to park. This happens for DTMF attended transfer, DTMF blind
+ transfer, and DTMF one-touch-parking if the party initiating
+ these features also initiated the call. * Fixed the return code
+ from the affected builtin features when parking a call. (closes
+ issue ASTERISK-15792) Reported by: Mat Murdock Tested by:
+ rmudgett, twilson JIRA AST-607 The courtesytone is not playing to
+ the expected call when picking up a parked call. This is mostly a
+ documentation problem. However, the option is not reset to the
+ default when features.conf is reloaded. * Updated
+ features.conf.sample documentation for courtesytone and
+ parkedplay options. * Reset the parkedplay option to default when
+ features.conf is reloaded. JIRA AST-615 AMI Park action followed
+ by features reload results in orphaned channels in parking lot. *
+ Reloading features.conf will not touch parking lots that have
+ calls still parked in them. Reload again at a later time. Misc
+ additional fixes: * Added unit test for parking lot dialplan
+ usage checking. * Made update connected line when a parked call
+ is retrieved from a parking lot. * Made retrieved parked call
+ stop ringing or MOH depending upon how the call was waiting in
+ the parking lot. * Made CLI "features show" indicate if the
+ parking lot is enabled for use. * Added PARKINGDYNEXTEN channel
+ variable to allow dynamic parking lots to specify the parking lot
+ access extension. * Made AMI ParkedCalls action ParkedCall events
+ have a Parkinglot header. * Made AMI ParkedCalls action
+ ParkedCallsComplete event have a Total header. * Fixed potential
+ deadlock from AMI Park action holding channel locks while calling
+ masq_park_call(). * Fixed several places where ast_strdupa() were
+ used inside of loops. (Mostly fixed by refactoring the loop body
+ into its own function.) * Fixed copy_parkinglot() copying too
+ much from the source parking lot. Extracted the parking lot
+ configuration settings into struct parkinglot_cfg. * Refactored
+ courtesytone playing code to put the channel not playing the tone
+ in autoservice. * Fix when pbx-parkingfailed is played that the
+ other channel is put in autoservice if it exists. * Fixed
+ parkinglot reference leak in parked_call_exec() error paths. *
+ Fixed parkinglot_unref() use of parkinglot after it was unreffed.
+ * Made destroy the struct ast_parkinglot parkings lock when done.
+ * Refactored the features.conf parking lot configuration code to
+ eliminate redundancy. * Fixed feature reload to better protect
+ parking lots. * Fixed parking lot container reference leak in
+ handle_parkedcalls(). * Fixed the total count in
+ handle_parkedcalls(). Review:
+ https://reviewboard.asterisk.org/r/1358/
+
+2011-08-16 15:06 +0000 [r332021-332026] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: use
+ DEFAULT_STORE_SIP_CAUSE to set the default value for the
+ 'storesipcause' option AST-580
+
+ * configs/sip.conf.sample, CHANGES, channels/chan_sip.c: Added the
+ 'storesipcause' option to sip.conf to allow the user to disable
+ the setting of HASH(SIP_CAUSE,<chan name>) on the channel. Having
+ chan_sip set HASH(SIP_CAUSE,<chan name>) on the channel carries a
+ significant performance penalty because of the usage of the
+ MASTER_CHANNEL() dialplan function. AST-580
+
+2011-08-15 17:24 +0000 [r331955] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Fix some minor chan_dahdi config load
+ issues. * Address chan_dahdi.conf dahdichan option todo item
+ about needing line number. * Make ignore_failed_channels option
+ also apply to dahdichan option. * Don't attempt to create a
+ default pseudo channel if the chan_dahdi.conf channel/channels
+ option is not allowed. * Add a similar check for dahdichan in
+ normal chan_dahdi.conf sections as is done in users.conf.
+
+2011-08-15 15:21 +0000 [r331886] Paul Belanger <pabelanger@digium.com>
+
+ * main/rtp_engine.c: Fix noisy message when briding channels
+ (closes issue ASTERISK-18270) Reported by: Federico Alves
+
+2011-08-15 15:12 +0000 [r331867] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Fixes locking inversion issues present in
+ the handling of the sip REFER method. (closes issue
+ ASTERISK-18082) Reported by: James Van Vleet
+
+2011-08-12 19:01 +0000 [r331774] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_queue.c: Unlock the channel before calling update_queue.
+ Holding the channel lock when calling update_queue which attempts
+ to lock the queue lock can cause a deadlock. This deadlock
+ involves the following chain: 1. hold chan lock -> wait queue
+ lock 2. hold queue lock -> wait agent list lock 3. hold agent
+ list lock -> wait chan list lock 4. hold chan list lock -> wait
+ chan lock
+
+2011-08-12 18:58 +0000 [r331714-331771] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Suppress warning message when using
+ DAHDITransfer or DAHDIHangup. * The fake event should only be
+ processed by the channel that currently owns the private and not
+ the associated call waiting or 3-way channel. JIRA AST-620 JIRA
+ SWP-3616
+
+ * channels/chan_dahdi.c: AMI actions DAHDIHangup and DAHDITransfer
+ have no effect. The AMI actions DAHDIHangup and DAHDITransfer
+ have no effect on a DAHDI channel. These two AMI actions are
+ highly specialized to analog channels and appear to make the
+ channel behave like a jack port for headsets. * Made the faked
+ DAHDI event get processed before a normal media stream read in
+ dahdi_read() instead of trying to trigger an exception read by
+ setting the AST_FLAG_EXCEPTION flag. Apparently a change was made
+ long ago that changed how AST_FLAG_EXCEPTION is processed in the
+ core. Unfortunately, the faked DAHDI events no longer worked when
+ that happened. * Updated the DAHDI AMI action documentation for
+ the following actions: DAHDITransfer, DAHDIHangup,
+ DAHDIDialOffhook, DAHDIDNDon, DAHDIDNDoff, DAHDIShowChannels, and
+ DAHDIRestart. * Made use sscanf() instead of atoi() for better
+ error checking of the DAHDIChannel header string. JIRA AST-620
+ JIRA SWP-3616
+
+2011-08-12 16:30 +0000 [r331658] Terry Wilson <twilson@digium.com>
+
+ * tests/test_netsock2.c: Fix netsock2 multiple zero-expansion test
+ Remove erroneous single bracket.
+
+2011-08-12 16:20 +0000 [r331649] Kinsey Moore <kmoore@digium.com>
+
+ * main/logger.c: Logger does not warn of failure to open logging
+ channels Currently, logger only prints an error message to stderr
+ when it fails to open a logger channel where many users will not
+ see it because the logger lock is held. The alternative provided
+ by this patch is to log the error to all attached consoles in the
+ hopes that it will be easier to see. Additionally, this patch
+ prevents the failed logger channel from being added to the list
+ where it would silently fail on each call to the Asterisk logger.
+ (closes issue ASTERISK-16231) Review:
+ https://reviewboard.asterisk.org/r/1338
+
+2011-08-12 15:49 +0000 [r331635] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_dial.c, apps/app_meetme.c: Fixes 32bit compilation
+ warnings brought on by 331634 in app_dial and app_meetme
+
+2011-08-11 21:46 +0000 [r331578] Jason Parker <jparker@digium.com>
+
+ * apps/app_dial.c, apps/app_meetme.c: Use proper values for 64-bit
+ option flags. Also, reusing bits es no bueno, so change the value
+ of a duplicate. (issue ASTERISK-18239)
+
+2011-08-11 21:39 +0000 [r331575] Richard Mudgett <rmudgett@digium.com>
+
+ * funcs/func_shell.c: Segfault in shell_helper in func_shell.c. The
+ return value of popen() was not checked for failure to open.
+ (closes issue ASTERISK-18109) JIRA SWP-3633 Reported by: Michael
+ Myles Tested by: rmudgett
+
+2011-08-10 22:23 +0000 [r331517] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c: SIP Notify via AMI or CLI leaks SIP PVTs Any
+ SIP notify sent via AMI or CLI leaks a SIP PVT with ref count +2.
+ Removing the additional ref just before the invite and adding an
+ unref following it corrects the issue as seen via REF_DEBUG. The
+ unref existed in a distant revision and it appears as though the
+ wrong ref operation was removed. (closes issue ASTERISK-18091)
+ Review: https://reviewboard.asterisk.org/r/1332/
+
+2011-08-10 20:29 +0000 [r331461] Richard Mudgett <rmudgett@digium.com>
+
+ * main/logger.c: Output of queue log not started until logger
+ reloaded. ASTERISK-15863 caused a regression with queue logging.
+ The output of the queue log is not started until the logger
+ configuration is reloaded. * Queue log initialization is
+ completely delayed until the first message is posted to the queue
+ log system. Including the initial opening of the queue log file.
+ * Fixed rotate_file() ROTATE strategy to give the file just
+ rotated out to the configured exec function after rotate. Just
+ like the other strategies. * Fixed logger reload to always post
+ the queue reload entry instead of just if there is a queue log
+ file. * Refactored some code to eliminate some redundancy and to
+ reduce stack utilization. (closes issue ASTERISK-17036) JIRA
+ SWP-2952 Reported by: Juan Carlos Valero Patches:
+ jira_asterisk_17036_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: rmudgett (closes issue ASTERISK-18208)
+ Reported by: Christian Pinedo Review:
+ https://reviewboard.asterisk.org/r/1333/
+
+2011-08-31 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.6.0 Released.
+
+2011-08-25 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.6.0-rc3 Released.
+
+ ------------------------------------------------------------------------
+ r333201 | qwell | 2011-08-25 10:27:06 -0500 (Thu, 25 Aug 2011) | 8 lines
+
+ Fix installation into directories containing spaces.
+
+ This also vastly simplifies the logic in sounds/Makefile
+
+ (Closes issue ASTERISK-18290)
+ Reported by: Paul Belanger
+ Review: https://reviewboard.asterisk.org/r/1379/
+ ------------------------------------------------------------------------
+
+2011-08-22 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.6.0-rc2 Released.
+
+ ------------------------------------------------------------------------
+ r331575 | rmudgett | 2011-08-11 16:39:58 -0500 (Thu, 11 Aug 2011) | 9 lines
+
+ Segfault in shell_helper in func_shell.c.
+
+ The return value of popen() was not checked for failure to open.
+
+ (closes issue ASTERISK-18109)
+ JIRA SWP-3633
+ Reported by: Michael Myles
+ Tested by: rmudgett
+ ------------------------------------------------------------------------
+ r332355 | tilghman | 2011-08-17 14:21:36 -0500 (Wed, 17 Aug 2011) | 13 lines
+
+ Re-add support for spaces in pathnames, including now spaces in DESTDIR.
+
+ This was initially added to 1.8 prior to release, primarily to support the
+ standard paths on Mac OS X, but was partially reverted recently in Subversion,
+ due to the lack of support for spaces in DESTDIR. This commit restores support
+ for the standard paths on Mac OS X, and also includes support for spaces in
+ DESTDIR.
+
+ (closes issue ASTERISK-18290)
+ Reported by: pabelanger
+
+ Review: https://reviewboard.asterisk.org/r/1326/
+ ------------------------------------------------------------------------
+ r332559 | twilson | 2011-08-18 16:26:01 -0500 (Thu, 18 Aug 2011) | 7 lines
+
+ Fix possible error on stringification of IPv4-mapped addrs
+
+ The FreeBSD netsock2 test has been failing for a while. We were
+ pasing sa->len to getnameinfo instead of sa_tmp->len.
+
+ ASTERISK-18289
+ ------------------------------------------------------------------------
+
+2011-08-10 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 1.8.6.0-rc1 Released.
+
+2011-08-10 13:47 +0000 [r331315] Kinsey Moore <kmoore@digium.com>
+
+ * main/manager.c: AMI action ModuleReload returns Error if Module:
+ missing or empty An empty string was not being checked for
+ properly causing identification of the module to be reloaded to
+ fail and return an Error with message "No such module." (closes
+ issue AST-616)
+
+2011-08-09 22:12 +0000 [r331248] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_iax2.c, apps/app_parkandannounce.c, main/pbx.c,
+ channels/chan_sip.c, main/features.c: Misc minor items found in
+ code. * Add some reentrancy protection in pbx.c when creating the
+ contexts_table hash table. * Fix inverted test in chan_sip.c
+ conditional code. * Fix uninitialized variable and use of the
+ wrong variable in chan_iax2.c. * Fix test of return value in
+ app_parkandannounce.c. Explicitly testing for -1 is bad if the
+ function does not actually return that value when it fails. *
+ Fixup some comments and add some curly braces in features.c.
+
+2011-08-09 16:13 +0000 [r331146] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooGkClient.c,
+ addons/chan_ooh323.c: move ast_cond_signal for admitted call
+ after all data filled/freed clear all log channels by pointed
+ number not only first free allocated callToken in ooh323_answer
+
+2011-08-09 15:58 +0000 [r331142] Jason Parker <jparker@digium.com>
+
+ * doc/asterisk.8: Regenerate asterisk man page from sgml.
+
+2011-08-08 20:52 +0000 [r331038] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_musiconhold.c: In-queue MOH stops after a periodic
+ announcement If the seek value is past the end of file when
+ resuming G.722 MOH, MOH will cease to function for the duration
+ of the MOH session through all starts and stops until saved state
+ is cleared. Adjusting the code to guarantee a single valid read
+ (which is already assumed) fixes the bug. (closes issue
+ ASTERISK-18077) Review: https://reviewboard.asterisk.org/r/1328/
+ Tested-by: Jonathan Rose <jrose@digium.com>
+
+2011-08-04 20:29 +0000 [r330843] Terry Wilson <twilson@digium.com>
+
+ * configure, configure.ac: Make libsrtp instructions more explicit
+ when linking fails (closes issue ASTERISK-18139)
+
+2011-08-04 19:37 +0000 [r330827] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooCmdChannel.c,
+ addons/ooh323c/src/ooGkClient.c: change gk client behaivour on
+ rrq/grq failures to setup timers and next tries after timeout
+ instead of complete failure in the ooh323 stack
+
+2011-08-03 15:14 +0000 [r330705-330762] Kinsey Moore <kmoore@digium.com>
+
+ * main/Makefile: editing files in main/editline does not ensure
+ rebuild of libedit.a When editing a source file in main/editline,
+ the build system does not rebuild libedit.a and uses the already
+ existing one instead. Adding a PHONY to CHECK_SUBDIR fixes this
+ problem. (closes issue ASTERISK-16221) Patch-by: Walter Doekes
+
+ * channels/chan_dahdi.c, channels/sig_analog.c: Call pickup broken
+ for DAHDI channels when beginning with # The call pickup feature
+ did not work on DAHDI devices for anything other than feature
+ codes beginning with * since all feature codes in chan_dahdi were
+ originally hard-coded to begin with *. This patch is also applied
+ to chan_dahdi.c to fix this bug with radio modes. (closes issue
+ AST-621) Review: https://reviewboard.asterisk.org/r/1336/
+
+2011-08-02 20:51 +0000 [r330648] Kevin P. Fleming <kpfleming@digium.com>
+
+ * res/res_jabber.c: Convert an error message to actually be
+ helpful.
+
+2011-08-02 16:15 +0000 [r330575-330581] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: Fixes crash in chan_iax2. Fixes crash in
+ chan_iax2 resulting from an edge case in the way control frames
+ are queued during calltoken negotiation is complete. (closes
+ issue ASTERISK-17610) Reported by: mgrobecker
+
+ * channels/chan_sip.c: Optimization to buffer initialization fix.
+
+ * channels/chan_sip.c: Fixes uninitialized string buffer in log
+ message. (closes issue ASTERISK-17200) Reported by: lmadsen
+
+2011-08-01 15:22 +0000 [r330433] Kinsey Moore <kmoore@digium.com>
+
+ * main/say.c: Incorrect playback for Spanish in some circumstances
+ When you say the time in spanish and it is 01:00 - 01:59 or 13:00
+ - 13:59 you must use female pronunciation "1F". The function
+ "say_date_with_format_es" does not take this in account. (closes
+ ASTERISK-15016) Patch-by: Luis Jimenez
+
+2011-07-30 23:56 +0000 [r330368] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Remove some redundant locking code in
+ ast_do_masquerade(). Also updated some comments.
+
+2011-07-30 15:25 +0000 [r330311] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * main/channel.c: prevent double masqurading channels when one is
+ been hung up and deadlock avoidance is used. There is a race
+ condition in ast_do_masquerade / ast_hangup (at least) Reported
+ by me signed off by schmidts with input from David Vossel Review:
+ https://reviewboard.asterisk.org/r/1323/
+
+2011-07-29 17:18 +0000 [r330203-330213] Sean Bright <sean@malleable.com>
+
+ * formats/format_wav.c: Correct the check for O_RDONLY.
+
+ * formats/format_wav.c: Only write to wav files that were opened to
+ be written to.
+
+2011-07-28 21:42 +0000 [r330107] Terry Wilson <twilson@digium.com>
+
+ * main/term.c: Make console colors work for TERM=xterm-256color
+
+2011-07-28 17:04 +0000 [r330050] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Merged revisions 330033 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r330033 | rmudgett | 2011-07-28 11:26:38 -0500 (Thu,
+ 28 Jul 2011) | 15 lines Datacalls with B410P fail. Incoming and
+ outgoing call legs of a data call are using different formats:
+ a-law, u-law. When the call is bridged, the media stream is run
+ through translation to convert the media formats. The translation
+ is bad for data calls. * Make incoming call that does not
+ explicitly specify u-law or a-law use the DAHDI channel's default
+ law. The outgoing call always uses the default law from the DAHDI
+ channel. (closes issue ABE-2800) Patches:
+ jira_abe_2800_companding.patch (license #5621) patch uploaded by
+ rmudgett ..........
+
+2011-07-28 15:45 +0000 [r329994] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c: Fix a SIP transfer deadlock. The locking in
+ this function is very scary. There are like 6 structs involved.
+ (closes issue AST-470)
+
+2011-07-28 15:26 +0000 [r329991] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: check for CONFIG_STATUS_FILE_INVALID when loading
+ the res_fax config file Patch by: tzafrir Reported by: tzafrir
+ (closes issue ASTERISK-18161)
+
+2011-07-28 11:34 +0000 [r329895] Sean Bright <sean@malleable.com>
+
+ * channels/chan_sip.c: Make the output of Externhost in 'sip show
+ settings' more consistent.
+
+2011-07-27 19:27 +0000 [r329782] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_confbridge.c: Change support for ConfBridge() in 1.8 to
+ Extended.
+
+2011-07-27 19:17 +0000 [r329767] Sean Bright <sean@malleable.com>
+
+ * Makefile.moddir_rules: Explicitly sort the module list so that
+ the menuselect lists are sorted. (closes issue ASTERISK-18141)
+ Reported by: Richard Miller Patches: sort-order.diff uploaded by
+ seanbright (License #5060) Tested by: leifmadsen
+
+2011-07-27 18:10 +0000 [r329709] Jonathan Rose <jrose@digium.com>
+
+ * configs/indications.conf.sample: Fix New Zealand indications
+ profile based on http://www.telepermit.co.nz/TNA102.pdf (closes
+ issue ASTERISK-16263) Reported by: richardf Patches:
+ nz-indications.patch uploaded by richardf (License #6015)
+
+2011-07-27 04:23 +0000 [r329613] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * cdr/cdr_odbc.c: Duration and billsec are swapped in high
+ resolution time. Closes ASTERISK-18024 Patches:
+ 20110726__ASTERISK-18024.diff by Tilghman Lesher (License 5003)
+
+2011-07-26 14:04 +0000 [r329527-329529] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_voicemail.c: Changes sound file for prepend
+ "then-press-pound" to "vm-then-pound" which is the same prompt,
+ only it turned out "then-press-pound" was part of extra sounds.
+ Also, vm is more appropriate anyway.
+
+ * main/app.c, apps/app_voicemail.c, include/asterisk/app.h,
+ configs/voicemail.conf.sample: Fixes some voicemail forwarding
+ behavior based around prepend mode. Formerly, prepend forwarding
+ would have the user record a message with no useful prompt and an
+ expectation for the user to push a button on the phone when
+ finished recording. If a length of silence was detected instead,
+ the recording would be canceled and the user would re-enter the
+ voicemail forwarding menu. Subsequent time-outs in prepend
+ recording would also bug out in the sense that they would write
+ over the original message and get sent to the recipient
+ regardless of whether they timed out or were accepted. This patch
+ fixes this issue and adds a prompt which will be played after a
+ timeout informing the user that they needed to press a button.
+ Currently, the sound files that we have are somewhat inadquate
+ for this, so after the call we simply have Allison say "Please
+ try again. Then press pound." which actually relies on two
+ separate sound files. Just one would be more appropriate.
+ reporter: Vlad Povorozniuc Review:
+ https://reviewboard.asterisk.org/r/1327/
+
+2011-07-25 19:49 +0000 [r329471] Paul Belanger <pabelanger@digium.com>
+
+ * main/enum.c: Decrease verbose messages to debug, to help clean up
+ CLI.
+
+2011-07-22 21:10 +0000 [r329144-329333] Richard Mudgett <rmudgett@digium.com>
+
+ * main/pbx.c: Fix memory leak in an allocation error path of
+ handle_statechange(). * Make use buffer accessor function in
+ handle_statechange() rather than directly accessing the struct
+ member. * Make use less redundant loop construct for iterating
+ over hints.
+
+ * main/pbx.c: Deadlocks dealing with dialplan hints during reload.
+ There are two remaining different deadlocks reported dealing with
+ dialplan hints. The deadlock in ASTERISK-17666 is caused by
+ invalid locking order in ast_remove_hint(). The hints container
+ must be locked before the hint object. The deadlock in
+ ASTERISK-17760 is caused by a catch-22 situation in
+ handle_statechange(). The deadlock is caused by not having the
+ conlock before calling the watcher callbacks. Unfortunately,
+ having that lock causes a different deadlock as reported in
+ ASTERISK-16961. * Fixed ast_remove_hint() locking order. * Made
+ handle_statechange() no longer call the watcher callbacks holding
+ any locks that matter. * Made hint ao2 destructor do the watcher
+ callbacks for extension deactivation to guarantee that they get
+ called. * Fixed hint reference leak in ast_add_hint() if the
+ callback container constructor failed. * Fixed hint reference
+ leak in complete_core_show_hint() for every hint it found for CLI
+ tab completion. * Adjusted locking in
+ ast_merge_contexts_and_delete() for safety. * Added
+ context_merge_lock to prevent ast_merge_contexts_and_delete() and
+ handle_statechange() from interfering with each other. * Fixed
+ ast_change_hint() not taking into account that the extension is
+ used for the hash key. (closes issue ASTERISK-17666) Reported by:
+ irroot Tested by: irroot JIRA SWP-3318 (closes issue
+ ASTERISK-17760) Reported by: Byron Clark Tested by: irroot JIRA
+ SWP-3393 Review: https://reviewboard.asterisk.org/r/1313/
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Document
+ parkinglot in chan_dahdi.conf.sample. * Document existing feature
+ in chan_dahdi.conf.sample. * Remove some dead code related to the
+ parkinglot option.
+
+ * apps/app_directed_pickup.c: Update PickupChan documentation. The
+ PickupChan uses the ampersand as the argument separator. Was
+ documented as: PickupChan(channel[,channel2[,...][,options]])
+ Fixed documentation to:
+ PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options])
+ This is a continuation of ASTERISK-17494 for v1.8 and later.
+ (closes issue ASTERISK-18144) Reported by: Erik Smith Patches:
+ pickupchan_ducumentation-v2.patch (License #6263) patch uploaded
+ by Erik Smith Tested by: Erik Smith
+
+ * main/features.c: Dialplan bridge() app mutex 'current_dest_chan'
+ freed more times than we've locked! This appears to be a leftover
+ from when ast_channel was converted to ao2 objects. Simply
+ removed the extraneous unlock. (closes issue ASTERISK-17772)
+
+2011-07-20 21:20 +0000 [r329027] Paul Belanger <pabelanger@digium.com>
+
+ * UPGRADE.txt: Asterisk now requires libpri 1.4.11+ for PRI
+ support.
+
+2011-07-20 20:52 +0000 [r329012] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c:
+ Backport useful CLI "pri show channels" command to v1.8. The "pri
+ show channels" command is useful for debuging to see if there are
+ any stuck B channels. .......... r307964 | rmudgett | 2011-02-15
+ 15:42:55 -0600 (Tue, 15 Feb 2011) | 9 lines Add CLI "pri show
+ channels" command. List the current mapping of DAHDI B channels
+ to Asterisk channel names and which calls are on hold or
+ call-waiting. Calls on hold or call-waiting are not associated
+ with any B channel. JIRA LIBPRI-27 JIRA SWP-2547 ..........
+ r308205 | rmudgett | 2011-02-17 14:21:56 -0600 (Thu, 17 Feb 2011)
+ | 1 line Add more verbage to CLI command 'pri show channels'
+ usage. .......... r312579 | rmudgett | 2011-04-04 11:17:58 -0500
+ (Mon, 04 Apr 2011) | 59 lines Change also updates 'pri show
+ channels' command with the "chan idle" column to report if a
+ channel is available for use.
+
+2011-07-20 20:16 +0000 [r328987] Terry Wilson <twilson@digium.com>
+
+ * tests/test_netsock2.c: We can't guarantee an eth0 is present
+ FreeBSD test fails on this case presumably because there is no
+ eth0 on the test machine. Better to just remove this test for
+ now.
+
+2011-07-20 19:00 +0000 [r328935] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c: Inband DTMF regression The functionality of
+ inband DTMF in chan_sip relied upon
+ ast_rtp_instance_dtmf_mode_get/set not working properly to avoid
+ calling ast_rtp_instance_dtmf_begin/end on RTP streams with
+ inband DTMF. According to documentation,
+ ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF,
+ never inband. This fixes the regression introduced in revision
+ 328823.
+
+2011-07-19 21:29 +0000 [r328878] Kevin P. Fleming <kpfleming@digium.com>
+
+ * sounds/Makefile, Makefile, Makefile.moddir_rules: Revert partial
+ attempt at handling pathnames with spaces. Revision 299794
+ attempted to improve the build system to be able to handle
+ pathnames (primarily DESTDIR) with spaces in them, since this is
+ common on some platforms (including Mac OSX). Unfortunately, the
+ changes were incomplete and did not actually provide the desired
+ behavior, and as a side effect the functionality that ensured
+ that stale headers in the Asterisk 'include' directory were
+ removed got broken. In addition, the check for stale (and
+ possibly incompatible) modules in the Asterisk 'modules'
+ directory also got broken, and would never report any stale
+ modules. Users upgrading to this version or later versions would
+ then see unexpected module load errors. Since there are few users
+ who actually want to install Asterisk into paths that contain
+ spaces, and a proper fix for the build system would take many
+ hours, the best solution for now is to just revert the partial
+ solution.
+
+2011-07-19 17:57 +0000 [r328770-328823] Kinsey Moore <kmoore@digium.com>
+
+ * include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c,
+ main/rtp_engine.c, channels/chan_sip.c: RTP bridge away with
+ inband DTMF and feature detection When deciding whether Asterisk
+ was allowed to bridge the call away from the core, chan_sip did
+ not take into account the usage of features on dialed channels
+ that require monitoring of DTMF on channels utilizing inband
+ DTMF. This would cause Asterisk to allow the call to be locally
+ or remotely bridged, preventing access to the data required to
+ detect activations of such features. (closes 17237) Review:
+ https://reviewboard.asterisk.org/r/1302/
+
+ * apps/app_meetme.c: MeetMe requests a PIN twice in some
+ circumstances If a call to MeetMe includes both the dynamic(D)
+ and always request PIN(P) options, MeetMe will ask for the PIN
+ two times: once for creating the conference and once for entering
+ the conference. This behavior was introduced in rev 311616 when
+ adding the CONFFLAG_ALWAYSPROMPT option to the logic branch
+ controlling PIN entry for joining a conference. (closes AST-601)
+ Review: https://reviewboard.asterisk.org/r/1305/
+
+2011-07-19 01:35 +0000 [r328716] Terry Wilson <twilson@digium.com>
+
+ * tests/test_linkedlists.c (added), include/asterisk/linkedlists.h:
+ Make AST_LIST_REMOVE safer AST_LIST_REMOVE shouldn't modify the
+ element passed in if it isn't found. This commit also adds linked
+ list unit tests. Review: https://reviewboard.asterisk.org/r/1321/
+
+2011-07-18 20:47 +0000 [r328593-328663] Mark Murawki <markm@intellasoft.net>
+
+ * apps/app_dial.c: app_dial may double free a channel datastore
+ When starting a call with originate, and having the callee
+ channel run Bridge() on pickup, we will double free the
+ dialed_interface_info datastore, causing a crash. Make sure to
+ check if the datastore still exists before trying to free it.
+ (closes issue ASTERISK-17917) Reported by: Mark Murawski Tested
+ by: Mark Murawski
+
+ * channels/chan_sip.c: If the sip private structure is null,
+ sip_setoption() will defref the null pointer and crash. Ideally,
+ sip_setoption shouldn't be called if there is a lack of a sip
+ private structure. But this will fix a crash. (closes issue
+ ASTERISK-17909) Reported by: Mark Murawski Tested by: Mark
+ Murawski
+
+ * main/asterisk.c: Fixed invalid read and null pointer deref on
+ asterisk shutdown. In some cases when starting asterisk with -c
+ and hitting control-c to shutdown, there will be an invalid read
+ and null pointer deref causing a crash. (closes issue
+ ASTERISK-17927) Reported by: Mark Murawski Tested by: Mark
+ Murawski, Kinsey Moore, Tilghman Lesher
+
+2011-07-18 07:10 +0000 [r328540] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * funcs/func_odbc.c: Typo
+
+2011-07-15 20:41 +0000 [r328446] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_macro.c, channels/chan_jingle.c, apps/app_dahdibarge.c,
+ apps/app_readfile.c, apps/app_setcallerid.c,
+ channels/chan_vpb.cc, apps/app_meetme.c, cdr/cdr_sqlite.c,
+ channels/chan_h323.c: Revert changes to defaultenabled state for
+ modules in Asterisk 1.8
+
+2011-07-15 19:22 +0000 [r328427] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooGkClient.c: small gk processing fixes: -
+ decrease for 1 second registration ttl for very low expirations
+ (some providers expire few earlier than TTL) - delete rrq and
+ registration expire timers on URQ received as we make new
+ registration.
+
+2011-07-14 23:12 +0000 [r328302] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Missing SIP pvt and channel unlock in
+ sip_set_rtp_peer(). Regression introduced by -r326144. Add
+ missing SIP pvt and channel unlock in sip_set_rtp_peer().
+
+2011-07-14 20:13 +0000 [r328209] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_image.c, res/res_http_post.c, formats/format_wav_gsm.c,
+ utils/stereorize.c, pbx/pbx_loopback.c, funcs/func_shell.c,
+ main/features.c, channels/chan_alsa.c, apps/app_externalivr.c,
+ formats/format_jpeg.c, res/res_speech.c, formats/format_gsm.c,
+ apps/app_milliwatt.c, formats/format_g719.c,
+ apps/app_saycounted.c, apps/app_fax.c, apps/app_echo.c,
+ funcs/func_math.c, channels/chan_agent.c, apps/app_dahdiras.c,
+ utils/astman.c, res/res_ael_share.c, apps/app_transfer.c,
+ apps/app_playback.c, res/res_config_curl.c, funcs/func_curl.c,
+ apps/app_waitforring.c, channels/chan_misdn.c, tests/test_skel.c,
+ addons/cdr_mysql.c, codecs/codec_ilbc.c, apps/app_zapateller.c,
+ apps/app_chanspy.c, apps/app_cdr.c, tests/test_substitution.c,
+ funcs/func_md5.c, utils/muted.c, tests/test_gosub.c,
+ funcs/func_sysinfo.c, funcs/func_vmcount.c, funcs/func_sha1.c,
+ cdr/cdr_radius.c, formats/format_siren7.c,
+ apps/app_controlplayback.c, funcs/func_config.c, main/manager.c,
+ bridges/bridge_builtin_features.c, funcs/func_volume.c,
+ cdr/cdr_sqlite.c, funcs/func_aes.c, funcs/func_frame_trace.c,
+ tests/test_devicestate.c, res/res_agi.c, tests/test_astobj2.c,
+ apps/app_confbridge.c, apps/app_ivrdemo.c,
+ res/res_clioriginate.c, res/res_calendar_icalendar.c,
+ funcs/func_dialplan.c, funcs/func_db.c,
+ tests/test_ast_format_str_reduce.c, res/res_fax.c,
+ res/res_limit.c, apps/app_festival.c, apps/app_waitforsilence.c,
+ apps/app_waituntil.c, channels/chan_console.c,
+ apps/app_getcpeid.c, apps/app_queue.c, funcs/func_global.c,
+ funcs/func_extstate.c, channels/chan_usbradio.c,
+ apps/app_flash.c, codecs/codec_ulaw.c, channels/chan_nbs.c,
+ formats/format_g729.c, funcs/func_dialgroup.c, funcs/func_env.c,
+ res/res_timing_dahdi.c, funcs/func_strings.c,
+ res/res_calendar_caldav.c, apps/app_chanisavail.c,
+ formats/format_sln16.c, apps/app_ices.c, apps/app_exec.c,
+ bridges/bridge_multiplexed.c, cel/cel_odbc.c,
+ formats/format_pcm.c, pbx/pbx_ael.c, formats/format_h263.c,
+ cdr/cdr_manager.c, res/res_clialiases.c, funcs/func_sprintf.c,
+ tests/test_app.c, apps/app_softhangup.c, codecs/codec_g726.c,
+ apps/app_morsecode.c, utils/smsq.c, bridges/bridge_simple.c,
+ tests/test_sched.c, apps/app_talkdetect.c, apps/app_db.c,
+ res/res_calendar_ews.c, funcs/func_callcompletion.c,
+ tests/test_acl.c, funcs/func_cdr.c, utils/ael_main.c,
+ utils/streamplayer.c, res/res_calendar.c, cel/cel_radius.c,
+ channels/chan_vpb.cc, res/res_snmp.c, apps/app_dictate.c,
+ apps/app_authenticate.c, res/res_phoneprov.c, funcs/func_logic.c,
+ res/res_jabber.c, funcs/func_uri.c,
+ funcs/func_audiohookinherit.c, res/res_config_odbc.c,
+ funcs/func_odbc.c, res/res_realtime.c, codecs/codec_resample.c,
+ formats/format_h264.c, apps/app_rpt.c, channels/chan_mgcp.c,
+ tests/test_amihooks.c, codecs/codec_lpc10.c, channels/chan_sip.c,
+ cdr/cdr_syslog.c, funcs/func_lock.c, res/res_adsi.c,
+ utils/conf2ael.c, tests/test_pbx.c, apps/app_channelredirect.c,
+ formats/format_vox.c, res/res_stun_monitor.c, tests/test_aoc.c,
+ formats/format_g723.c, utils/extconf.c, tests/test_poll.c,
+ addons/chan_ooh323.c, cdr/cdr_sqlite3_custom.c,
+ funcs/func_module.c, apps/app_sayunixtime.c,
+ cdr/cdr_adaptive_odbc.c, res/res_smdi.c, tests/test_time.c,
+ apps/app_skel.c, funcs/func_srv.c, apps/app_amd.c,
+ pbx/pbx_realtime.c, apps/app_url.c, apps/app_dial.c,
+ apps/app_page.c, channels/chan_bridge.c, apps/app_privacy.c,
+ codecs/codec_speex.c, apps/app_disa.c, res/res_mutestream.c,
+ res/res_monitor.c, apps/app_macro.c, res/res_timing_kqueue.c,
+ res/res_fax_spandsp.c, channels/chan_unistim.c,
+ funcs/func_base64.c, addons/app_mysql.c,
+ channels/chan_multicast_rtp.c, apps/app_meetme.c,
+ utils/hashtest.c, res/res_musiconhold.c, apps/app_followme.c,
+ res/res_config_sqlite.c, cdr/cdr_csv.c,
+ tests/test_security_events.c, formats/format_ilbc.c,
+ funcs/func_enum.c, channels/chan_phone.c,
+ tests/test_stringfields.c, funcs/func_groupcount.c,
+ tests/test_locale.c, addons/chan_mobile.c, cdr/cdr_custom.c,
+ res/res_security_log.c, apps/app_parkandannounce.c,
+ apps/app_while.c, apps/app_jack.c, res/res_rtp_asterisk.c,
+ apps/app_nbscat.c, codecs/codec_a_mu.c, tests/test_dlinklists.c,
+ res/res_convert.c, pbx/pbx_lua.c, utils/astcanary.c,
+ channels/chan_oss.c, tests/test_strings.c, res/res_srtp.c,
+ cdr/cdr_tds.c, res/res_timing_pthread.c,
+ apps/app_directed_pickup.c, channels/chan_h323.c,
+ cel/cel_sqlite3_custom.c, apps/app_senddtmf.c,
+ funcs/func_callerid.c, addons/app_saycountpl.c, cel/cel_pgsql.c,
+ funcs/func_speex.c, apps/app_dahdibarge.c, channels/chan_local.c,
+ tests/test_logger.c, apps/app_record.c, apps/app_playtones.c,
+ bridges/bridge_softmix.c, apps/app_alarmreceiver.c,
+ channels/chan_iax2.c, res/res_pktccops.c,
+ res/res_rtp_multicast.c, channels/chan_dahdi.c, pbx/pbx_spool.c,
+ funcs/func_pitchshift.c, channels/chan_skinny.c,
+ apps/app_dumpchan.c, main/http.c, cdr/cdr_odbc.c,
+ utils/refcounter.c, res/res_calendar_exchange.c, res/res_ais.c,
+ codecs/codec_g722.c, tests/test_expr.c, funcs/func_timeout.c,
+ cel/cel_tds.c, formats/format_wav.c, formats/format_ogg_vorbis.c,
+ funcs/func_cut.c, apps/app_speech_utils.c, apps/app_sendtext.c,
+ funcs/func_channel.c, utils/hashtest2.c, pbx/pbx_config.c,
+ funcs/func_iconv.c, apps/app_mixmonitor.c, formats/format_g726.c,
+ res/res_odbc.c, apps/app_voicemail.c, tests/test_heap.c,
+ addons/format_mp3.c, formats/format_sln.c, apps/app_readexten.c,
+ apps/app_userevent.c, codecs/codec_gsm.c, channels/chan_gtalk.c,
+ cdr/cdr_pgsql.c, tests/test_func_file.c, apps/app_setcallerid.c,
+ apps/app_osplookup.c, cel/cel_manager.c, cel/cel_custom.c,
+ tests/test_utils.c, apps/app_minivm.c, apps/app_mp3.c,
+ res/res_timing_timerfd.c, apps/app_directory.c,
+ res/res_config_ldap.c, formats/format_siren14.c,
+ apps/app_adsiprog.c, res/res_config_pgsql.c, apps/app_read.c,
+ funcs/func_version.c, codecs/codec_alaw.c, agi/eagi-test.c,
+ res/res_crypto.c, apps/app_originate.c, channels/chan_jingle.c,
+ apps/app_forkcdr.c, funcs/func_blacklist.c, pbx/pbx_dundi.c,
+ apps/app_sms.c, apps/app_stack.c, funcs/func_devstate.c,
+ apps/app_verbose.c, addons/res_config_mysql.c,
+ utils/check_expr.c, funcs/func_rand.c, apps/app_readfile.c,
+ codecs/codec_adpcm.c, apps/app_test.c, tests/test_event.c:
+ Introduce <support_level> tags in MODULEINFO. This change
+ introduces MODULEINFO into many modules in Asterisk in order to
+ show the community support level for those modules. This is used
+ by changes committed to menuselect by Russell Bryant recently
+ (r917 in menuselect). More information about the support level
+ types and what they mean is available on the wiki at
+ https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
+
+2011-07-14 19:21 +0000 [r328205] Jonathan Rose <jrose@digium.com>
+
+ * res/res_monitor.c: Monitor application arguments requirements
+ fixed. Monitor was requiring options in spite of no individual
+ option on Monitor being required. Review:
+ https://reviewboard.asterisk.org/r/1320/
+
+2011-07-13 18:46 +0000 [r328014] Richard Mudgett <rmudgett@digium.com>
+
+ * configs/features.conf.sample: Add ATXFER_NULL_TECH note in
+ features.conf.sample.
+
+2011-07-12 22:53 +0000 [r327950] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/manager.c: Correct double-free situation in manager output
+ processing. The process_output() function calls ast_str_append()
+ and xml_translate() on its 'out' parameter, which is a pointer to
+ an ast_str buffer. If either of these functions need to
+ reallocate the ast_str so it will have more space, they will free
+ the existing buffer and allocate a new one, returning the address
+ of the new one. However, because process_output only receives a
+ pointer to the ast_str, not a pointer to its caller's variable
+ holding the pointer, if the original ast_str is freed, the caller
+ will not know, and will continue to use it (and later attempt to
+ free it). (reported by jkroon on #asterisk-dev)
+
+2011-07-12 20:07 +0000 [r327890] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_directory.c: search in the current context for 'a' and
+ 'o' instead of 'default'
+
+2011-07-12 19:38 +0000 [r327888] Jason Parker <jparker@digium.com>
+
+ * Makefile: Fix uninstall target, so that modules dir gets cleared
+ again.
+
+2011-07-12 19:10 +0000 [r327852] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_voicemail.c: Added additional checks for mailbox /
+ password beginning with '*' character A bug existed such that if
+ a user entered a password with '*', and the extension 'a' did not
+ exist, an invalid mailbox would be created and the user
+ authenticated. The code was changed to prevent this from
+ occurring, and to prevent users from having mailboxes or
+ passwords defined that begin with the '*' character. (closes
+ issue ASTERISK-17443) Reported by: Kevin Scott Adams Tested by:
+ Matt Jordan Review: https://reviewboard.asterisk.org/r/1316/
+
+2011-07-12 15:35 +0000 [r327793] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * tests/test_substitution.c: Use 'printf' (POSIX issue 4) instead
+ of 'echo -n', for portability. The problem with using 'echo -n'
+ is that it is not portable. While BSD systems required that the
+ '-n' option be removed and interpreted, System V required that
+ all strings should be echoed with no interpretation of options.
+ This fundamental difference of behavior means that it is never
+ possible to use the '-n' flag to echo in tests which are meant to
+ be portable. In this case, on Mac OS X 10.6, the /bin/sh shell
+ builtin 'echo' uses the System V semantics of the command, and
+ thus the SHELL test failed on that platform.
+ http://pubs.opengroup.org/onlinepubs/009695399/utilities/echo.html#tag_04_41_16
+
+2011-07-11 19:41 +0000 [r327682] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/jingle.h, channels/chan_gtalk.c: Update
+ chan_gtalk to work with changed GMail-based calls The messages
+ sent by the GMail client have changed, but include the old-style
+ messages as well. This patch checks for this case and uses the
+ old-style offer. (closes issue ASTERISK-18084) Review:
+ https://reviewboard.asterisk.org/r/1312/
+
+2011-07-11 13:53 +0000 [r327512] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, tests/test_substitution.c: reset our buffer each
+ iteration when doing variable substitution
+
+2011-07-11 10:56 +0000 [r327411-327412] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * main/Makefile: Properly building the Debian armhf (HardFloat)
+ port. Remove the line that should have been removed in r327411.
+
+ * main/Makefile: fix building the Debian armhf (HardFloat) port
+ Fixes
+ http://buildd.debian-ports.org/status/fetch.php?pkg=asterisk&arch=armhf&ver=1%3A1.8.4.4~dfsg-2&stamp=1309935385
+ (Missing pthreads)
+
+2011-07-08 22:27 +0000 [r327258] Jason Parker <jparker@digium.com>
+
+ * main/db1-ast/mpool, addons, cdr, formats, codecs/gsm/src, funcs,
+ addons/ooh323c/src, bridges, codecs/lpc10, main/db1-ast/btree,
+ codecs/g722, main, main/db1-ast/recno, channels/sip, res, pbx,
+ res/ael, channels, main/stdtime, addons/ooh323c/src/h323, codecs,
+ utils, main/db1-ast/hash, cel, apps, main/db1-ast/db: Add .o
+ files to svn:ignore property, since it's only ignored if locally
+ configured to do so.
+
+2011-07-08 21:41 +0000 [r327211] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: INVITE 403 Forbidden response always
+ retransmits the maximum times. Asterisk sends a 403 Forbidden
+ response if authentication fails for an INVITE as required.
+ However, it ignores the ACK and keeps retransmitting the
+ response. * Made not delete the to-tag in the dialog so the
+ expected ACK can be matched with the dialog and stop the
+ retransmissions.
+
+2011-07-08 19:52 +0000 [r327106] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, tests/test_substitution.c: Reset our ast_str before
+ passing it on to dialplan function backends. It is possible for a
+ dialplan backend to not modify the given buffer or ast_str and
+ still return success. This causes any previous value stored in
+ the buffer to be used as if the new function call provided it.
+ Some functions also append to the given buffer assuming it is
+ empty. The test_substitution unit test has also been modified to
+ detect this problem. (closes issue ASTERISK-17878)
+
+2011-07-08 16:00 +0000 [r327044-327046] Russell Bryant <russell@digium.com>
+
+ * tests/test_netsock2.c: Fix an error and add more log message info
+ to help see why this fails on FreeBSD.
+
+ * channels/chan_dahdi.c: Resolve some set-but-unused-variable
+ warnings.
+
+2011-07-08 01:08 +0000 [r326985] Richard Mudgett <rmudgett@digium.com>
+
+ * main/pbx.c: Some code cleanup in pbx.c * Mostly comment and
+ format changes. * ast_context_remove_extension_callerid() and
+ ast_add_extension_nolock() will write lock the found specific
+ context. * ast_context_find() will now tolerate a NULL name. *
+ Eliminated some inlined versions of find_context() and
+ find_context_locked().
+
+2011-07-07 19:17 +0000 [r326830] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * res/res_http_post.c: libgen.h is also needed on Darwin for
+ basename(3)
+
+2011-07-07 16:04 +0000 [r326689] Jonathan Rose <jrose@digium.com>
+
+ * res/res_config_odbc.c: res_odbc patch by tilghman to fix integers
+ with null values Addresses some improper sql statements in
+ res_odbc that would cause an update to fail on realtime peers due
+ to trying to set as "(NULL)" rather than an actual NULL. (closes
+ issue #1922STERISK-17791) Reported by: marcelloceschia Patches:
+ 20110505__issue19223.diff.txt uploaded by tilghman (license 14)
+
+2011-07-07 15:28 +0000 [r326681-326683] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: use sips: or sip: depending on the transport
+ in use when building reply digest URIs
+
+ * channels/chan_sip.c: make the uri parameter used in reply digests
+ more standards compliant in certain cases by prepending "sip:" or
+ "sips:" to it
+
+2011-07-06 15:26 +0000 [r326484] David Vossel <dvossel@digium.com>
+
+ * res/res_timing_timerfd.c: Reverts fix for timerfd locking issue.
+ jrose discovered a performance issue with this fix that prevents
+ his analog phones from working when using timerfd as a timing
+ source. Until it is understood what is causing this performance
+ problem, this patch is being reverted.
+
+2011-07-06 14:35 +0000 [r326411-326469] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * pbx/pbx_dundi.c, channels/chan_gtalk.c, apps/app_queue.c,
+ channels/chan_iax2.c, res/res_jabber.c, apps/app_stack.c,
+ channels/chan_mgcp.c, apps/app_voicemail.c,
+ channels/chan_jingle.c, channels/chan_dahdi.c,
+ funcs/func_speex.c, channels/chan_sip.c, codecs/codec_speex.c,
+ funcs/func_aes.c: Removing type attributes, as a change to
+ menuselect makes them no longer necessary.
+
+ * pbx/pbx_dundi.c, channels/chan_gtalk.c, apps/app_queue.c,
+ channels/chan_iax2.c, res/res_jabber.c, apps/app_stack.c,
+ channels/chan_mgcp.c, apps/app_voicemail.c,
+ channels/chan_jingle.c, channels/chan_dahdi.c,
+ funcs/func_speex.c, channels/chan_sip.c, codecs/codec_speex.c,
+ funcs/func_aes.c: Add the attribute "type" to each "<use>" for
+ menuselect. This matters only when autoconf fails to detect that
+ weak linking is supported. External optional dependencies will
+ become optional in both cases, as they are removed at compile
+ time when not detected. However, runtime-optional modules are
+ made mandatory when weak linking is not found. This change
+ affects only the external optional dependencies; previously, they
+ were incorrectly required when weak linking support was not
+ detected. Patches: 20110702__issue18062__asterisk_trunk.diff.txt
+ by tilghman (License #5003) Tested by: iasgoscouk
+
+2011-07-05 17:22 +0000 [r326291] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: Used auth=
+ parameter freed during "sip reload" causes crash. If you use the
+ auth= parameter and do a "sip reload" while there is an ongoing
+ call. The peer->auth data points to free'd memory. The patch does
+ several things: 1) Puts the authentication list into an ao2
+ object for reference counting to fix the reported crash during a
+ SIP reload. 2) Converts the authentication list from open coding
+ to AST list macros. 3) Adds display of the global authentication
+ list in "sip show settings". (closes issue ASTERISK-17939)
+ Reported by: wdoekes Patches: jira_asterisk_17939_v1.8.patch
+ (license #5621) patch uploaded by rmudgett Review:
+ https://reviewboard.asterisk.org/r/1303/ JIRA SWP-3526
+
+2011-07-05 13:23 +0000 [r326209] Matthew Jordan <mjordan@digium.com>
+
+ * main/file.c: Updated filestream destructor to block until move is
+ complete when cache is used When a cache directory is used, the
+ process is forked and a mv command is executed to move the
+ temporary file to the permanent location. This caused issues with
+ voicemail, where a race condition occurred when the parent
+ expected the file to be in the permanent location prior to the mv
+ command completing. The parent process is now blocked until the
+ mv command completes. (closes issue ASTERISK-17724) Reported by:
+ Adiren P. Tested by: mjordan
+
+2011-07-01 21:07 +0000 [r326144] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Better way to get chan and pvt lock for
+ issue ASTERISK-17431. Redoes -r308945 for issue ASTERISK-17431
+ deadlock fix for sip_set_udptl_peer() and sip_set_rtp_peer(). *
+ Lock the channels in the defined order and avoid the need for a
+ deadlock avoidance loop. * Lock the channel before getting the
+ pointer to the private structure to be sure that the pointer will
+ not change due to a masquerade or channel hangup. * To preserve
+ sanity, check that chan and p->owner are the same. (Pointer
+ rearangements should not happen without the protection of locks
+ because bad things tend to happen otherwise.)
+
+2011-06-30 20:39 +0000 [r325935] Richard Mudgett <rmudgett@digium.com>
+
+ * configs/sip.conf.sample, channels/chan_sip.c: Misc minor changes
+ in chan_sip. * Add load failure exit if primary SIP container(s)
+ could not get created in chan_sip.c:load_module(). * Removed a
+ redundant static prototype. * Some typos. * Some whitespace.
+
+2011-06-30 20:09 +0000 [r325877] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_voicemail.c: Patched voicemail user option for emailbody
+ / emailsubject Incorporated changes per ASTERISK-16795; updated
+ unit tests to check for vmu->emailbody / vmu->emailsubject
+ (closes issue ASTERISK-16795) Reported by: mdeneen Tested by:
+ mjordan
+
+2011-06-30 19:17 +0000 [r325821] Jonathan Rose <jrose@digium.com>
+
+ * res/res_musiconhold.c: Fixes an issue with Music on Hold classes
+ losing files in playlist when realtime is used. The bug occurs
+ rather intermittently and I relied on the reporters to test the
+ patch. After a sanity check and some testing, I'm giving it an
+ OK. (closes issue ASTERISK-17875) Reported by: David Cunningham
+ Patches: res_musiconhold.c.mohrt17875_v1 uploaded by Igor
+ Goncharovsky (license #5009)
+
+2011-06-29 21:49 +0000 [r325740] Kinsey Moore <kmoore@digium.com>
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: chan_sip:
+ cleanup from the introduction of ast_str Remove the length field
+ from sip_req and sip_pkt in chan_sip since they are redundant
+ (ast_str holds its own length) and refactor the necessary
+ functions. Review: https://reviewboard.asterisk.org/r/1281/
+
+2011-07-11 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.5.0 Released.
+
+ * r326484 | dvossel | 2011-07-06 10:26:49 -0500 (Wed, 06 Jul 2011)
+
+ Reverts fix for timerfd locking issue.
+
+ jrose discovered a performance issue with this
+ fix that prevents his analog phones from working
+ when using timerfd as a timing source. Until
+ it is understood what is causing this performance
+ problem, this patch is being reverted.
+
+2011-06-29 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.5-rc1 Released.
+
+2011-06-29 18:59 +0000 [r325673] David Vossel <dvossel@digium.com>
+
+ * res/res_timing_timerfd.c: Fixes timerfd locking issue. (closes
+ ASTERISK-17867, ASTERISK-17415) Patches: fix uploaded by kobaz
+ https://reviewboard.asterisk.org/r/1255/
+
+2011-06-29 18:16 +0000 [r325610-325614] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_queue.c: Fixed some error exit cleanup in app_queue.c. *
+ Fixed error exit cleanup in app_queue.c copy_rules() and
+ reload_queue_rules().
+
+ * apps/app_queue.c: Response to QueueRule manager command does not
+ contain ActionID if it was specified. * Add ActionID support as
+ documented for the QueueRule AMI action. * Remove documentation
+ for ActionID with the Queues AMI action. The output does not
+ follow normal AMI response output and there is no place to put an
+ ActionID header. (closes issue AST-602) Reported by: Vlad
+ Povorozniuc Patches: jira_ast_602_v1.8.patch (license #5621)
+ patch uploaded by rmudgett Tested by: Vlad Povorozniuc, rmudgett
+ Review: https://reviewboard.asterisk.org/r/1295/ JIRA SWP-3575
+
+2011-06-29 16:18 +0000 [r325537-325545] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c: make framehooks prevent native bridging (for real
+ this time)
+
+ * apps/app_dial.c, main/rtp_engine.c: don't do native/remote
+ bridging if a framehook is active on the channel
+
+2011-06-28 21:50 +0000 [r325416] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: Fix random misspelling noticed on
+ asterisk-users.
+
+2011-06-28 20:31 +0000 [r325339] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Fixes locking inversion caused by holding
+ sip pvt lock during async_goto. (closes ASTERISK-17352)
+
+2011-06-28 20:07 +0000 [r325279] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 325277 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r325277 | twilson | 2011-06-28 15:06:16 -0500
+ (Tue, 28 Jun 2011) | 9 lines Merged revisions 325275 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r325275 | twilson | 2011-06-28 15:03:19 -0500 (Tue, 28
+ Jun 2011) | 2 lines Don't leak SIP username information ........
+ ................
+
+2011-06-28 17:30 +0000 [r325212] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Use the device name and not the channel
+ name to initialize the device state. Correct ASTERISK-11323
+ implementation as I don't see how it ever worked as claimed when
+ it used the channel name and not the device name. (issue
+ ASTERISK-11323)
+
+2011-06-28 15:46 +0000 [r325152] Jonathan Rose <jrose@digium.com>
+
+ * res/res_musiconhold.c: Fixes moh reload breaking custom mode moh
+ classes when the config file is untouched (closes issue
+ ASTERISK-17730) Reported by: sdolloff
+
+2011-06-28 15:12 +0000 [r325091] Leif Madsen <lmadsen@digium.com>
+
+ * build_tools/prep_tarball: Remove line from prep_tarball that
+ kills mkrelease.
+
+2011-06-27 16:30 +0000 [r324955] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/asterisk.c: Save and restore errno from within signal
+ handlers. This is recommended by the POSIX standard, as well as
+ by the sigaction(2) manpage for various platforms that we support
+ (e.g. Mac OS X).
+
+2011-06-27 15:37 +0000 [r324914] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: When subscribing MWI to an unsolicited
+ mailbox the first notification is incorrect. A remote peer
+ subscribed to MWI with the unsolicited option and a local phone
+ subscribed to the remote mailbox. The notify message-summary
+ events are sent correctly except for the first one when
+ subscribing, which will always be 0. This means the phone MWI
+ indicator will be wrong until the mailbox read/unread count
+ changes and the event is fired. Looks like this is a regression
+ from ASTERISK-16149. * Fix the logic to check the cache and if
+ allowed then fallback to manually counting mailbox messages.
+ (closes issue ASTERISK-17997) Reported by: rsw686 Patches:
+ jira_asterisk_17997_v1.8.patch (license #5621) uploaded by
+ rmudgett Tested by: rsw686 JIRA SWP-3551
+
+2011-06-24 20:46 +0000 [r324849] Richard Mudgett <rmudgett@digium.com>
+
+ * pbx/pbx_config.c: Syntax errors in dialplan do not display the
+ file name. When issuing the CLI command "dialplan reload" syntax
+ errors and warnings are displayed on the console. The offending
+ line number is displayed on the console, but the file name is not
+ displayed. Errors caught in main/config.c do display the file
+ name. (closes issue ASTERISK-17985) Reported by: ulogic Patches:
+ pbx_config.patch uploaded by ulogic (License #5685) modified
+ format Tested by: rmudgett JIRA SWP-3554
+
+2011-06-24 16:48 +0000 [r324768] Jonathan Rose <jrose@digium.com>
+
+ * include/asterisk/logger.h: DTMF wasn't being logged on connected
+ consoles when enabled in logger.conf Previously in order for DTMF
+ to be logged in a connected console session, the user would have
+ to do logger set channel DTMF on. This corrects that so that it
+ is on by default. This issue was caused by an off by one error
+ incurred by a logger level count of 6 in logger.h where it should
+ have been 7. (closes issue: ASTERISK-17974) Reported by: Luke H
+
+2011-06-23 18:31 +0000 [r324685] David Vossel <dvossel@digium.com>
+
+ * channels/sip/reqresp_parser.c: Fixes sip crash when calling
+ remove_uri_parameters with NULL AST-2011-009 (closes issue
+ ASTERISK-18017) Reported by: jaredmauch
+
+2011-06-23 18:29 +0000 [r324678] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 324643 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r324643 | kmoore | 2011-06-23 13:21:12 -0500 (Thu, 23 Jun 2011) |
+ 4 lines Addresses AST-2011-008, memory corruption and remote
+ crash in SIP driver. AST-2011-008 ........
+
+2011-06-23 18:23 +0000 [r324652] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, include/asterisk/frame.h, /,
+ main/features.c: Merged revisions 324634 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r324634 | dvossel | 2011-06-23 13:18:46 -0500
+ (Thu, 23 Jun 2011) | 13 lines Merged revisions 324627 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r324627 | dvossel | 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011)
+ | 7 lines Addresses AST-2011-010, remote crash in IAX2 driver
+ Thanks to twilson for identifying the issue and providing the
+ patches. AST-2011-010 ........ ................
+
+2011-06-23 03:10 +0000 [r324557] Terry Wilson <twilson@digium.com>
+
+ * tests/test_netsock2.c: Remove tests for parsing address with
+ invalid port getaddrinfo on OS X returns with EAI_NONAME error
+ when passed a port greater than 65535. Linux throws no error, so
+ remove the tests for now.
+
+2011-06-22 19:16 +0000 [r324491] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Use correct variable for text SRTP media.
+
+2011-06-22 18:52 +0000 [r324484] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/netsock2.h, tests/test_netsock2.c (added),
+ main/netsock2.c, channels/chan_sip.c: Stop sending IPv6
+ link-local scope-ids in SIP messages The idea behind the patch
+ listed below was used, but in a more targeted manner. There are
+ now address stringification functions for addresses that are
+ meant to be sent to a remote party. Link-local scope-ids only
+ make sense on the machine from which they originate and so are
+ stripped in the new functions. There is also a host sanitization
+ function added to chan_sip which is used for when peer and dialog
+ tohost fields or sip_registry hostnames are used to craft a SIP
+ message. Also added are some basic unit tests for netsock2
+ address parsing. (closes issue ASTERISK-17711) Reported by:
+ ch_djalel Patches: asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded
+ by ch_djalel (license 1251) Review:
+ https://reviewboard.asterisk.org/r/1278/
+
+2011-06-22 18:41 +0000 [r324479-324481] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Timout or error on INFO or MESSAGE
+ transaction causes call to be lost. When exchanging INFO messages
+ within a call, 4xx error causes the call to be disconnected
+ although RFC 2976 explicitly states that such transactions do not
+ modify the state of the dialog. When exchanging MESSAGE messages
+ within a call, 4xx error causes the call to be disconnected. To
+ provide least surprise, we should not disconnect the call since a
+ MESSAGE is like INFO in this case. (Implied by RFC 3428 Section
+ 2) (closes issue ASTERISK-17901) Reported by: neutrino88 Review:
+ https://reviewboard.asterisk.org/r/1257/ Review:
+ https://reviewboard.asterisk.org/r/1258/ JIRA SWP-3486
+
+ * channels/chan_sip.c: Comments and whitespace in chan_sip.c
+
+2011-06-21 20:11 +0000 [r324364] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/pbx.h, main/pbx.c: Fixes locking inversion issue
+ in ast_async_goto() During this function we can not hold the
+ "chan" lock while doing the masquerade, the explicit goto on the
+ tmp chan, or the channel alloc. Instead we need to get the
+ channel lock, store off information about the channel that we
+ need, and then let the channel lock go for the remainder of the
+ function. Review: https://reviewboard.asterisk.org/r/1275/
+
+2011-06-21 16:09 +0000 [r324305] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_confbridge.c: ConfBridge does not handle hangup properly
+ When playing back a prompt to a channel, confbridge neglects to
+ check for hangup events causing lockup condititions for hangups
+ that occur before actually joining the conference. This change
+ ensures that the user is removed from the conference in the event
+ of a premature hangup. Review:
+ https://reviewboard.asterisk.org/r/1277/
+
+2011-06-20 18:12 +0000 [r324239-324241] Leif Madsen <lmadsen@digium.com>
+
+ * configs/queuerules.conf.sample: Remove extra 'the'. Reported by
+ Vlad Povorozniuc
+
+ * configs/queuerules.conf.sample,
+ contrib/scripts/asterisk.logrotate: Revert previous merge which
+ had extra changes.
+
+ * configs/queuerules.conf.sample,
+ contrib/scripts/asterisk.logrotate: Remove extra 'the'. Reported
+ by Vlad Povorozniuc
+
+2011-06-20 17:33 +0000 [r324237] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Ignore media offers with a port of 0 Section
+ 5.1 of RFC3264 states: A port number of zero in the offer
+ indicates that the stream is offered but MUST NOT be used.
+ (closes issue ASTERISK-17845) Reported by: jacco Patches:
+ issue19281_2.patch uploaded by jacco (license 1277) Tested by:
+ jacco, twilson
+
+2011-06-17 18:51 +0000 [r324176-324178] Leif Madsen <lmadsen@digium.com>
+
+ * main/manager.c: Add Username and Secret fields to manager Login
+ action. Pointed out by Vlad Povorozniuc
+
+ * apps/app_meetme.c: Fix typo in documentation. Pointed out by Vlad
+ Povorozniuc
+
+2011-06-17 18:23 +0000 [r324174] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Add header string to libpri debug output.
+ Add header string to libpri debug output so the libpri output can
+ be found/extracted easier from huge debug trace files.
+
+2011-06-17 15:14 +0000 [r324115] Leif Madsen <lmadsen@digium.com>
+
+ * main/pbx.c: Fix grammar in documentation for Goto() and GotoIf()
+ (closes issue ASTERISK-18023) Reported by: Tim Osman
+
+2011-06-16 22:41 +0000 [r324048-324049] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c: Shame on me
+
+ * include/asterisk/channel.h, main/channel.c,
+ channels/chan_local.c, channels/chan_sip.c: Lock the channel
+ before calling the setoption callback The channel needs to be
+ locked before calling these callback functions. Also,
+ sip_setoption needs to lock the pvt and a check p->rtp is
+ non-null before using it. Review:
+ https://reviewboard.asterisk.org/r/1220/
+
+2011-06-16 18:12 +0000 [r323990] Richard Mudgett <rmudgett@digium.com>
+
+ * tests/test_event.c: The test_event unit test is occasionally
+ failing. Wait for the special posted event to process before
+ adding a new subscription.
+
+2011-06-16 15:58 +0000 [r323754-323932] Terry Wilson <twilson@digium.com>
+
+ * Makefile: Don't assume ASTDBDIR exists It most likely doesn't on
+ FreeBSD
+
+ * tests/test_db.c: Remove now-useless cast of ARRAY_LEN
+
+ * include/asterisk/utils.h: Make ARRAY_LEN() return the same type
+ on x86 and x86_64 systems
+
+ * tests/test_db.c: Fix more ARRAY_LEN format string issues
+
+ * /, main/features.c: Merged revisions 323733 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r323733 | twilson | 2011-06-15 13:13:00 -0500
+ (Wed, 15 Jun 2011) | 16 lines Merged revisions 323732 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r323732 | twilson | 2011-06-15 13:06:24 -0500 (Wed, 15 Jun 2011)
+ | 9 lines Fix DYNAMIC_FEATURES DYNAMIC_FEATURES were broken by a
+ recent DTMF change. This patch makes sure that dynamic features
+ are also checked when deciding whether or not to pass DTMF
+ through or store it for interpreting. (closes issue
+ ASTERISK-17914) Reported by: vrban ........ ................
+
+2011-06-15 17:42 +0000 [r323730] Jonathan Rose <jrose@digium.com>
+
+ * res/res_config_pgsql.c: Adds locking to find_table in
+ res_configure_pgsql to prevent a crash. Bryonclark described the
+ problem as occuring during this function because of multiple
+ simultaneous database operations causing corruption against a
+ pgsqlConn object. (closes issue ASTERISK-17811) Reported by:
+ byronclark Patches: pgsql_find_table_locking.patch uploaded by
+ byronclark (license 1200)
+
+2011-06-15 17:09 +0000 [r323672] Terry Wilson <twilson@digium.com>
+
+ * tests/test_db.c: Cast ARRAY_LEN to size_t for ast_logging 32-bit
+ and 64-bit machines return different types for ARRAY_LEN(), so
+ cast it before using in a format string.
+
+2011-06-15 16:43 +0000 [r323669-323670] Richard Mudgett <rmudgett@digium.com>
+
+ * tests/test_event.c: Add a test to the event unit tests to catch
+ ASTERISK-18002. The new tests check to see if there are ANY
+ subscribers to the event type when ast_event_check_subscriber()
+ is not passed any specific ie values. (issue ASTERISK-18002)
+
+ * main/event.c: [regression] Voicemail MWI is no longer sent. When
+ leaving a voicemail, the MWI message is never sent. The same
+ thing happens when checking a voicemail and marking it as read.
+ If you restart Asterisk, everything comes up at that state
+ correctly, but changes to the messages in voicemail causes the
+ light to not be set appropriately. Very easy to reproduce. * Made
+ ast_event_check_subscriber() return TRUE if there are ANY
+ subscribers to an event type when there are no restricting ie
+ values passed. This allows an event being queued to be queued.
+ (closes issue ASTERISK-18002) Reported by: lmadsen Tested by:
+ lmadsen, irroot Patches: jira_asterisk_18002_v1.8.patch uploaded
+ by rmudgett (License #5621) (closes issue ASTERISK-18019)
+
+2011-06-15 16:09 +0000 [r323610] Jonathan Rose <jrose@digium.com>
+
+ * res/res_config_pgsql.c: Adds PQclear calls on result to various
+ parts of res_conf_pgsql (closes issue ASTERISK-17812) Reported
+ by: byronclark Patches: pgsql_pqclear.patch uploaded by
+ byronclark (license 1200)
+
+2011-06-15 15:31 +0000 [r323608] Sean Bright <sean@malleable.com>
+
+ * main/manager.c, /: Merged revisions 323579 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r323579 | seanbright | 2011-06-15 11:22:50 -0400
+ (Wed, 15 Jun 2011) | 32 lines Merged revisions 323559 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun
+ 2011) | 25 lines Resolve a segfault/bus error when we try to map
+ memory that falls on a page boundary. The fix for ASTERISK-15359
+ was incorrect in that it added 1 to the length of the mmap'd
+ region. The problem with this is that reading/writing to that
+ extra byte outside of the bounds of the underlying fd causes a
+ bus error. The real issue is that we are working with both a FILE
+ * and the raw fd underneath it and not synchronizing between
+ them. The code that was removed in ASTERISK-15359 was correct,
+ but we weren't flushing the FILE * before mapping the fd. Looking
+ at the manager code in 1.4 reveals that the FILE * in 'struct
+ mansession' is never used except to create a temporary file that
+ we immediately fdopen. This means we just need to write a 0 byte
+ to the fd and everything will just work. The other branches
+ require a call to fflush() which, while not a guaranteed fix,
+ should reduce the likelihood of a crash. This all makes sense in
+ my head. (closes issue ASTERISK-16460) Reported by:
+ Ravelomanantsoa Hoby (hoby) Patches:
+ issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license
+ #5060) ........ ................
+
+2011-06-15 00:50 +0000 [r323392-323456] Richard Mudgett <rmudgett@digium.com>
+
+ * main/event.c: Add missing break in ast_event_get_cached().
+
+ * main/netsock2.c: Made ast_sockaddr_split_hostport() port warning
+ msgs more meaningful.
+
+ * main/dnsmgr.c: Add more strict hostname checking to
+ ast_dnsmgr_lookup(). Change suggested in review. Review:
+ https://reviewboard.asterisk.org/r/1240/
+
+2011-06-14 16:38 +0000 [r323371] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c: Changes contact use in build_peer to use the
+ FORCE_RPORT flag instead of RPORT_PRESENT It turned out that this
+ was causing NAT=Yes to always use rport when present which was
+ against 1.6.2 behavior and the check itself was redundant since
+ the only way this segment of code could be reached was if
+ RPORT_PRESENT was already evaluated as true earlier. (closes
+ issue ASTERISK-17789) Reported by: byronclark Patches:
+ use_sip_nat_force_rport.patch uploaded by byronclark (license
+ 1200)
+
+2011-06-14 16:33 +0000 [r323370] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c,
+ main/rtp_engine.c, channels/chan_sip.c: Add rtpkeepalives back to
+ 1.8 The RTP-engine conversion left out support for handling
+ rtpkeepalives. This patch adds them back. (closes issue
+ ASTERISK-17304) Reported by: lmadsen Review:
+ https://reviewboard.asterisk.org/r/1226/
+
+2011-06-13 20:22 +0000 [r323154-323234] Leif Madsen <lmadsen@digium.com>
+
+ * configs/sip.conf.sample: Additional documentation for bindaddr.
+ Note that bindaddr will only enable UDP instead of both UDP and
+ TCP which is what I would expect for backwards compatibility with
+ systems being upgraded which only support UDP transportation.
+ (closes issue ASTERISK-17976) Reported by: Sean Darcy
+
+ * main/channel.c: Avoid dividing by zero with L() option to Dial()
+ Reported by: nicolasom Patches: issue-17995.patch - nicolasom
+ (License #5994)
+
+ * res/res_agi.c: Tweak documentation for AGI Hangup command.
+ (closes issue ASTERISK-17999) Reported by: Ben Klang Patches:
+ hangup-doc.diff - uploaded by Ben Klang (License #5876)
+
+2011-06-10 19:20 +0000 [r323040] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Unlock the sip channel during fax detection
+ like chan_dahdi does to prevent a deadlock with
+ ast_autoservice_stop. (closes issue ASTERISK-17798) tested by
+ mnicholson
+
+2011-06-10 15:29 +0000 [r322865-322981] Terry Wilson <twilson@digium.com>
+
+ * main/db.c: Avoid a DB1 infinite loop bug Explicity check the last
+ entry in the DB and make sure that we don't iterate past it.
+ Since there can be no duplicates, this just makes sure that we
+ stop after matching the last key. This patch also refactors the
+ code to get away from some code duplication. A previous patch
+ added many astdb tests and this patch passed them. Review:
+ https://reviewboard.asterisk.org/r/1259/
+
+ * tests/test_db.c (added): Add some astdb unit tests
+
+ * include/asterisk/astdb.h: Correct ast_db_deltree documentation
+ ast_db_deltree returns -1 on error, otherwise the number of
+ deletions
+
+2011-06-09 17:37 +0000 [r322807] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: don't drop any voice frames when checking
+ for T.38 during early media (closes issue ASTERISK-17705) Review:
+ https://reviewboard.asterisk.org/r/1186/ patch by oej reported by
+ oej
+
+2011-06-09 16:31 +0000 [r322749] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/features.h, apps/app_directed_pickup.c,
+ main/features.c: Remove potential deadlock in call pickup race.
+ Deadlock is possible in ast_do_pickup() when holding the target
+ channel lock and trying to get the chan channel lock. Also,
+ holding the target lock when calling ast_channel_masquerade() is
+ not a good idea because that routine does deadlock avoidance. *
+ Removed the need to hold the target lock after marking the target
+ with a datastore and getting the connected line data off of the
+ target channel. * Moved can_pickup() to ast_can_pickup() in
+ features.c. Now all the call pickup methods use the same basic
+ call pickup availability check. Review:
+ https://reviewboard.asterisk.org/r/1234/
+
+2011-06-09 14:06 +0000 [r322585] Jonathan Rose <jrose@digium.com>
+
+ * main/utils.c, include/asterisk/utils.h, channels/chan_sip.c,
+ tests/test_utils.c: Adds ast_escape_encoded utility to properly
+ handle escaping of quoted field before uri. This commit backports
+ a feature in trunk affecting initreqprep so that display name
+ won't be encoded improperly. Also includes unit tests for the
+ ast_escape_quoted function. This patch gives 1.8 a much improved
+ outlook in countries which don't use standard ASCII characters.
+ (closes issue ASTERISK-16949) Reported by: Örn Arnarson Review:
+ https://reviewboard.asterisk.org/r/1235/
+
+2011-06-08 20:46 +0000 [r322425-322484] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_queue.c: Ring all queue with more than 255 agents will
+ cause crash. 1. Create a ring-all queue with 500 permanent
+ agents. 2. Call it. 3. Asterisk will crash. The watchers array in
+ app_queue.c has a hard limit of 255. Bounds checking is not done
+ on this array. No sane person should put 255 people in a ring-all
+ queue, but we should not crash anyway. * Added bounds checking to
+ the watchers array. JIRA AST-464 JIRA SWP-2903
+
+ * main/dnsmgr.c: SRV lookup attempted for SIP peers listed as an IP
+ address. Asterisk attempts to SRV lookup a host name even if the
+ host name is an IP address. Regression introduced when IPv6
+ support was added. * Restored the check in ast_dnsmgr_lookup() to
+ see if the given host name is an IP address. The IP address could
+ be in either IPv4 or IPv6 formats. (closes issue ASTERISK-17815)
+ Reported by: Byron Clark Tested by: Byron Clark, Richard Mudgett
+ Patches: issue19248_v1.8.patch - uploaded by Richard Mudgett
+ (License #5621) Review: https://reviewboard.asterisk.org/r/1240/
+
+2011-06-08 06:18 +0000 [r322322] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * channels/chan_sip.c: Make handle_request_publish do dialog
+ expiration and destruction. This patch fixes
+ handle_request_publish so that it does dialog expiration and
+ destruction. Without this patch the incoming PUBLISH requests
+ will get stuck in the dialog list. Restarting asterisk is the
+ only way to remove them. Personal observation on one system the
+ server hung up while looping through the channels rendering
+ asterisk unusable and all sip phones unregisterd when they try
+ reregister more requests are added. (closes issue #18898)
+ Reported by: gareth Tested by: loloski, Chainsaw, wimpy, se, kuj,
+ irroot Jira:
+ https://issues.asterisk.org/jira/browse/ASTERISK-17915 Review:
+ https://reviewboard.asterisk.org/r/1253
+
+2011-06-07 17:59 +0000 [r322189] Paul Belanger <pabelanger@digium.com>
+
+ * configs/sip_notify.conf.sample: Use correct syntax for 'sip
+ notify snom-reboot' (closes issue ASTERISK-17915)
+
+2011-06-06 19:07 +0000 [r322069] Jonathan Rose <jrose@digium.com>
+
+ * main/asterisk.c, include/asterisk/logger.h: Fixes level toggling
+ for logger set levels since it was reversed (closes issue
+ ASTERISK-17850) Reported by: Luke H Tested by: jrose, Luke H
+ Review: https://reviewboard.asterisk.org/r/1244/
+
+2011-06-03 22:09 +0000 [r321812-321926] Richard Mudgett <rmudgett@digium.com>
+
+ * cdr/cdr_radius.c, cel/cel_radius.c: Asterisk crash when unloading
+ cdr_radius/cel_radius. The rc_openlog() API call is passed a
+ string that is used by openlog() to format log messages. The
+ openlog() does not copy the string it just keeps a pointer to it.
+ When the module is unloaded, the string is gone from memory.
+ Depending upon module load order and if the other module then has
+ an error, a crash happens. * Pass rc_openlog() a strdup'd string
+ with the understanding that there will be a small memory leak if
+ the cdr_radius/cel_radius modules are unloaded. * Call
+ rc_destroy() to free the rc handle memory when the module is
+ unloaded. JIRA AST-483 JIRA SWP-3062
+
+ * main/ccss.c: Be more explicit for CCSS generic device state event
+ subscription. Make CCSS generic device state event subscription
+ specify the AST_EVENT_IE_STATE ie exists to be safe.
+
+ * main/event.c, tests/test_event.c: Event subscription fixes. Must
+ commit the subscription fixes together with the integration
+ subscription tests. The subscription fixes cause an erroneously
+ passing test to fail. The new subscription tests detect errors
+ without the subscription fixes. * Added missing event_names[]
+ table entry. * Reworked
+ ast_event_check_subscriber()/match_sub_ie_val_to_event() to
+ correctly detect if a subscriber exists for the proposed event. *
+ Made match_ie_val() and match_sub_ie_val_to_event() check the
+ buffer length for RAW payload types. * Fixed error handling
+ memory leak in ast_event_sub_activate(), ast_event_unsubscribe(),
+ and ast_event_queue(). * Made ast_event_new() and
+ ast_event_check_subscriber() better protect themselves from an
+ invalid payload type. * Added container lock protection between
+ removing old cache events and adding the new cached event in
+ ast_event_queue_and_cache()/event_update_cache(). * Added new
+ event subscription tests.
+
+ * main/event.c, include/asterisk/event.h: Constify subscription
+ description parameter string.
+
+ * channels/chan_iax2.c, channels/chan_sip.c: Correct IAX2 and SIP
+ event subscription description string.
+
+2011-06-03 18:32 +0000 [r321753] Russell Bryant <russell@digium.com>
+
+ * tests/test_astobj2.c: Backport an astobj2 unit test so that it
+ runs on 1.8 as well.
+
+2011-06-03 13:17 +0000 [r321685] Leif Madsen <lmadsen@digium.com>
+
+ * configs/queues.conf.sample: Also document the 'queue-minute'
+ option. (closes issue #19386) Reported by: juanmol
+
+2011-06-01 23:11 +0000 [r321547] Richard Mudgett <rmudgett@digium.com>
+
+ * main/cdr.c: CDR comment tweaks.
+
+2011-06-01 20:10 +0000 [r321537] Brett Bryant <bbryant@digium.com>
+
+ * apps/app_voicemail.c: This patch fixes an issue with using the
+ wrong voicemail folders with greetings. (closes issue #17871)
+ Reported by: edhorton Patches: digium_bug_17871_2 uploaded by
+ fhackenberger (license 592) Tested by: edhorton, fhackenberger
+
+2011-06-01 10:40 +0000 [r321528] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/oochannels.c, addons/chan_ooh323.c,
+ addons/ooh323c/src/ooh245.c: Fix double alerting, add forced
+ alerting before answer Fix double alerting (it wasn't fixed here
+ by issue #18542) Add forced alerting before connect (if it wasn't
+ before) Try to send all packets from outgoing queue rather than
+ one only Call goes into clearing state when disconnect command is
+ received (closes issue #19361) Reported by: vmikhelson Patches:
+ issue19361-3.patch uploaded by may213 (license 454) Tested by:
+ vmikhelson
+
+2011-05-31 20:54 +0000 [r321517] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/dnsmgr.h, include/asterisk/acl.h: Update some
+ comments.
+
+2011-05-31 18:52 +0000 [r321515] David Vossel <dvossel@digium.com>
+
+ * channels/chan_local.c: Chan_local locking cleanup. This patch
+ removes all of the unnecessary deadlock avoidance loops that
+ occur in chan_local. It also resolves an issue with a deadlock
+ triggered by local channel optimizations. (issue #18028) Review:
+ https://reviewboard.asterisk.org/r/1231/
+
+2011-05-31 16:04 +0000 [r321511] Leif Madsen <lmadsen@digium.com>
+
+ * channels/chan_sip.c: Enhance NOTICE message to know who couldn't
+ access the dialplan. (closes issue #19390) Reported by: lmadsen
+ Patches: __20110531-sip-notice-tweak.txt uploaded by lmadsen
+ (license 10) Tested by: russell
+
+2011-05-28 00:27 +0000 [r321337-321436] Richard Mudgett <rmudgett@digium.com>
+
+ * res/res_agi.c: Some hagi launch cleanup. Inspired by issue 19256.
+ This patch would also fix the crash.
+
+ * main/srv.c: Crash when using hagi and no servers are available.
+ When none of the servers returned by the SRV querey respond,
+ asterisk crashes. The problem is that if the loop over all the
+ SRV entries finishes then the srv_context has already been
+ cleaned up. * Make ast_srv_cleanup() check to see if the context
+ is already cleaned up. (closes issue #19256) Reported by:
+ byronclark
+
+ * apps/app_privacy.c: The app_privacy args have undocumented
+ "options" position, interferes with "context" position. * Add
+ documention for unused "options" position to match existing code.
+ (closes issue #19273) Reported by: mdavenport
+
+2011-05-27 21:54 +0000 [r321333-321335] Leif Madsen <lmadsen@digium.com>
+
+ * include/asterisk/frame.h, main/file.c: Fix issue with playback of
+ H.261 video. (closes issue #19379) Reported by: neutrino88
+ Patches: videoprompt.patch uploaded by neutrino88 (license 297)
+ (changes by russell)
+
+ * main/features.c: Allow parking lot hints and musicclass to be
+ set. (closes issue #19378) Reported by: sboily_proformatique
+ Patches: pf_parkinghint_music_fix uploaded by sboily
+ proformatique (license 206) Tested by: russell
+
+2011-05-27 21:31 +0000 [r321330] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_privacy.c: The app_privacy args have undocumented
+ "options" position, interferes with "context" position. * Add
+ documention for unused "options" position to match existing code.
+ The trunk(v1.10) version will remove the unused options position.
+ (closes issue #19273) Reported by: mdavenport
+
+2011-05-27 14:59 +0000 [r321273] Jonathan Rose <jrose@digium.com>
+
+ * channels/sip/reqresp_parser.c: markm committed a patch I was
+ working on yesterday, this fixes it to mesh up with suggestions
+ by mnicholson.
+
+2011-05-27 08:31 +0000 [r321211] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/features.c: Fix *8 directed pickup locks system during
+ pickupsound play out move playout from sip_pickup_thread to
+ bridge using BRIDGE_PLAY_SOUND method, This stop the clash of 2
+ threads trying to write audio to same channel. In addition fixes
+ choppy audio beep in issue 19177. (issue #18654) (issue #19177)
+ Reported by: Docent Patches: review1232-1.88888888 alecdavis
+ (license 585) Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1232/
+
+2011-05-26 21:48 +0000 [r321100-321155] Mark Murawki <markm@intellasoft.net>
+
+ * channels/chan_sip.c, channels/sip/reqresp_parser.c: Fixed build
+ problem with dev mode enabled, which was caused by commit 321100.
+ Reformulated patch to be more generic. Moved the sip uri parse
+ variable initalization to parse_uri_full in reqresp_parser.c.
+ This will ensure that any use of parse uri will have null output
+ variables if the parse fails. (closes issue #19346) Reported by:
+ kobaz Tested by: kobaz,JonathanRose Review: [full review board
+ URL with trailing slash]
+
+ * main/netsock2.c, channels/chan_sip.c: ast_sockaddr_resolve() in
+ netsock2.c may deref a null pointer Added a null check in
+ netsock2 ast_sockaddr_resolve() as well as added default
+ initalizers in chan_sip parse_uri_legacy_check() to make sure
+ that invalid uris will make null (and not undefined)
+ user,pass,domain,transport variables (closes issue #19346)
+ Reported by: kobaz Patches: netsock2.patch uploaded by kobaz
+ (license 834) Tested by: kobaz, Marquis
+
+2011-05-26 18:10 +0000 [r321044] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/netsock2.h: Update ast_sockaddr comment with an
+ important note.
+
+2011-05-26 17:29 +0000 [r321042] Terry Wilson <twilson@digium.com>
+
+ * main/rtp_engine.c: Initialize stack-allocated ast_sockaddrs
+ before use It is important to always initialize ast_sockaddrs
+ before use--even if they are passed to ast_sockaddr_copy as the
+ underlying storage could be bigger than what ends up being
+ copied--leaving part of the data unitialized.
+
+2011-05-26 15:57 +0000 [r320947] Russell Bryant <russell@digium.com>
+
+ * channels/chan_alsa.c, channels/chan_mgcp.c: Remove some variables
+ that were set but unused.
+
+2011-05-25 22:25 +0000 [r320796-320883] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Native SIP CCSS sends bad CC cancel
+ SUBSCRIBE message. The SUBSCRIBE message used to cancel a CC
+ request has incorrect To/From SIP headers. They are reversed and
+ the dialog tags are the same when they should not be. If pedantic
+ mode was disabled, then the cancel would have succeeded despite
+ the incorrect message. * The SIP_OUTGOING flag was not set
+ correctly for the dialog and I had to move some CC subscribe
+ handling code as a result. * Initialized the dialog subscribed
+ type to CALL_COMPLETION earlier. If a CC request SUBSCRIBE
+ message comes in and the CC instance is not found, the 404
+ response was duplicated. JIRA AST-568 JIRA SWP-3493
+
+ * UPGRADE.txt, CHANGES, apps/app_queue.c, apps/app_dial.c,
+ main/channel.c, main/manager.c, apps/app_meetme.c,
+ apps/app_fax.c, main/features.c: The AMI Newstate event contains
+ different information between v1.4 and v1.8. The addition of
+ connected line support in v1.8 changes the behavior of the
+ channel caller ID somewhat. The channel caller ID value no longer
+ time shares with the connected line ID on outgoing call legs. The
+ timing of some AMI events/responses output the connected line ID
+ as caller ID. These party ID's are now separate. * The
+ ConnectedLineNum and ConnectedLineName headers were added to many
+ AMI events/responses if the CallerIDNum/CallerIDName headers were
+ also present. (closes issue #18252) Reported by: gje Tested by:
+ rmudgett Review: https://reviewboard.asterisk.org/r/1227/
+
+ * include/asterisk/channel.h, main/channel.c, main/features.c: Give
+ zombies a safe channel driver to use. Recent crashes from zombie
+ channels suggests that they need a safe home to goto. When a
+ masquerade happens, the physical part of the zombie channel is
+ hungup. The hangup normally sets the channel private pointer to
+ NULL. If someone then blindly does a callback to the channel
+ driver, a crash is likely because the private pointer is NULL.
+ The masquerade now sets the channel technology of zombie channels
+ to the kill channel driver. Related to the following issues:
+ (issue #19116) (issue #19310) Review:
+ https://reviewboard.asterisk.org/r/1224/
+
+2011-05-25 00:49 +0000 [r320716] Terry Wilson <twilson@digium.com>
+
+ * addons/chan_mobile.c: Cast data as char * before using S_OR This
+ is required for compiling successfully under dev mode
+
+2011-05-23 17:53 +0000 [r320650] Richard Mudgett <rmudgett@digium.com>
+
+ * CHANGES, main/manager.c: Add ConnectedLineNum/Name headers to
+ output of AMI action Status. * Add ConnectedLineNum and
+ ConnectedLineName headers to the output of the AMI action Status.
+ This makes it easier to find out who the channel is connected to
+ without having to lookup BridgedChannel or when they are
+ connected to an application (e.g.: VoiceMail) which has no
+ bridged channel. * Bridged channels with no CallerID had ""
+ instead of "<unknown>" output, that might be a bug as "<unknown>"
+ was what older versions used. (closes issue #18158) Reported by:
+ gareth Patches: svn-292308.diff uploaded by gareth (license 208)
+
+2011-05-23 16:19 +0000 [r320573] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * configure, configure.ac: GNU libiconv uses symbol "libiconv_open"
+ instead of "iconv_open". (closes issue #19344) Reported by:
+ rohanl Patches: iconv-check.patch uploaded by rohanl (license
+ 1284)
+
+2011-05-23 16:18 +0000 [r320568] David Vossel <dvossel@digium.com>
+
+ * main/tcptls.c, /: Merged revisions 320562 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r320562 | dvossel | 2011-05-23 11:15:18 -0500 (Mon, 23 May 2011)
+ | 9 lines Adds missing part to the ast_tcptls_server_start fails
+ second attempt to bind patch. (closes issue #19289) Reported by:
+ wdoekes Patches:
+ issue19289_delay_old_address_setting_tcptls_2.patch uploaded by
+ wdoekes (license 717) ........
+
+2011-05-23 15:47 +0000 [r320560] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configure, configure.ac: Don't generate spurious "No: command not
+ found" messages when running the configure script on a system
+ that has neither gmime-config nor pkg-config.
+
+2011-05-23 14:33 +0000 [r320504] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c: Fixes segfault occuring in chan_sip.c at
+ __set_address_from_contact Checks to see if domain contains
+ anything before sending it off to ast_sockaddr_resolve which is
+ where the segfault was occuring due to null str. (closes issue
+ #18857) Reported by: sybasesql Review:
+ https://reviewboard.asterisk.org/r/1225/
+
+2011-05-22 23:34 +0000 [r320445] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * res/res_odbc.c, /: Merged revisions 320444 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r320444 | tilghman | 2011-05-22 18:25:51 -0500 (Sun, 22 May 2011)
+ | 8 lines Don't crash when the connection fails. (closes issue
+ #19250) Reported by: seadweller Patches:
+ 20110514__issue19250.diff.txt uploaded by tilghman (license 14)
+ Tested by: seadweller, sum ........
+
+2011-05-20 21:39 +0000 [r320338] David Vossel <dvossel@digium.com>
+
+ * main/tcptls.c, /: Merged revisions 320271 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r320271 | dvossel | 2011-05-20 16:24:48 -0500 (Fri, 20 May 2011)
+ | 8 lines Fixes issue with ast_tcptls_server_start failing on
+ second attempt to bind. (closes issue #19289) Reported by:
+ wdoekes Patches:
+ issue19289_delay_old_address_setting_tcptls.patch uploaded by
+ wdoekes (license 717) ........
+
+2011-05-20 20:49 +0000 [r320237] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 320236 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r320236 | rmudgett | 2011-05-20 15:44:54 -0500
+ (Fri, 20 May 2011) | 20 lines Merged revisions 320235 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r320235 | rmudgett | 2011-05-20 15:38:22 -0500 (Fri, 20 May 2011)
+ | 13 lines The meetme CLI command completion leaves conferences
+ mutex locked. When issuing a meetme kick CLI command and an
+ invalid (non-existent) conference number is specified, pressing
+ Tab leaves the conferences mutex locked and, therefore, all
+ conferences deadlock. Add missing unlock. (closes issue #19336)
+ Reported by: zvision Patches: app_meetme.diff uploaded by zvision
+ (license 798) ........ ................
+
+2011-05-20 18:48 +0000 [r320180] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: This commit modifies the way polling is done
+ on TLS sockets. Because of the buffering the TLS layer does,
+ polling is unreliable. If poll is called while there is data
+ waiting to be read in the TLS layer but not at the network layer,
+ the messaging processing engine will not proceed until something
+ else writes data to the socket, which may not occur. This change
+ modifies the logic around TLS sockets to only poll after a failed
+ read on a non-blocking socket. This way we know that there is no
+ data waiting to be read from the buffering layer. (closes issue
+ #19182) Reported by: st Patches: ssl-poll-fix3.diff uploaded by
+ mnicholson (license 96) Tested by: mnicholson
+
+2011-05-20 18:12 +0000 [r320162] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_voicemail.c: Fixes an imapfolder related crash
+ imapfolders being set in the general section of voicemail would
+ cause the inbox folder name to change. Since sound file names are
+ made based on the names of the folders, this would cause the
+ audio related to that folder name to change and if Asterisk
+ attempted to play it, the channel would instantly hang up when
+ the audio file couldn't be found. This patch searches for the
+ name of the folder first to leave existing behavior in tact and
+ if that fails, it uses the normal inbox name to get the sound
+ file instead. (closes issue #16104) Reported by: blkline Review:
+ https://reviewboard.asterisk.org/r/1215/
+
+2011-05-20 17:03 +0000 [r319997-320059] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: Misc comment cleanup in features.c.
+
+ * main/channel.c, main/features.c: Crash while transferring a call
+ during DTMF feature timeout. When a call is being attended
+ transferred during the time between AST_FRAME_DTMF_BEGIN and
+ AST_FRAME_DTMF_END, the transferred channel becomes a zombie (so
+ tech data is not available), making ast_dtmf_stream() segfault
+ when it tries to send the DTMF digit (at least with SIP
+ channels). Patch based on feature-end-zombie.patch uploaded by
+ Irontec (license 1256) * Check for zombies when
+ ast_channel_bridge() returns. * Guarantee that the fo parameter
+ value is initialized in ast_channel_bridge() before any returns.
+ (closes issue #19116) Reported by: Irontec Tested by: rmudgett
+
+ * apps/app_directed_pickup.c, main/features.c: Change some variable
+ names to make pickup code easier to understand.
+
+ * apps/app_directed_pickup.c, main/features.c: Crash when using
+ directed pickup applications. The directed pickup applications
+ can cause a crash if the pickup was successful because the
+ dialplan keeps executing. This patch does the following: *
+ Completes the channel masquerade on a successful pickup before
+ the application returns. The channel is now guaranteed a zombie
+ and must not continue executing the dialplan. * Changes the
+ return value of the directed pickup applications to return zero
+ if the pickup failed and nonzero(-1) if the pickup succeeded. *
+ Made some code optimizations that no longer require re-checking
+ the pickup channel to see if it is still available to pickup.
+ (closes issue #19310) Reported by: remiq Patches:
+ issue19310_v1.8_v2.patch uploaded by rmudgett (license 664)
+ Tested by: alecdavis, remiq, rmudgett Review:
+ https://reviewboard.asterisk.org/r/1221/
+
+2011-05-20 13:28 +0000 [r319938] Jonathan Rose <jrose@digium.com>
+
+ * configs/sip.conf.sample, channels/sip/include/sip.h,
+ channels/chan_sip.c: Adds legacy_useroption_parsing to address
+ interoperability concerns. With the new option engaged, Asterisk
+ should interpret user fields with useroptions contained within
+ the userfield of the uri by stripping them out of the original
+ message whenever a semicolon is encountered in the userfield
+ string. (closes issue #18344) Reported by: danimal Tested by:
+ jrose Review: https://reviewboard.asterisk.org/r/1223/
+
+2011-05-19 23:28 +0000 [r319920] Terry Wilson <twilson@digium.com>
+
+ * main/bridging.c, include/asterisk/bridging_technology.h,
+ include/asterisk/bridging.h: Revert part of a change to the
+ bridging API code The capabilities used in the bridging API are
+ very different than the ones used for formats. When the
+ conversion was made expanding the bit width of codecs, the
+ bridging code was accidentally accosted in ways that it didn't
+ deserve.
+
+2011-05-19 18:32 +0000 [r319866] Jonathan Rose <jrose@digium.com>
+
+ * main/features.c: Fix Randomize option on Park() The randomize
+ option was generally not working like it should have at all on
+ Park(). This patch restores intended functionality. (closes issue
+ #18862) Reported by: davidw Tested by: jrose Review:
+ https://reviewboard.asterisk.org/r/1222/
+
+2011-05-19 17:59 +0000 [r319812] Mark Murawki <markm@intellasoft.net>
+
+ * cel/cel_odbc.c: In cel_odbc, an uninitialized RWLIST is attempted
+ to be locked. Added INIT and DESTROY for the RWLIST odbc_tables
+ (closes issue #19331) Reported by: kobaz Patches: odbc_cel.patch
+ uploaded by kobaz (license 834)
+
+2011-05-19 16:50 +0000 [r319758] Richard Mudgett <rmudgett@digium.com>
+
+ * main/ccss.c: CCSS generic agent with POTS and ISDN phones fail
+ caller busy call-back test. If the following is true after a CCSS
+ activation: * The generic agent is for an analog phone or ISDN
+ phone. (Caller party) * The called party becomes available. * The
+ caller party is not available. When the caller party becomes
+ available, the caller is not alerted to the called party being
+ available. The generic agent still thinks the caller is busy. *
+ Fixed the generic agent device state event subscription to look
+ for all device states that are considered available. *
+ Encapsulated the device state test for CCSS generic device
+ available in cc_generic_is_device_available(). Made the generic
+ agent and monitor use the new function instead of the manually
+ coded inline equivalent. JIRA AST-559 JIRA SWP-3462
+
+2011-05-18 23:15 +0000 [r319529-319654] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 319653 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r319653 | twilson | 2011-05-18 16:11:57 -0700
+ (Wed, 18 May 2011) | 15 lines Merged revisions 319652 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r319652 | twilson | 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011)
+ | 8 lines Make sure everyone gets an unhold when a transfer
+ succeeds Some phones, like the Snom phones, send a hold to the
+ transfer target after before sending the REFER. We need to make
+ sure that we unhold the parties that are being connected after
+ the masquerade. If Local channels with the /nm option are used
+ when dialing the parties, hold music would still be playing on
+ the transfer target, even after being connected with the
+ transferee. ........ ................
+
+ * channels/chan_sip.c: Unbreak the storing of registrations for
+ restart The fix for issue 18882 broke retrieving non-realtime
+ peers from the ast_db on restart/reload. This patch tries to
+ unbreak things while leaving the intent of the original fix
+ intact. (closes issue #19318) Reported by: remiq Patches:
+ diff.txt uploaded by twilson (license 396) Tested by: lmadsen,
+ remiq
+
+ * apps/app_dial.c, /: Merged revisions 319528 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r319528 | twilson | 2011-05-18 13:02:06 -0700
+ (Wed, 18 May 2011) | 17 lines Merged revisions 319527 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r319527 | twilson | 2011-05-18 12:56:08 -0700 (Wed, 18 May 2011)
+ | 10 lines Fix app_dial ring groups Revert part of r315643. We
+ need to remove the datastore here as well. The code in bridging
+ code will catch anything that app_dial might miss. (closes issue
+ #19311) Reported by: mspuhler Patches: issue_19311_no_answer.diff
+ uploaded by elguero (license 37) ........ ................
+
+2011-05-17 21:57 +0000 [r319469] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c: Merged revision 319468 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r319468 | rmudgett | 2011-05-17 16:49:31 -0500 (Tue,
+ 17 May 2011) | 15 lines The mISDN HDLC mode is prevented on
+ dialed channels. The use of mISDN HDLC mode is prevented if the
+ mISDN dial technology option 'h1' is used when config option
+ astdtmf=yes. There is a bug in channels/misdn/isdn_lib.c which
+ prevents the use of HDLC mode. Instead of setting the channel to
+ HDLC mode it is set to transparent(no dsp, no hdlc), although
+ hdlc is not "no hdlc". I.e the logging message is correct, but
+ the if condition is not. Make check the nodsp and hdlc flags.
+ JIRA ABE-2787 JIRA SWP-3437 ..........
+
+2011-05-17 12:53 +0000 [r319365-319367] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_voicemail.c: Don't create [general] voicemail context
+ when using users.conf Prior to this patch, app_voicemail would
+ create a [general] context when parsing users.conf. (closes issue
+ #18891) Reported by: pdugas Patches:
+ app_voicemail-ignore-general.patch uploaded by pdugas (license
+ 1222) app_voicemail-ignore-general-style-guidelines.patch
+ uploaded by seanbright (license 71) Tested by: pdugas
+
+ * contrib/init.d/rc.debian.asterisk: Make Debian init script lsb
+ compliant (closes issue #18896) Reported by: manwe Patches:
+ debian_init_lsb.patch uploaded by manwe (license 1223)
+
+2011-05-16 21:00 +0000 [r319261] Jonathan Rose <jrose@digium.com>
+
+ * main/dsp.c: Makes busy detection in dsp.c always allow for at
+ least one frame (20ms) of error so that 200ms tone lengths don't
+ get ignored by single frame error lengths.
+
+2011-05-16 20:33 +0000 [r319259] Richard Mudgett <rmudgett@digium.com>
+
+ * main/ccss.c: Deadlock between generic CCSS agent and native ISDN
+ CCSS. Deadlock can occur when the generic CCSS agent is deleting
+ duplicate CC offers and the native ISDN CC driver is processing
+ an incoming CC message. The cc_core_instances container lock
+ cannot be held when an agent or monitor callback is invoked
+ without the possibility of a deadlock. * Make
+ kill_duplicate_offers() remove the reference in cc_core_instances
+ outside of the container lock. JIRA AST-566 JIRA SWP-3469
+
+2011-05-16 18:17 +0000 [r319204] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 319202 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r319202 | twilson | 2011-05-16 11:00:21 -0700 (Mon, 16 May 2011)
+ | 4 lines Unlink a peer from peers_by_ip when expiring a
+ registration Review: https://reviewboard.asterisk.org/r/1218/
+ ........
+
+2011-05-16 15:57 +0000 [r319145] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 319144 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r319144 | dvossel | 2011-05-16 10:56:16 -0500 (Mon, 16 May 2011)
+ | 2 lines Fixes issue with peer ref-counting during
+ handle_request_subscribe. (closes issue #19293) Reported by:
+ irroot ........
+
+2011-05-16 15:53 +0000 [r319142] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Make sure tcptls_session exists before
+ dereferencing it. (closes issue #19192) Reported by: stknob
+ Patches: 10-tcptls-unreachable-peer-segfault.patch uploaded by
+ Chainsaw (license 723) Tested by: vois, Chainsaw
+
+2011-05-16 14:35 +0000 [r319085] Paul Belanger <pabelanger@digium.com>
+
+ * res/res_http_post.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Support gmime-2.4 (closes issue #18863) Reported
+ by: tzafrir Patches: gmime-2.4-18.diff uploaded by tzafrir
+ (license 46) Tested by: tzafrir Review:
+ https://reviewboard.asterisk.org/r/1213/
+
+2011-05-16 14:26 +0000 [r319083] David Vossel <dvossel@digium.com>
+
+ * formats/format_wav.c: Fixes Big Endian build issue. (closes issue
+ #19298) Reported by: tzafrir
+
+2011-05-13 18:09 +0000 [r318917-318921] Brett Bryant <bbryant@digium.com>
+
+ * main/channel.c: Fixes a segmentation fault in dynamic hints when
+ a channel technology isn't loaded for a hint. (closes issue
+ #18495) Reported by: bertrand Tested by: bertrand
+
+ * res/res_srtp.c: This patch fixes an issue with SRTP which makes
+ HOLD/UNHOLD impossible when too much time has passed between
+ sending audio. (closes issue #18206) Reported by: bernhardsi
+ Patches: res_srtp_unhold.patch uploaded by bernhards (license
+ 1138) Tested by: bernhards, notthematrix
+
+ * channels/chan_sip.c: This patch allows TCP peers into the ast_db
+ where they were previously restricted. (closes issue #18882)
+ Reported by: cmaj Patches:
+ patch-chan_sip-1.8.3-rc2-allow-tcp-peer-store-db-and-readonly-rt-backend.diff.txt
+ uploaded by cmaj (license 830) Tested by: cmaj
+
+2011-05-13 16:28 +0000 [r318783-318868] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: CDR's are being written immediately on caller
+ hangup. CDR's are being written immediately on caller hangup. The
+ dialplan is not able to modify it in the h exten. The h exten in
+ the initial context is not run before closing CDR's when the
+ bridge is unlinked if a macro is active and does not have an h
+ exten. * Make ast_bridge_call() check for an h exten in the
+ current context and if a macro is active then the initial
+ context. The first h exten found is then run before closing the
+ CDR. (closes issue #18212) Reported by: leearcher Patches:
+ issue18212_v1.8.patch uploaded by rmudgett (license 664) Tested
+ by: rmudgett Review: https://reviewboard.asterisk.org/r/1206/
+
+ * channels/sig_pri.c: PRI early media won't ring. And another way
+ to pass early media. Don't indicate that there is inband
+ information present, just assume that the B channel is connected.
+ * Restore clearing the dialing flag Rx squelch unconditionally
+ when a PROCEEDING message comes in. (closes issue #19268)
+ Reported by: tbsky Patches: issue19268_v1.8.patch uploaded by
+ rmudgett (license 664) Tested by: tbsky
+
+2011-05-12 23:35 +0000 [r318720] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/sip/reqresp_parser.c: Handle ipv6 addresses in the
+ sent-by Via: field. This change fixes a regression in via header
+ parsing and ipv6 handling. (closes issue #18951)
+
+2011-05-12 22:52 +0000 [r318671] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * include/asterisk/features.h, channels/chan_sip.c,
+ apps/app_directed_pickup.c, main/features.c: Fix directed group
+ pickup feature code *8 with pickupsounds enabled Since 1.6.2, the
+ new pickupsound and pickupfailsound in features.conf cause many
+ issues. 1). chan_sip:handle_request_invite() shouldn't be playing
+ out the fail/success audio, as it has 'netlock' locked. 2).
+ dialplan applications for directed_pickups shouldn't beep. 3).
+ feature code for directed pickup should beep on success/failure
+ if configured. Created a sip_pickup() thread to handle the pickup
+ and playout the audio, spawned from handle_request_invite. Moved
+ app_directed:pickup_do() to features:ast_do_pickup(). Functions
+ below, all now use the new ast_do_pickup() app_directed_pickup.c:
+ pickup_by_channel() pickup_by_exten() pickup_by_mark()
+ pickup_by_part() features.c: ast_pickup_call() (closes issue
+ #18654) Reported by: Docent Patches:
+ ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license
+ 585) Tested by: lmadsen, francesco_r, amilcar, isis242,
+ alecdavis, irroot, rymkus, loloski, rmudgett Review:
+ https://reviewboard.asterisk.org/r/1185/
+
+2011-05-11 18:47 +0000 [r318549-318550] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Comment out the REF_DEBUG that slipped in
+ during debugging
+
+ * /, channels/chan_sip.c: Merged revisions 318548 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r318548 | twilson | 2011-05-11 12:15:39 -0500 (Wed, 11 May 2011)
+ | 19 lines Clean up several chan_sip reference leaks Several
+ situations in the code could lead to peers or sip_pvt references
+ being leaked. This would cause RTP ports to never be destroyed
+ (leading to exhaustion of all available RTP ports) and memory
+ leaks. The original patch for this issue from rgagnon was the
+ result of an obscene amount of testing and hard work, for which I
+ am very grateful. I did some cleanup and added a few additional
+ refcount fixes that I found. (closes issue #17255) Reported by:
+ kvveltho Patches: tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff
+ uploaded by rgagnon (license 1202) Tested by: rgagnon, twilson,
+ wdoekes, loloski Review: https://reviewboard.asterisk.org/r/1101/
+ Review: https://reviewboard.asterisk.org/r/1207/ Review:
+ https://reviewboard.asterisk.org/r/1210/ ........
+
+2011-05-10 23:41 +0000 [r318499] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c, channels/sig_ss7.c: Unable to pickup
+ DAHDI/PRI call because call state is reported as DIALING. The
+ channel state is not updated to RINGING when an ALERTING message
+ is received. Regression caused when sig_pri.c (also sig_ss7.c)
+ extracted from chan_dahdi.c. * Added missing channel state update
+ to RINGING when the AST_CONTROL_RINGING frame is queued for ISDN
+ and SS7. (closes issue #19257) Reported by: alecdavis Patches:
+ issue19257_v1.8_v2.patch uploaded by rmudgett (license 664)
+ Tested by: alecdavis, rmudgett
+
+2011-05-10 18:46 +0000 [r318485] Leif Madsen <lmadsen@digium.com>
+
+ * main/manager.c: Filter out blacklisted manager events when using
+ eventfilter. Merging change from trunk in revision 306432.
+ (closes issue #19260) Reported by: dhubbard Tested by: dhubbard
+
+2011-05-10 15:13 +0000 [r318436] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: chan_iax2: change LOG_NOTICE to LOG_DEBUG
+ in iax2_read().
+
+2011-05-09 23:15 +0000 [r318351] Richard Mudgett <rmudgett@digium.com>
+
+ * res/Makefile, res/res_features.exports.in (removed): Remove
+ references to res_features and its export file. The contents of
+ res/res_features.c was moved to into main/features.c awhile ago.
+ There is no longer any need for the res/Makefile to reference
+ res_features or the res_features linker exports file to exist.
+
+2011-05-09 20:23 +0000 [r318337] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 318331 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r318331 | twilson | 2011-05-09 15:04:41 -0500 (Mon, 09 May 2011)
+ | 12 lines Don't offer video to directmedia callee unless caller
+ offered it as well Make sure that when directmedia is enabled,
+ that video is not offered to the callee even if it supports it.
+ p->vrtp will not exist since the caller didn't offer video.
+ (closes issue #19195) Reported by: one47 Patches:
+ sip_cant_add_video_rtp uploaded by one47 (license 23) ........
+
+2011-05-09 19:07 +0000 [r318282] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: Hangup extension executed twice. When a user
+ hangs up a call, in certain circumstances, the hangup extension
+ can end up being executed twice: 1) If a call is bridged and the
+ 'h' extension executes the Hangup application, then the 'h'
+ extension will be executed twice. 2) If a call is bridged within
+ a macro (Dial or Queue), it has its own 'h' extension, the main
+ context also has an 'h' extension, and the macro 'h' extension
+ executes the Hangup application, then both 'h' extensions will be
+ executed. * Revert originally commited fix for #16106 and just
+ set AST_FLAG_BRIDGE_HANGUP_RUN unconditionally in
+ ast_bridge_call(). The bridge code just executed an 'h' extension
+ so the main PBX loop does not need to execute one as well. (issue
+ #16106) Reported by: ajohnson (issue #16548) Reported by: hajekd
+
+2011-05-09 17:09 +0000 [r318233] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 318230 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r318230 | dvossel | 2011-05-09 11:51:45 -0500 (Mon, 09 May 2011)
+ | 7 lines Fixes cases where sip_set_rtp_peer can return too early
+ during media path reset. (closes issue #19225) Reported by: one47
+ Patches: sip_set_rtp_peer.patch uploaded by one47 (license 23)
+ ........
+
+2011-05-09 16:57 +0000 [r318231] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Don't get early media for ISDN on outgoing
+ calls. It looks to be a long-standing misinterpretation of the
+ progress indicator ie values: 1 - Call is not end-to-end ISDN;
+ further call progress information may be available in-band. 8 -
+ In-band information or an appropriate pattern is now available.
+ Only value 8 is handled by chan_dahdi/sig_pri. The 1 value is not
+ handled as early media probably because the meaning of the second
+ half of it's description was overlooked. * Test to see if either
+ PRI_PROG_CALL_NOT_E2E_ISDN(1) or PRI_PROG_INBAND_AVAILABLE(8)
+ bits are set to open the media path. (closes issue #18868)
+ Reported by: isrl Patches: issue18868_19246_v1.8.patch uploaded
+ by rmudgett (license 664) Tested by: satish_lx .......... No
+ inband progress on PRI_EVENT_RINGING even if inband flag set. My
+ ISDN-PRI provider sends an ALERTING with "Inband information or
+ appropriate pattern now available", but Asterisk only generates
+ and passes the RING to the SIP extension, not the inband message.
+ Unfortunately, the inband message is not a ringback tone but a
+ prompt that says the number is not in service. The SIP extension
+ then hears two rings and the call is hungup which confuses the
+ caller. * Post an AST_CONTROL_PROGRESS as well as opening the
+ media path if inband audio is indicated with an ALERTING message.
+ (closes issue #19246) Reported by: cristiandimache Patches:
+ issue19246_v1.8.patch uploaded by rmudgett (license 664) Tested
+ by: cristiandimache
+
+2011-05-09 14:18 +0000 [r318148] Jonathan Rose <jrose@digium.com>
+
+ * configs/features.conf.sample: Documenting an observed behavior of
+ features in features.conf. Since parkinglots use an integer for
+ the parkinglot extensions, leading zeros specified in the
+ configuration file are ignored.
+
+2011-05-09 14:09 +0000 [r318142] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c: Make indicate/control frames WRITE events on
+ framehooks. Also, if a framehook returns a non-control frame,
+ don't forward it to the channel. (closes issue #19251) Reported
+ by: irroot Patches: (modified) framehook_indicate.patch2 uploaded
+ by irroot (license 52) Tested by: irroot
+
+2011-05-07 23:35 +0000 [r318055-318057] Russell Bryant <russell@digium.com>
+
+ * res/res_config_curl.c: res_config_curl: fix a crash with static
+ realtime. (closes issue #18413) Reported by: jmls Patches:
+ 20101202__issue18413.diff.txt uploaded by tilghman (license 14)
+ Tested by: jmls
+
+ * channels/chan_iax2.c: chan_iax2: Don't overwrite port found with
+ an SRV lookup. (closes issue #17291) Reported by: jcovert
+ Patches: chan_iax2.c.1.8.3-srvlookup-corrected.patch uploaded by
+ jcovert (license 551)
+
+2011-05-06 21:49 +0000 [r317967-317969] Russell Bryant <russell@digium.com>
+
+ * apps/app_meetme.c: Use the right variable to print the time in a
+ debug message. The original patch also increased some buffer
+ sizes, but that was already done in this version. (closes issue
+ #17034) Reported by: sysreq Patches: asterisk-issue-17034.patch
+ uploaded by sysreq (license 1009)
+
+ * apps/app_meetme.c: Fix some more "set but unused" compiler
+ warnings.
+
+2011-05-06 21:06 +0000 [r317918] David Vossel <dvossel@digium.com>
+
+ * res/res_rtp_asterisk.c: Fixes missing colon from To/From headers
+ in RTCP manager events. (closes issue #18221) Reported by:
+ clegall_proformatique Patches: 18221_1.patch uploaded by ebroad
+ (license 878)
+
+2011-05-06 21:06 +0000 [r317861-317917] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c: Fix calculation of free RAM to make minmemfree option
+ work. (closes issue #17124) Reported by: loic Patches: pbx_c.diff
+ uploaded by loic (license 1020)
+
+ * channels/chan_sip.c: chan_sip: Destroy variables on a sip_pvt
+ before copying vars from the sip_peer. Don't duplicate variables
+ on the sip_pvt. Just reset the variable list each time. (closes
+ issue #19202) Reported by: wdoekes Patches:
+ issue19202_destroy_challenged_invite_chanvars.patch uploaded by
+ wdoekes (license 717)
+
+ * channels/chan_sip.c: chan_sip: fix a deadlock in
+ check_rtp_timeout. Don't block doing silly deadlock avoidance.
+ Just return and try again later. The funciton gets called often
+ enough that it's fine. Also, this change was already made in
+ trunk. (closes issue #18791) Reported by: irroot Patches:
+ chan_sip.rtptimeout.patch uploaded by irroot (license 52)
+
+ * channels/chan_sip.c: URI encode less characters in the RPID and
+ Contact headers. If this change causes any problems, we will need
+ to backport the more extensive uri encoding and decoding handling
+ changes that are in trunk/1.10. (closes issue #18686) Reported
+ by: wolfgang Patches: quick-and-dirty.patch uploaded by wdoekes
+ (license 717) Tested by: wdoekes, devellow, wolfgang, mav3rick
+
+2011-05-06 19:31 +0000 [r317858] Matthew Nicholson <mnicholson@digium.com>
+
+ * pbx/pbx_lua.c: pbx_lua autoservice fixes Don't start an
+ autoservice in pbx_lua if pbx_lua already started one and don't
+ stop one if we didn't start one. Also start and stop the
+ autoservice when transferring control from and to the pbx.
+
+2011-05-06 19:24 +0000 [r317805-317837] Russell Bryant <russell@digium.com>
+
+ * addons/app_mysql.c: Fix a crash in the MySQL() application. This
+ code was not handling channel datastores safely. The channel must
+ be locked. (closes issue #17964) Reported by: wuwu Patches:
+ issue17964_addon_1.6.2_svn.patch uploaded by seanbright (license
+ 71) Tested by: wuwu
+
+ * contrib/realtime/mysql/sipfriends.sql: Add a new sipfriends.sql
+ for MySQL that has more fields in it. (closes issue #16399)
+ Reported by: pabelanger Patches: sipfriends.sql.v3 uploaded by
+ pabelanger (license 224)
+
+2011-05-06 16:19 +0000 [r317670] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Fix SIP connected line updates. This patch
+ fixes a couple SIP connected line update problems: 1) The
+ connected line needs to be updated when the initial INVITE is
+ sent if there is a peer callerid configured. Previously, the
+ connected line information did not get reported until the call
+ was connected so SIP could not report connected line information
+ in ringing or progress messages. 2) The connected line should not
+ be updated on initial connect if there is no connected line
+ information. Previously, all it did was wipe out any default
+ preset CONNECTEDLINE information set by the dialplan with empty
+ strings. (closes issue #18367) Reported by: GeorgeKonopacki
+ Patches: issue18367_v1.8.patch uploaded by rmudgett (license 664)
+ Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1199/
+
+2011-05-06 08:18 +0000 [r317584] Terry Wilson <twilson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 317575 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r317575 | twilson | 2011-05-06 01:04:17 -0700
+ (Fri, 06 May 2011) | 13 lines Merged revisions 317574 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r317574 | twilson | 2011-05-06 00:55:21 -0700 (Fri, 06 May 2011)
+ | 6 lines Re-fix queue round-robin This part of the change for
+ r315596 was incorrect. No bridge occurs when doing a roundrobin
+ dial and no one answers, so this code shouldn't have been
+ removed. ........ ................
+
+2011-05-05 23:46 +0000 [r317425-317530] Russell Bryant <russell@digium.com>
+
+ * Makefile: If the configure script runs, force a rebuild of
+ menuselect-tree. Some contents in the menuselect tree are
+ dependent on configure script parameters, namely
+ --enable-dev-mode. (closes issue #17219) Reported by: Nick_Lewis
+ Patches: issue_17219.rev1.txt uploaded by russell (license 2)
+
+ * contrib/realtime/mysql/queue_log.sql,
+ contrib/realtime/mysql/sipfriends.sql: Fix some more realtime
+ MySQL schema issues. (closes issue #18537) Reported by: denzs
+ Patches: sipfriends.sql.svndiff uploaded by denzs (license 1182)
+ queue_log.sql.svndiff uploaded by denzs (license 1182)
+ meetme.sql.svndiff uploaded by denzs (license 1182)
+
+ * contrib/realtime/mysql/sipfriends.sql,
+ contrib/realtime/mysql/meetme.sql: Fix some errors in sample
+ MySQL realtime schema files. (closes issue #18915) Reported by:
+ Dovid Patches: sipfriends.patch uploaded by Dovid (license 652)
+ meetme.patch uploaded by Dovid (license 652)
+
+ * cdr/cdr_syslog.c: Don't lose cdr_syslog config on a reload.
+ (closes issue #18679) Reported by: enegaard Patches:
+ issue18679_seanbright.patch uploaded by seanbright (license 71)
+ Tested by: enegaard
+
+ * channels/chan_alsa.c, channels/chan_console.c,
+ channels/chan_oss.c, channels/chan_mgcp.c,
+ channels/misdn_config.c, channels/chan_unistim.c,
+ channels/chan_usbradio.c, channels/chan_dahdi.c,
+ channels/chan_sip.c, channels/chan_skinny.c,
+ channels/chan_h323.c: Fix some consistency issues with
+ jitterbuffer config. Store the defaults noted in the sample
+ config files in the jitterbuffer config data structure. This
+ makes the CLI commands that output these settings show the right
+ thing. Also only show the settings that are relevant in the
+ settings CLI commands, based on which jitterbuffer is selected
+ and whether it's enabled. (closes issue #19083) Reported by:
+ rgagnon Patches: issue-19083-trunk-r313139.diff uploaded by
+ rgagnon (license 1202)
+
+ * pbx/pbx_lua.c: Add a datastore fixup to fix a pbx_lua crash.
+ (closes issue #19055) Reported by: jamhed Patches:
+ lua_datastore_fixup1.diff uploaded by mnicholson (license 96)
+ Tested by: mnicholson, jamhed
+
+ * channels/iax2-provision.c, pbx/pbx_dundi.c,
+ channels/chan_console.c, cdr/cdr_radius.c, channels/chan_iax2.c,
+ res/res_jabber.c, res/res_config_sqlite.c, cel/cel_pgsql.c,
+ channels/chan_jingle.c, channels/sip/sdp_crypto.c,
+ res/res_config_odbc.c, channels/chan_sip.c, res/res_crypto.c,
+ pbx/pbx_lua.c: Fix more "set but unused" warnings.
+
+ * main/dsp.c: Only display inband DTMF warning if inband DTMF
+ detection is enabled. (closes issue #18901) Reported by: irroot
+
+ * apps/app_rpt.c: Fix potential memory leak, and use of
+ uninitialized memory. (closes issue #16476) Reported by: junky
+ Patches: M16476.diff uploaded by junky (license 177)
+
+ * main/manager.c: Add missing ActioID handling to Events action.
+ (closes issue #18949) Reported by: edersohe Patches:
+ 0018949.patch uploaded by edersohe (license 1228)
+
+2011-05-05 20:25 +0000 [r317370] Sean Bright <sean@malleable.com>
+
+ * addons/res_config_mysql.c: Don't duplicate our data on the stack
+ and just use the MYSQL_ROW directly. With large result sets we
+ were blowing out the stack. (closes issue #19090) Reported by:
+ mickecarlsson Patches: issue19090_trunk_svn.patch uploaded by
+ seanbright (license 71) Tested by: mickecarlsson
+
+2011-05-05 19:55 +0000 [r317336] Russell Bryant <russell@digium.com>
+
+ * apps/app_queue.c: Increase buffer size to be PATH_MAX for a path.
+ (closes issue #19239) Reported by: byronclark Patches:
+ queue_announce_length.patch uploaded by byronclark (license 1200)
+
+2011-05-05 19:09 +0000 [r317283] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c: Resolves a deadlock that occurs during
+ sip_new This is based on an uncommitted patch by jpeeler for the
+ issue. Instead of relocking and then unlocking the channel
+ though, we keep the lock on the channel until we are finished
+ doing what we need to the channel. (closes issue #18441) Reported
+ by: Alric
+
+2011-05-05 18:39 +0000 [r317280-317281] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 317255 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r317255 | russell | 2011-05-05 13:29:53 -0500
+ (Thu, 05 May 2011) | 22 lines Merged revisions 317211 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r317211 | russell | 2011-05-05 13:20:29 -0500 (Thu, 05 May 2011)
+ | 15 lines chan_sip: fix broken realtime peer count, fix memory
+ leak This patch addresses two bugs in chan_sip: 1) The count of
+ realtime peers and users was off. The increment checked the value
+ of the caching option, while the decrement did not. 2) Add a
+ missing regfree() for a regex. (closes issue #19108) Reported by:
+ vrban Patches: missing_regfree.patch uploaded by vrban (license
+ 756) sip_object_counter.patch uploaded by vrban (license 756)
+ ........ ................
+
+ * /: Restore branch-1.6.2-merged and branch-1.6.2-blocked
+ properties.
+
+2011-05-05 18:02 +0000 [r317196] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Set SO_KEEPALIVE on SIP TCP sockets so that
+ they eventually go away when a peer abruptly disappears. This
+ mostly occurs after a successful registration. (closes issue
+ #17544) Reported by: marcelloceschia Patches: (modified)
+ tcptls.patch uploaded by st (license 907)
+
+2011-05-05 15:04 +0000 [r317058-317104] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/safe_asterisk, /: Merged revisions 317102 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r317102 | lmadsen | 2011-05-05 10:54:46 -0400 (Thu, 05 May 2011)
+ | 8 lines Disable console colourization inside safe_asterisk
+ checks. (closes issue #19213) Reported by: lefoyer Patches:
+ issue19213_strip_color_in_safe_asterisk-svn.patch uploaded by
+ wdoekes (license 717) Tested by: wdoekes, lefoyer ........
+
+ * Makefile, configs/cel.conf.sample: Remove unused directory and
+ clear up some documentation. (closes issue #19193) Reported by:
+ bchia Patches: cel-csv.diff uploaded by lathama (license 1028)
+ Tested by: lathama, Marquis42
+
+2011-05-05 02:30 +0000 [r316917-316919] Sean Bright <sean@malleable.com>
+
+ * main/http.c: Use the correct HTTP method when generating our
+ digest, otherwise we always fail. When calculating the 'A2'
+ portion of our digest for verification, we need the HTTP method
+ that is currently in use. Unfortunately our mapping function was
+ incorrect, resulting in invalid hashes being generated and, in
+ turn, failures in authentication. (closes issue #18598) Reported
+ by: ksn
+
+ * main/utils.c: Look at the correct buffer for our digest info
+ instead of an empty one. (issue #18598) Reported by: ksn
+
+ * main/manager.c: Make sure that tcptls_session is properly
+ initialized. (issue #18598) Reported by: ksn
+
+2011-05-04 20:50 +0000 [r316874] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooSocket.c: Fix trivial bug in ooSocket.c
+ codes Revert condition for result code of ast_gethostbyname
+ (closes issue #19185) Reported by: dswartz Patches:
+ issue19185-patch uploaded by may213 (license 454)
+
+2011-05-04 18:51 +0000 [r316831] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_meetme.c: Wait for leader with Music On Hold allows
+ crosstalk between participants. Parenthesis in the wrong
+ position. Regression from issue #14365 when expanding conference
+ flags to use 64 bits. (closes issue #18418) Reported by: MrHanMan
+ Tested by: rmudgett
+
+2011-05-04 16:15 +0000 [r316663-316709] Sean Bright <sean@malleable.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 316708 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r316708 | seanbright | 2011-05-04 12:10:59 -0400
+ (Wed, 04 May 2011) | 15 lines Merged revisions 316707 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r316707 | seanbright | 2011-05-04 12:08:50 -0400 (Wed, 04 May
+ 2011) | 8 lines If sox fails when processing a voicemail, don't
+ delete the original file. (closes issue #18111) Reported by:
+ sysreq Patches: issue18111_trunk.patch uploaded by seanbright
+ (license 71) Tested by: seanbright ........ ................
+
+ * main/manager.c: Only return a single error via AMI when
+ requesting a forbidden action. (closes issue #19216) Reported by:
+ oej Patches: issue19216-1.8-r316204.patch uploaded by seanbright
+ (license 71) Tested by: seanbright
+
+2011-05-04 14:25 +0000 [r316617-316650] David Vossel <dvossel@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 316644 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r316644 | dvossel | 2011-05-04 09:23:39 -0500 (Wed, 04 May 2011)
+ | 9 lines Fixes one-way-audio when chanspy activated with the 'o'
+ option (closes issue #18382) Reported by: jkister Patches:
+ 0001-Bugfix-18382-one-way-audio-when-chanspy-activated.patch.txt
+ uploaded by malin (license ) Tested by: firstsip, Greenlightcrm,
+ malin, wdoekes, boroda, dvossel ........
+
+ * /, channels/chan_sip.c: Merged revisions 316616 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r316616 | dvossel | 2011-05-04 08:40:41 -0500 (Wed, 04 May 2011)
+ | 12 lines Fixes session-timers=refuse not being enforced for
+ *caller* During handle_request_invite, the session timer mode was
+ retrieved from a cached variable. This patch forces a peer lookup
+ of the session timer mode in the case of an incoming invite.
+ (closes issue #18804) Reported by: wdoekes Patches:
+ issue18804_session_timer_refuse_caller.patch uploaded by wdoekes
+ (license 717) issue_18804_v2.diff uploaded by dvossel (license
+ 671) ........
+
+2011-05-04 02:34 +0000 [r316476] Sean Bright <sean@malleable.com>
+
+ * /, apps/app_meetme.c: Merged revisions 316475 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r316475 | seanbright | 2011-05-03 22:23:01 -0400 (Tue, 03 May
+ 2011) | 10 lines Honor the C option to MeetMe when L is passed.
+ This fixes a case that r304773 and friends missed. (closes issue
+ #17317) Reported by: var Patches: meetme-continue-on-l_16218.diff
+ uploaded by var (license 1227) Tested by: seanbright ........
+
+2011-05-04 00:12 +0000 [r316429] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * addons/cdr_mysql.c, addons/res_config_mysql.c: Escape column
+ names in case they contain illegal characters ('-') or reserved
+ words. (closes issue #19063) Reported by: festr Patches: patch
+ uploaded by festr (license 443)
+
+2011-05-03 22:13 +0000 [r316336] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_dundi.c, channels/chan_mgcp.c, channels/chan_skinny.c:
+ Use htons() instead of ntohs() in some places. (closes issue
+ #19200) Reported by: wdoekes Patches: issue19200-trunk.patch
+ uploaded by wdoekes (license 717) issue19200-1.8.x.patch uploaded
+ by wdoekes (license 717)
+
+2011-05-03 22:05 +0000 [r316334] David Vossel <dvossel@digium.com>
+
+ * main/channel.c: Fixes framehook segfault on indicate (closes
+ issue #19215) Reported by: irroot Patches:
+ framehook_indicate.patch uploaded by irroot (license 52)
+
+2011-05-03 21:41 +0000 [r316331] Russell Bryant <russell@digium.com>
+
+ * apps/app_minivm.c: Resolve another warning.
+
+2011-05-03 21:37 +0000 [r316330] David Vossel <dvossel@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 316329 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r316329 | dvossel | 2011-05-03 16:29:55 -0500
+ (Tue, 03 May 2011) | 17 lines Merged revisions 316328 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r316328 | dvossel | 2011-05-03 16:27:59 -0500 (Tue, 03 May 2011)
+ | 10 lines Fixes chan_local crashs in local_fixup() Thanks OEJ
+ for tracking down the issue and submitting the patch. (closes
+ issue #19053) Reported by: oej Tested by: oej Review:
+ https://reviewboard.asterisk.org/r/1158/ ........
+ ................
+
+2011-05-03 19:55 +0000 [r316265] Russell Bryant <russell@digium.com>
+
+ * res/res_musiconhold.c, apps/app_ices.c, apps/app_followme.c,
+ main/config.c, main/cdr.c, main/channel.c, channels/chan_phone.c,
+ funcs/func_enum.c, main/manager.c, channels/chan_skinny.c,
+ res/res_agi.c, main/plc.c, main/features.c, apps/app_minivm.c,
+ apps/app_amd.c, main/pbx.c, res/res_fax.c, formats/format_wav.c,
+ apps/app_festival.c, channels/chan_agent.c, apps/app_originate.c,
+ apps/app_queue.c, codecs/lpc10/dyptrk.c,
+ include/asterisk/linkedlists.h, main/audiohook.c,
+ pbx/pbx_config.c, main/asterisk.c, main/dsp.c,
+ res/res_calendar.c, apps/app_voicemail.c, main/udptl.c,
+ channels/chan_unistim.c, main/fskmodem_float.c,
+ main/rtp_engine.c: Fix a bunch of compiler warnings generated by
+ gcc 4.6.0. Most of these are -Wunused-but-set-variable, but there
+ were a few others mixed in here, as well.
+
+2011-05-03 19:18 +0000 [r316224] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c, channels/chan_dahdi.c, channels/sig_analog.c:
+ The dahdi_hangup() call does not clean up the channel fully.
+ After dahdi_hangup() has supposedly hungup an ISDN channel there
+ is still traffic on the S0-bus because the channel was not
+ cleaned up fully. Shuffled the hangup code to include some
+ missing cleanup. Also fixed some code formatting in the area. I
+ think the primary missing clean up code was the call to
+ tone_zone_play_tone() to turn off any active tones on the
+ channel. (closes issue #19188) Reported by: jg1234 Patches:
+ issue19188_v1.8.patch uploaded by rmudgett (license 664) Tested
+ by: jg1234
+
+2011-05-03 18:59 +0000 [r316215-316217] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Never put the Require: timer header in an
+ Invite. This has already been discussed and should have been
+ resolved earlier. View revsion 285565's log for more information
+ about why it is important to not put timer in the Require header.
+ (closes issue #18704) Reported by: mfrager
+
+ * res/res_odbc.c: Fixes a random crash (NULL reference) in
+ res_odbc.c. (closes issue #19180) Reported by: pruiz Patches:
+ tmp.diff uploaded by pruiz (license 1152) Tested by: pruiz,
+ seanbright
+
+2011-05-03 18:17 +0000 [r316206] Sean Bright <sean@malleable.com>
+
+ * main/manager.c: If we aren't interested in events, don't generate
+ the FullyBooted event on AMI login. (closes issue #19089)
+ Reported by: bklang Patches: issue19089-1.8-r316204.patch
+ uploaded by seanbright (license 71) Tested by: seanbright
+
+2011-05-03 10:57 +0000 [r316193] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * autoconf/ast_check_pwlib.m4, configure: Re-fix bashism in
+ ./configure: s/let/$(( ))/ A forward-port in r278985 accidentally
+ re-introduced issue 17485. Fixing it. Thanks to Jilles Tjoelker
+ for the good report. (closes issue #17485)
+
+2011-05-02 19:09 +0000 [r316094] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * funcs/func_curl.c, /: Merged revisions 316093 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r316093 | tilghman | 2011-05-02 14:04:36 -0500 (Mon, 02 May 2011)
+ | 8 lines More possible crashes based upon invalid inputs.
+ (closes issue #18161) Reported by: wdoekes Patches:
+ 20110301__issue18161.diff.txt uploaded by tilghman (license 14)
+ Tested by: wdoekes ........
+
+2011-04-27 19:14 +0000 [r315894] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/reqresp_parser.c: Merged
+ revisions 315893 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r315893 | mnicholson | 2011-04-27 14:03:05 -0500
+ (Wed, 27 Apr 2011) | 21 lines Merged revisions 315891 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r315891 | mnicholson | 2011-04-27 13:57:56 -0500 (Wed, 27 Apr
+ 2011) | 14 lines Fix our compliance with RFC 3261 section 18.2.2.
+ This change optimizes the free_via() function and removes some
+ redundant null checking. It also fixes compliance with RFC 3261
+ section 18.2.2 by always using the port specified in the Via
+ header for routing responses (even when maddr is not set). Also
+ the htons() function is now used when setting the port.
+ Additional documentation comments have been added in various
+ places to make the logic in the code clearer. (closes issue
+ #18951) Reported by: jmls Patches:
+ issue18951_set_proper_port_from_via.patch uploaded by wdoekes
+ (license 717) (modified) ........ ................
+
+2011-04-27 15:55 +0000 [r315810] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c: Set the copyright year to 2011 in the startup
+ message.
+
+2011-04-27 12:36 +0000 [r315765] Leif Madsen <lmadsen@digium.com>
+
+ * sounds/sounds.xml, sounds/Makefile: Enable Russian core sound
+ selection in menuselect. (closes issue #18724) Reported by:
+ pbxware
+
+2011-04-26 22:56 +0000 [r315673] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 315672 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r315672 | twilson | 2011-04-26 15:52:25 -0700
+ (Tue, 26 Apr 2011) | 18 lines Merged revisions 315671 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r315671 | twilson | 2011-04-26 15:47:56 -0700 (Tue, 26 Apr 2011)
+ | 11 lines Make sure unregistering a peer unlinks it from the
+ peer container Instead of mostly copying the code from
+ expire_register, just use the function that "does the right
+ thing". (closes issue #16033) Reported by: kkm Patches:
+ 016033-tilgman-fixed-refcount.diff uploaded by kkm (license 888)
+ Tested by: kkm, tilghman, twilson ........ ................
+
+2011-04-26 22:14 +0000 [r315645] Richard Mudgett <rmudgett@digium.com>
+
+ * main/pbx.c: The 'e' special extension fails to trigger in at
+ least two cases. The 'e' extension is a fall back for the 'i',
+ 't', or 'T' extensions if any of them do not exist. Many of the
+ places the 'e' extension was supposed to be invoked fail because
+ the priority was set wrong. There were two places where the 'e'
+ extension was not even checked for fall back. * Made invoke the
+ 'e' extension similarly to the previous 'i', 't', or 'T'
+ extension check and added the 'e' extension as a fall back to the
+ two missing locations. * Prioritized and optimized some hangup
+ tests associated with the 'e' extension. (closes issue #19136)
+ Reported by: kshumard Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1196/
+
+2011-04-26 21:39 +0000 [r315644] Terry Wilson <twilson@digium.com>
+
+ * apps/app_queue.c, apps/app_dial.c, /, main/features.c: Merged
+ revisions 315643 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r315643 | twilson | 2011-04-26 14:27:44 -0700
+ (Tue, 26 Apr 2011) | 25 lines Merged revisions 315596 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011)
+ | 18 lines Allow transfer loops without allowing forwarding loops
+ We try to avoid the situation where two phones may be forwarded
+ to each other causing an infinite loop by storing each dialed
+ interface in a channel datastore and checking the list before
+ dialing out. This works, but currently breaks situations like A
+ calls B, A transfers B to C, B transfers C to A, and A transfers
+ C to B. Since human interaction is happening here and not an
+ automated forwarding loop, it should be allowed. This patch
+ removes the dialed_interfaces datastore when a call is bridged (a
+ suggestion from the brilliant mmichelson). If a call is being
+ bridged, it should be safe to assume that we aren't stuck in a
+ loop. Since we are now handling this is the bridge code, the
+ previous attempts at handling it in app_dial and app_queue are
+ removed. Review: https://reviewboard.asterisk.org/r/1195/
+ ........ ................
+
+2011-04-26 19:32 +0000 [r315503] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * include/asterisk/select.h, /: Merged revisions 315502 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r315502 | tilghman | 2011-04-26 14:22:52 -0500
+ (Tue, 26 Apr 2011) | 21 lines Merged revisions 315501 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r315501 | tilghman | 2011-04-26 14:18:46 -0500 (Tue, 26 Apr 2011)
+ | 14 lines Fix the bounds-checking code. The code that set the
+ bit within the select bitfield was correct, but the
+ bounds-checking code was not. The change to that line uses the
+ new _bitsize macro for clarity. Also, FD_ZERO macro did not
+ zero-out anything but the first word of the bitfield, so this
+ could have caused problems with modules using that macro with the
+ expanded bitfield. (closes issue #18773) Reported by: jamicque
+ Patches: 20110423__issue18773.diff.txt uploaded by tilghman
+ (license 14) Tested by: chris-mac ........ ................
+
+2011-04-26 18:00 +0000 [r315452] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c: Add missing set of name valid flag when dialing.
+
+2011-04-26 17:40 +0000 [r315446] Russell Bryant <russell@digium.com>
+
+ * channels/chan_local.c: chan_local: resolve a deadlock. This patch
+ resolves a fairly complex deadlock that can occur with the
+ combination of chan_local and a dialplan switch, such as dynamic
+ realtime extensions, which pulls autoservice into the picture
+ when doing a dialplan lookup. (closes issue #18818) Reported by:
+ nic Patches: issue18818.patch uploaded by jthurman (license 614)
+ 18818.v1.txt uploaded by russell (license 2) Tested by: nic,
+ jthurman, kterzi, steve-howes, sysreq, IshMalik
+
+2011-04-26 02:18 +0000 [r315394] Paul Belanger <pabelanger@digium.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 315393 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r315393 | pabelanger | 2011-04-25 22:17:43 -0400 (Mon, 25 Apr
+ 2011) | 7 lines Add back CLI command 'dialplan save' (closes
+ issue #19140) Reported by: lmadsen Patches:
+ __20110419_dialplan_save.patch.txt uploaded by lmadsen (license
+ 10) ........
+
+2011-04-25 21:49 +0000 [r315349] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_mgcp.c: When using MGCP realtime gateway
+ definitions, random crashes occur. Fixed incorrect linked list
+ node removal for realtime gateways. (closes issue #18291)
+ Reported by: nahuelgreco Patches:
+ dangling-pointers-when-pruning.patch uploaded by nahuelgreco
+ (license 162)
+
+2011-04-25 19:37 +0000 [r315213-315259] Russell Bryant <russell@digium.com>
+
+ * /, formats/format_wav.c: Merged revisions 315258 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r315258 | russell | 2011-04-25 14:31:44 -0500
+ (Mon, 25 Apr 2011) | 17 lines Merged revisions 315257 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r315257 | russell | 2011-04-25 14:28:41 -0500 (Mon, 25 Apr 2011)
+ | 10 lines Be more flexible with unknown chunks in wav files.
+ This patch makes format_wav ignore unknown chunks instead of
+ erroring out on them. (closes issue #18306) Reported by: jhirsch
+ Patches: wav_skip_unknown_blocks.diff uploaded by jhirsch
+ (license 1156) ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 315212 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r315212 | russell | 2011-04-25 14:00:24 -0500 (Mon, 25 Apr 2011)
+ | 7 lines Don't link non-cached realtime peers into the
+ peers_by_ip container. (closes issue #18924) Reported by: wdoekes
+ Patches: issue18924_uncached_realtime_peers_leak-1.6.2.17.patch
+ uploaded by wdoekes (license 717) ........
+
+2011-04-25 07:14 +0000 [r315053] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/chan_local.c, /: Merged revisions 315052 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r315052 | alecdavis | 2011-04-25 19:11:12 +1200
+ (Mon, 25 Apr 2011) | 16 lines Merged revisions 315051 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r315051 | alecdavis | 2011-04-25 19:06:29 +1200 (Mon, 25 Apr
+ 2011) | 11 lines chan_local:check_bridge() misplaced misplaced
+ ast_mutex_unlock if !p->chan->_bridge->_softhangup path isn't
+ followed, brigde remains locked. (closes issue #19176) Reported
+ by: alecdavis Patches: bug19176.diff.txt uploaded by alecdavis
+ (license 585) ........ ................
+
+2011-04-22 22:59 +0000 [r315001] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/chan_dahdi.c: chan_dahdi: Can't return to normal ring
+ after distinctive ring on FXS clear a previous distinctivering
+ pattern before each new call (closes issue #18985) Reported by:
+ bromont Patches: bug18985.diff.txt uploaded by alecdavis (license
+ 585) Tested by: alecdavis, bromont
+
+2011-04-22 21:20 +0000 [r314959] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_agent.c: Merged revisions 314958 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r314958 | mnicholson | 2011-04-22 15:49:45 -0500
+ (Fri, 22 Apr 2011) | 17 lines Merged revisions 311203,314908 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r311203 | mnicholson | 2011-03-17 14:14:37 -0500 (Thu, 17 Mar
+ 2011) | 4 lines Don't hold the pvt lock while streaming a file.
+ ABE-2756 ........ r314908 | mnicholson | 2011-04-22 15:01:48
+ -0500 (Fri, 22 Apr 2011) | 4 lines Prevent the login thread and
+ the app threads from using the asterisk channel at the same time.
+ ABE-2756 ........ ................
+
+2011-04-22 14:02 +0000 [r314780] Russell Bryant <russell@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 314778 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r314778 | russell | 2011-04-22 08:58:03 -0500 (Fri, 22 Apr 2011)
+ | 11 lines Initialize buffers in getvar and getvarfull.
+ Initialize the buffers used to hold the result from GET VARIABLE
+ or GET VARIABLE FULL. The bug report shows func_read returning
+ garbage in the result. It assumed that the buffer passed in was
+ initialized, like many other functions do. In the more common
+ code path (through the dialplan), it is initialized, so just
+ initialize it here too. (closes issue #19050) Reported by: johnz
+ ........
+
+2011-04-22 13:59 +0000 [r314779] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * res/res_fax_spandsp.c, channels/chan_unistim.c: Fix a few typos
+ (shown by Lintian)
+
+2011-04-22 13:35 +0000 [r314777] Russell Bryant <russell@digium.com>
+
+ * /: Recorded merge of revisions 314776 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r314776 | russell | 2011-04-22 08:35:22 -0500 (Fri, 22 Apr 2011)
+ | 10 lines Fix handling of some call parking config options. This
+ patch adjusts the handling of some call parking config options to
+ fix some issues that have already been addressed in 1.8 and
+ trunk. (closes issue #19167) Reported by: bluecrow76 Patches:
+ asterisk-1.6.2.17.2-fix-build-parkinglot-parked-AST_FEATURE_FLAGS.diff
+ uploaded by bluecrow76 (license 270) ........
+
+2011-04-21 22:38 +0000 [r314732] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Correct DAHDIShowChannels XML
+ documentation.
+
+2011-04-21 18:24 +0000 [r314628] Matthew Nicholson <mnicholson@digium.com>
+
+ * configs/sip.conf.sample, configs/skinny.conf.sample,
+ channels/sip/include/sip.h, configs/http.conf.sample,
+ main/manager.c, /, channels/chan_sip.c, channels/chan_skinny.c,
+ main/http.c: Merged revisions 314620 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r314620 | mnicholson | 2011-04-21 13:22:19 -0500
+ (Thu, 21 Apr 2011) | 20 lines Merged revisions 314607 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr
+ 2011) | 14 lines Added limits to the number of unauthenticated
+ sessions TCP based protocols are allowed to have open
+ simultaneously. Also added timeouts for unauthenticated sessions
+ where it made sense to do so. Unrelated, the manager interface
+ now properly checks if the user has the "system" privilege before
+ executing shell commands via the Originate action. AST-2011-005
+ AST-2011-006 (closes issue #18787) Reported by: kobaz (related to
+ issue #18996) Reported by: tzafrir ........ ................
+
+2011-04-21 00:23 +0000 [r314550] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 314549 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r314549 | twilson | 2011-04-20 17:17:34 -0700 (Wed, 20 Apr 2011)
+ | 6 lines Don't allocate more space than necessary for a sip_pkt
+ This extra allocation is a hold-over from when pkt->data was a
+ character array. Now that it is an allocated string, just
+ allocate enough for the sip_pkt. ........
+
+2011-04-20 16:54 +0000 [r314417] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/frame.h: AST_CONTROL_XXX comment changes.
+
+2011-04-20 05:25 +0000 [r314358] Terry Wilson <twilson@digium.com>
+
+ * main/lock.c: Initialize track pointer ast_reentrancy_init checks
+ to see if it is NULL before initializing with calloc
+
+2011-04-19 15:42 +0000 [r314203-314251] Leif Madsen <lmadsen@digium.com>
+
+ * main/tcptls.c: Use SSLv23_client_method instead of old SSLv2
+ only. (closes issue #19095) (closes issue #19138) Reported by:
+ tzafrir Patches: no_ssl2.diff uploaded by tzafrir (license 46)
+ Tested by: russell, chazzam
+
+ * /, funcs/func_channel.c: Merged revisions 314205 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r314205 | lmadsen | 2011-04-19 09:27:50 -0500 (Tue, 19
+ Apr 2011) | 6 lines Remove duplicate documentation from
+ func_channel.c (closes issue #18970) Reported by: IgorG Patches:
+ func_channel.c.doc.diff uploaded by IgorG (license 20) ........
+
+ * apps/app_dial.c, /: Merged revisions 314202 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r314202 | lmadsen | 2011-04-19 09:23:39 -0500 (Tue, 19 Apr 2011)
+ | 7 lines Update seconds to milliseconds in ast_verb output.
+ (closes issue #19084) Reported by: smurfix Patches:
+ app_dial.patch uploaded by smurfix (license 547) Tested by:
+ lmadsen, smurfix ........
+
+2011-04-18 16:10 +0000 [r314068-314069] Richard Mudgett <rmudgett@digium.com>
+
+ * res/res_agi.c: The AsyncAGI command loop is lax in the value it
+ returns for the return status. * Return correct status:
+ SUCCESS/FAILED/HANGUP. Previously, abnormal exits from the
+ command loop such as hangup would return SUCCESS. * The "asyncagi
+ break" command now returns SUCCESS and is now the only way to
+ break the command loop with that status. Previously, it returned
+ FAILED. * The AMI event AsyncAGI End is no longer sent if the
+ AsyncAGI Start event is not sent. Previously, this happened
+ because of an error setting up the AGI pipes. * All executed AGI
+ commands now get an AsyncAGI Exec result event. Previously, if
+ the command returned failure (because of hangup), the command
+ loop just exited with FAILURE and did not send the AsyncAGI Exec
+ result event. * Makes sure that the channel frame queue is empty
+ on hangup. Review: https://reviewboard.asterisk.org/r/1183/
+
+ * apps/app_dial.c: Unclear code in app_dial.c. Make code formatting
+ clear. (closes issue #19134) Reported by: oej
+
+2011-04-18 15:23 +0000 [r314017-314067] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Remove the need for deadlock avoidance in
+ chan_sip do_monitor. Deadlock avoidance between the sip pvt and
+ the pvt->owner is very difficult. Now that channel's are ao2
+ objects, this complication is no longer necessary. It turns out
+ the pvt's msg queue only exists because of deadlock avoidance
+ (when deadlock avoidance fails msgs were added to a queue to be
+ processed later), so this goes away as well. The technique used
+ in the new sip_lock_pvt_full() function should be used as a
+ template for replacing all locations where deadlock avoidance
+ occurs between a channel tech_pvt and the pvt's owner. My hope is
+ that this will begin a reversal of the invalid channel driver
+ locking architecture we have been using for so long. This patch
+ also resolves an issue where the pvt->owner gets unlocked during
+ processing the msg queue. (closes issue #18690) Reported by:
+ dvossel Review: https://reviewboard.asterisk.org/r/1182/
+
+ * include/asterisk/rtp_engine.h, main/rtp_engine.c,
+ channels/chan_sip.c: sip codec negotiation of dynamic rtp
+ payloads error fix This patch fixes how chan_sip handles dynamic
+ rtp payload types it does not understand. At the moment if a
+ dynamic payload's mime type does not match one we understand, the
+ payload does not get removed from our payload table. As a result
+ of this, the payload is set to whatever dynamic codec we use
+ internally for that payload number on outgoing INVITES. This is
+ incorrect. This patch fixes this by properly checking the rtpmap
+ set function's return code to make sure it was found. The
+ function can return both -1 and -2 depending on the source of the
+ mismatch. We were just checking -1 explicitly. Review:
+ https://reviewboard.asterisk.org/r/1169/
+
+2011-04-15 15:08 +0000 [r313860] Jonathan Rose <jrose@digium.com>
+
+ * main/cli.c, /: Merged revisions 313859 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r313859 | jrose | 2011-04-15 09:58:37 -0500 (Fri, 15 Apr 2011) |
+ 10 lines Fix a Tab Completion bug that occurs due to multiple
+ matches on a substring. Makes word_match function in cli.c repeat
+ a search for a command string until a proper match is found or
+ the string is searched to the last point. (closes issue #17494)
+ Reported by: ffossard Review:
+ https://reviewboard.asterisk.org/r/1180/ ........
+
+2011-04-14 20:59 +0000 [r313517-313780] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Leftover debug messages unconditionally
+ sent to the console. Executing Dial(DAHDI/1/18475551212,300,)
+ with the echotraining config option enabled outputs the following
+ debug messages unconditionally: Dialing T1847555121 on 1 Dialing
+ www2w on 1 * Made debug messages in my_dial_digits() normal debug
+ messages that do not get output unless enabled. * Reworded some
+ debug messages in my_dial_digits() to be clearer. * Replace
+ strncpy() with ast_copy_string() in my_dial_digits() which does
+ the same job better. (closes issue #18847) Reported by:
+ vmikhelson Tested by: rmudgett
+
+ * res/res_agi.c: Revert flushing stale AsyncAGI commands from
+ -r313615. It looks like it was intentional to leave any commands
+ or in-flight commands in the queue in case Async AGI is run again
+ on the call.
+
+ * res/res_agi.c: Miscellaneous AGI diagnostic message cleanup and
+ code optimization.
+
+ * res/res_agi.c: * Add missing channel lock to
+ handle_cli_agi_add_cmd(). * Flush any Async AGI commands left
+ over from earlier Async AGI control of the call.
+
+ * main/channel.c, /, res/res_agi.c: Merged revisions 313579 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r313579 | rmudgett | 2011-04-13 11:29:49 -0500
+ (Wed, 13 Apr 2011) | 48 lines Merged revisions 313545 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011)
+ | 41 lines Asterisk does not hangup a channel after endpoint
+ hangs up. If the call that the dialplan started an AGI script for
+ is hungup while the AGI script is in the middle of a command then
+ the AGI script is not notified of the hangup. There are many AGI
+ Exec commands that this can happen with. The reported
+ applications have been: Background, Wait, Read, and Dial. Also
+ the AGI Get Data command. * Don't wait on the Asterisk channel
+ after it has hung up. The channel is likely to never need
+ servicing again. * Restored the AGI script's ability to return
+ the AGI_RESULT_HANGUP value in run_agi(). It previously only
+ could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the
+ DeadAGI and AGI applications were merged. (closes issue #17954)
+ Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by
+ rmudgett (license 664) issue17954_v1.6.2.patch uploaded by
+ rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett
+ (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue
+ #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761
+ (closes issue #18935) Reported by: nvitaly Tested by: astmiv,
+ rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby
+ Tested by: rmudgett JIRA SWP-2727 Review:
+ https://reviewboard.asterisk.org/r/1165/ ........
+ ................
+
+ * apps/app_dumpchan.c: Bring the dumpchan application inline with
+ "core show channel". * Added fields that are in "core show
+ channel" to dumpchan output. * Fixed reuse of formatbuf before
+ the previous string stored there was used by snprintf. All output
+ strings now have their own buffer. * Adjusted the buffer sizes to
+ not be so abusive of the stack now that there are more buffers.
+ Change requested by oej.
+
+2011-04-12 18:47 +0000 [r313434-313436] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_dahdi.c, /: fixing stupid mistake with putting code
+ before variable declaration ........ Merged revisions 313435 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r313432 | jrose | 2011-04-12 13:12:29 -0500 (Tue, 12 Apr 2011) |
+ 14 lines reload Chan_dahdi memory leak caused by variables
+ chan_dahdi reloading with variables set via setvar in
+ chan_dahdi.conf would stay in the dahdi_pvt structs for
+ individual channels (causing them to just continue adding the new
+ ones to the list) and also there was a memory leak causes by the
+ conf objects. This patch resolves both of these by using
+ ast_variables_destroy during the loading process. (closes issue
+ #17450) Reported by: nahuelgreco Patches: patch.diff uploaded by
+ jrose (license 1225) Tested by: tilghman, jrose Review:
+ https://reviewboard.asterisk.org/r/1170/ ........ ........
+
+ * channels/chan_dahdi.c, /: Merged revisions 313432 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r313432 | jrose | 2011-04-12 13:12:29 -0500 (Tue, 12 Apr
+ 2011) | 14 lines reload Chan_dahdi memory leak caused by
+ variables chan_dahdi reloading with variables set via setvar in
+ chan_dahdi.conf would stay in the dahdi_pvt structs for
+ individual channels (causing them to just continue adding the new
+ ones to the list) and also there was a memory leak causes by the
+ conf objects. This patch resolves both of these by using
+ ast_variables_destroy during the loading process. (closes issue
+ #17450) Reported by: nahuelgreco Patches: patch.diff uploaded by
+ jrose (license 1225) Tested by: tilghman, jrose Review:
+ https://reviewboard.asterisk.org/r/1170/ ........
+
+2011-04-11 23:08 +0000 [r313366-313369] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c: Frames from the inbound channel should go to all
+ outbound channels in app_dial.c. In app_dial.c:wait_for_answer()
+ frames from the inbound channel should be sent to all outbound
+ channels instead of only if there is just one outbound channel.
+ Control frames like AST_CONTROL_CONNECTED_LINE need to be passed
+ to all of the the outbound channels. This can happen if a blond
+ transfer is done by a remote switch on the inbound channel. JIRA
+ AST-443 JIRA SWP-2730
+
+ * apps/app_dial.c: Backport a restructuring change from trunk to
+ make the next change stand out.
+
+ * main/cli.c: Added "Connected Line ID" and "Connected Line ID
+ Name" to "core show channel" output.
+
+2011-04-11 19:36 +0000 [r313279] Leif Madsen <lmadsen@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 313278 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r313278 | lmadsen | 2011-04-11 14:33:03 -0500
+ (Mon, 11 Apr 2011) | 14 lines Merged revisions 313277 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r313277 | lmadsen | 2011-04-11 14:30:20 -0500 (Mon, 11 Apr 2011)
+ | 6 lines Fix detection of OpenSSL 1.0 (closes issue #19093)
+ Reported by: tzafrir Patches: detect_openssl_10.diff uploaded by
+ tzafrir (license 46) ........ ................
+
+2011-04-11 15:40 +0000 [r313190] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+ 313189 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r313189 | rmudgett | 2011-04-11 10:32:53 -0500
+ (Mon, 11 Apr 2011) | 32 lines Merged revisions 313188 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r313188 | rmudgett | 2011-04-11 10:27:52 -0500 (Mon, 11 Apr 2011)
+ | 25 lines Stuck channel using FEATD_MF if caller hangs up at the
+ right time. The cause was actually a caller hanging up just at
+ the end of the Feature Group D DTMF tones that setup the call.
+ The reason for this is a "guard timer" that's implemented using
+ ast_safe_sleep(100). If the caller happens to hang up AFTER the
+ final tone of the DTMF string but BEFORE the end of that
+ ast_safe_sleep(), then ast_safe_sleep() will return non-zero.
+ This causes the code to bounce to the end of ss_thread(), but it
+ does NOT tear down the call properly. This should be a rare
+ occurrence because the caller has to hang up at EXACTLY the right
+ time. Nonetheless, it was happening quite regularly on the
+ reporter's system. It's not easily reproducible, unless you
+ purposely increase the guard-time to 2000 or more. Once you do
+ that, you can reproduce it every time by watching the DTMF debug
+ and hanging up just as it ends. Simply add an ast_hangup() before
+ goto quit. (closes issue #15671) Reported by: jcromes Patches:
+ issue15671.patch uploaded by pabelanger (license 224) Tested by:
+ jcromes ........ ................
+
+2011-04-09 20:56 +0000 [r313142] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c: fix trivial bug in ooh323_indicate on
+ AST_CONTROL_SRC... check p->rtp is not null
+
+2011-04-07 13:35 +0000 [r313048] Jonathan Rose <jrose@digium.com>
+
+ * /, main/features.c: Merged revisions 313047 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r313047 | jrose | 2011-04-07 08:23:01 -0500 (Thu, 07 Apr 2011) |
+ 9 lines Makes parking lots clear and rebuild properly when
+ features reload is invoked from CLI Before, default parkinglot in
+ context parkedcalls with ext 700 would always be present and when
+ reload was invoked, the previous parkinglots would not be
+ cleared. (closes issue #18801) Reported by: mickecarlsson Review:
+ https://reviewboard.asterisk.org/r/1161/ ........
+
+2011-04-07 10:24 +0000 [r313001-313002] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * apps/app_voicemail.c: app_voicemail: close_mailbox change
+ LOG_WARNING to LOG_NOTICE
+
+ * channels/sig_pri.c: Fix ISDN calling subaddr User Specified
+ Odd/Even Flag Calculation of the Odd/Even flag was wrong.
+ Implement correct algo, and set odd/even=0 if data would be
+ truncated. Only allow automatic calculation of the O/E flag,
+ don't let dialplan influence. (closes issue #19062) Reported by:
+ festr Patches: bug19062.diff2.txt uploaded by alecdavis (license
+ 585) Tested by: festr, alecdavis, rmudgett
+
+2011-04-05 18:45 +0000 [r312866-312949] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c:
+ Crash if ISDN span layer 1 is down on initial load. Regression
+ from -r312575 B channel shifting during negotiation. * Also
+ combine updating the alarm flag with clearing the resetting flag.
+
+ * channels/chan_sip.c: Add 416 response to OPTIONS packet. RFC3261
+ Section 11.2 says the response code to an OPTIONS packet needs to
+ be the same as if it were an INVITE.
+
+ * channels/chan_sip.c: Responding to OPTIONS packet with 404
+ because Asterisk not looking for "s" extension. The
+ get_destination() function was not using the "s" extension when
+ the request URI did not specify an extension. This is a
+ regression caused when the URI parsing code was extracted into
+ parse_uri(). Made get_destination() substitute the "s" extension
+ when the parsed URI results in an empty string. (closes issue
+ #18348) Reported by: shmaize Patches: issue18348_v1.8.patch
+ uploaded by rmudgett (license 664) Tested by: shmaize
+
+2011-04-05 14:14 +0000 [r312766] Matthew Nicholson <mnicholson@digium.com>
+
+ * configs/manager.conf.sample, main/manager.c, /: Merged revisions
+ 312764 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r312764 | mnicholson | 2011-04-05 09:13:07 -0500
+ (Tue, 05 Apr 2011) | 15 lines Merged revisions 312761 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r312761 | mnicholson | 2011-04-05 09:10:34 -0500 (Tue, 05 Apr
+ 2011) | 8 lines Limit the number of unauthenticated manager
+ sessions and also limit the time they have to authenticate.
+ AST-2011-005 (closes issue #18996) Reported by: tzafrir Tested
+ by: mnicholson ........ ................
+
+2011-04-05 14:13 +0000 [r312765] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 312762 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r312762 | jrose | 2011-04-05 09:11:36 -0500 (Tue, 05 Apr 2011) |
+ 1 line Backporting trunk change to add verbosity to 'L' option in
+ meetme ........
+
+2011-04-04 16:10 +0000 [r312575] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c, /:
+ Merged revisions 312574 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r312574 | rmudgett | 2011-04-04 11:00:02 -0500
+ (Mon, 04 Apr 2011) | 45 lines Merged revisions 312573 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r312573 | rmudgett | 2011-04-04 10:49:30 -0500 (Mon, 04 Apr 2011)
+ | 38 lines Issues with ISDN calls changing B channels during call
+ negotiations. The handling of the PROCEEDING message was not
+ using the correct call structure if the B channel was changed.
+ (The same for PROGRESS.) The call was also not hungup if the new
+ B channel is not provisioned or is busy. * Made all call
+ connection messages (SETUP_ACKNOWLEDGE, PROCEEDING, PROGRESS,
+ ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are
+ using the correct structure and B channel. If there is any
+ problem with the operations then the call is now hungup with an
+ appropriate cause code. * Made miscellaneous messages
+ (INFORMATION, FACILITY, NOTIFY) find the correct structure by
+ looking for the call and not using the channel ID. NOTIFY is an
+ exception with versions of libpri before v1.4.11 because a call
+ pointer is not available for Asterisk to use. * Made all hangup
+ messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find the correct
+ structure by looking for the call and not using the channel ID.
+ (closes issue #18313) Reported by: destiny6628 Tested by:
+ rmudgett JIRA SWP-2620 (closes issue #18231) Reported by:
+ destiny6628 Tested by: rmudgett JIRA SWP-2924 (closes issue
+ #18488) Reported by: jpokorny JIRA SWP-2929 JIRA AST-437 (The
+ issues fixed here are most likely causing this JIRA issue.) JIRA
+ DAHDI-406 JIRA LIBPRI-33 (Stuck resetting flag likely fixed)
+ ........ ................
+
+2011-04-01 23:15 +0000 [r312461-312509] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: When a call going out an NT-PTMP port gets
+ rejected, Asterisk crashes. If a call is sent to an ISDN phone
+ that rejects the call with RELEASE_COMPLETE(cause: call
+ reject(21), or busy(17)) Asterisk crashes. I could not get my
+ setup to crash. However, I could see the possibility from a race
+ condition between queuing an AST_CONTROL_BUSY to the core and
+ then queueing an AST_CONTROL_HANGUP. If the AST_CONTROL_BUSY is
+ processed before the AST_CONTROL_HANGUP is queued, the
+ ast_channel could be destroyed out from under chan_misdn. Avoid
+ this particular crash scenario by not queueing the
+ AST_CONTROL_HANGUP if the AST_CONTROL_BUSY was queued. (closes
+ issue #18408) Reported by: wimpy Patches: issue18408_v1.8.patch
+ uploaded by rmudgett (license 664) Tested by: rmudgett, wimpy
+ JIRA SWP-2679
+
+ * main/ccss.c: CallCompletionRequest()/CallCompletionCancel() exit
+ non-zero if fail. The
+ CallCompletionRequest()/CallCompletionCancel() dialplan
+ applications exit nonzero on normal failure conditions. The
+ nonzero exit causes the dialplan to hangup immediately. The
+ dialplan author has no opportunity to report success/failure to
+ the user. * Made always return zero so the dialplan can continue.
+ * Made set CC_REQUEST_RESULT/CC_REQUEST_REASON and
+ CC_CANCEL_RESULT/CC_CANCEL_REASON channel variables respectively.
+ Also documented the values set. * Reduced the warning about no
+ core instance in CallCompletionCancel() to a debug message. It is
+ a normal event and should not be output at the WARNING level.
+ (closes issue #18763) Reported by: p_lindheimer Patches:
+ ccss.patch uploaded by p lindheimer (license 558) Modified Tested
+ by: p_lindheimer, rmudgett JIRA SWP-3042
+
+2011-04-01 10:58 +0000 [r312286-312288] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/asterisk.c, include/asterisk/select.h, /: Merged revisions
+ 312287 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r312287 | tilghman | 2011-04-01 05:51:24 -0500
+ (Fri, 01 Apr 2011) | 14 lines Merged revisions 312285 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r312285 | tilghman | 2011-04-01 05:36:42 -0500 (Fri, 01 Apr 2011)
+ | 7 lines Found some leaking file descriptors while looking at
+ ast_FD_SETSIZE dead code. (issue #18969) Reported by: oej
+ Patches: 20110315__issue18969__14.diff.txt uploaded by tilghman
+ (license 14) ........ ................
+
+ * addons/cdr_mysql.c: Reload must react correctly against a
+ possibly changed table, so dropping the conditional reload flag.
+
+2011-04-01 09:03 +0000 [r312117-312211] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * apps/app_voicemail.c, /: Merged revisions 312210 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r312210 | alecdavis | 2011-04-01 21:47:29 +1300
+ (Fri, 01 Apr 2011) | 29 lines Merged revisions 312174 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r312174 | alecdavis | 2011-04-01 21:29:49 +1300 (Fri, 01 Apr
+ 2011) | 23 lines voicemail: get real last_message_index and
+ count_messages, ODBC resequence change last_message_index to read
+ the max msgnum stored in the database change count_messages to
+ actually count the number of messages. last_message_index change:
+ This fixed overwriting of the last message if msgnum=0 was
+ missing. Previously every incoming message would overwrite
+ msgnum=1. count_messages change: allows us to detect when
+ requencing is required in opneA_mailbox. resequence enabled for
+ ODBC storage: Assists with fixing up corrupt databases with gaps,
+ but only when a user actively opens there mailboxes. (closes
+ issue #18692,#18582,#19032) Reported by: elguero Patches: based
+ on odbc_resequence_mailbox2.1.diff uploaded by elguero (license
+ 37) Tested by: elguero, nivek, alecdavis Review:
+ https://reviewboard.asterisk.org/r/1153/ ........
+ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 312103 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r312103 | alecdavis | 2011-04-01 20:25:54 +1300
+ (Fri, 01 Apr 2011) | 22 lines Merged revisions 312070 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r312070 | alecdavis | 2011-04-01 19:46:56 +1300 (Fri, 01 Apr
+ 2011) | 16 lines app_voicemail: close_mailbox needs to respect
+ additional messages while mailbox is open. close_mailbox leave
+ gaps in message sequence if messages are deleted and new messages
+ arrive during this time, this is because the shuffle down to slot
+ 0, only shuffles the number of pre-existing messages when mailbox
+ is opened, ignoring new arrivals. Fix: in close_mailbox
+ re-evaluate number of messages before the shuffle, this then
+ includes new arrivals. Happens on filebased or ODBC storage.
+ (issues #19032,#18582,#18692,#18998) Reported by:
+ alecdavis,tootai,afosorio Review:
+ https://reviewboard.asterisk.org/r/1153/ ........
+ ................
+
+2011-03-31 20:11 +0000 [r312022] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: chan_misdn segfaults when DEBUG_THREADS is
+ enabled. The segfault happens because jb->mutexjb is
+ uninitialized from the ast_malloc(). The internals of
+ ast_mutex_init() were assuming a nonzero value meant mutex
+ tracking initialization had already happened. Recent changes to
+ mutex tracking code to reduce excessive memory consumption
+ exposed this uninitialized value. Converted misdn_jb_init() to
+ use ast_calloc() instead of ast_malloc(). Also eliminated
+ redundant zero initialization code in the routine. (closes issue
+ #18975) Reported by: irroot
+
+2011-03-31 06:43 +0000 [r311930] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * configs/cdr_mysql.conf.sample: Incorrect default example; the
+ field is actually internally named "clid", not "callerid".
+ (closes issue #19040) Reported by: wcselby Tested by: tilghman
+
+2011-03-30 01:56 +0000 [r311874] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Update some setup_dahdi_int() comments.
+
+2011-03-29 07:08 +0000 [r311799] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * cel/cel_odbc.c: Remove extraneous check from integer-type fields.
+ (closes issue #19027) Reported by: mlehner Review:
+ https://reviewboard.asterisk.org/r/1149/
+
+2011-03-28 22:00 +0000 [r311751] Russell Bryant <russell@digium.com>
+
+ * apps/app_voicemail.c: Cross-reference VoiceMail() and
+ VoiceMailMain() in the xml docs.
+
+2011-03-27 21:47 +0000 [r311687] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c: correct return values in ooh323_indicate
+ for AST_CONTROL_T38_PARAMETERS
+
+2011-03-23 21:54 +0000 [r311612-311615] Brett Bryant <bbryant@digium.com>
+
+ * apps/app_meetme.c: This patch fixes a bug with MeetMe behavior
+ where the 'P' option for always prompting for a pin is ignored
+ for the first caller. (closes issue #18070) Reported by: mav3rick
+ Review: https://reviewboard.asterisk.org/r/1132/
+
+ * channels/sip/reqresp_parser.c: Fix a possible crash in
+ sip/reqresp_parser.c that is caused by a possible null value.
+ (closes issue #18821) Reported by: cmaj Patches:
+ patch-reqresp_parser_sip_uri_domain_cmp_c_locale-crash-1.8.3-rc2.diff.tx
+ uploaded by cmaj (license 830)
+
+2011-03-23 02:24 +0000 [r311558] Terry Wilson <twilson@digium.com>
+
+ * channels/sip/reqresp_parser.c: Don't use static declared buf in
+ parse_name_andor_addr This function isn't used anywhere yet, but
+ we definitely don't want to keep the same value for buf between
+ calls to the function.
+
+2011-03-22 15:25 +0000 [r311497] David Vossel <dvossel@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 311496 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r311496 | dvossel | 2011-03-22 10:24:45 -0500 (Tue, 22 Mar 2011)
+ | 2 lines Fixes memory leak in MeetMe AMI action ........
+
+2011-03-18 16:19 +0000 [r311352] Jonathan Rose <jrose@digium.com>
+
+ * res/res_jabber.c, channels/chan_sip.c, res/res_fax.c: Changes
+ some print statements/events to use a blank string in place of
+ NULL if the string in question is NULL. This is supposed to
+ improve Solaris compatibility since Solaris goes berserk when
+ trying to output NULL strings. (closes issue #18759) Reported by:
+ bklang Patches: null-strings.patch uploaded by bklang (license
+ 919)
+
+2011-03-18 16:02 +0000 [r311342] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: Properly populate the LOCALSTATIONID channel
+ variable.
+
+2011-03-18 02:59 +0000 [r311295-311297] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Race condition when ISDN
+ CallRerouting/CallDeflection invoked. The queued AST_CONTROL_BUSY
+ could sometimes be processed before the call_forward dial string
+ is recognized. * Moved setting the call_forwarding dial string
+ after sending a response to the initiator and just queue an empty
+ frame to wake up the media thread instead of an AST_CONTROL_BUSY.
+ * Added check for empty rerouting/deflection number and respond
+ with an error.
+
+ * apps/app_dial.c: Merged revision 310986 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r310986 | rmudgett | 2011-03-16 13:56:28 -0500 (Wed,
+ 16 Mar 2011) | 28 lines Dial() o option broke when connected line
+ feature added. The patch restores the o option behavior and adds
+ the ability to specify the CallerID. The Dial o and f options are
+ complementary to each other. The o option stores the CallerID on
+ the outgoing channel as the channel's CallerID. The f option
+ forces the CallerID sent by the outgoing channel. o(x) - The
+ argument 'x' is optional. If not present, then specify that the
+ CallerID that was present on the *calling* channel be stored as
+ the CallerID on the *called* channel. This was the behavior of
+ Asterisk 1.0 and earlier. If present, then specify the CallerID
+ stored on the *called* channel. Note that o(${CALLERID(all)}) is
+ similar to option o without parameters. f(x) - The argument 'x'
+ is optional and its presence changes the behavior of this option.
+ If not present, then force the outgoing CallerID on a
+ call-forward or deflection to the dialplan extension for this
+ Dial() using a dialplan 'hint'. For example, some PSTNs do not
+ allow CallerID to be set to anything other than the numbers
+ assigned to you. If present, then force the outgoing CallerID to
+ 'x'. Patches: jira_abe_2752_dial_fo_options.patch uploaded by
+ rmudgett (license 664) Tested by: rmudgett JIRA ABE-2752 JIRA
+ SWP-3096 ..........
+
+2011-03-17 19:03 +0000 [r311197] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_chanspy.c: This fixes a nasty chanspy bug which was
+ causing a channel leak every time a spied on channel made a call.
+ In addition to the above, it makes certain channel destruction
+ occurs so that applications don't get stuck waiting for datastore
+ destruction while monitored by chanspy. (closes issue #18742)
+ Reported by: jkister Tested by: jkister, jcovert, jrose Review:
+ http://reviewboard.digium.internal/r/106/
+
+2011-03-17 15:00 +0000 [r311141] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/manager.c, /: Merged revisions 311140 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r311140 | mnicholson | 2011-03-17 09:58:52 -0500 (Thu, 17 Mar
+ 2011) | 4 lines Don't write items to the manager socket twice.
+ AST-2011-003 (closes issue 0018987) Reported by: ks-steven
+ ........
+
+2011-03-17 10:49 +0000 [r311050] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, configs/indications.conf.sample: Merged revisions 311049 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r311049 | alecdavis | 2011-03-17 23:45:47 +1300
+ (Thu, 17 Mar 2011) | 17 lines Merged revisions 311048 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r311048 | alecdavis | 2011-03-17 23:43:35 +1300 (Thu, 17 Mar
+ 2011) | 12 lines Remove extra quote in indications.conf Picking
+ low hanging fruit. (closes issue #18971) Reported by: IgorG
+ Patches: based on indications.conf.sample.diff uploaded by IgorG
+ (license 20) Tested by: IgorG ........ ................
+
+2011-03-16 19:47 +0000 [r310902-310999] Terry Wilson <twilson@digium.com>
+
+ * main/tcptls.c, /: Merged revisions 310998 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r310998 | twilson | 2011-03-16 14:46:36 -0500 (Wed, 16 Mar 2011)
+ | 11 lines Fix crash on fdopen failure See security advisory
+ AST-2011-004 (closes issue #18845) Reported by: cmaj Patches:
+ patch-main-tcptls-1.8.3-rc2-open-session-crash-take2.diff.txt
+ uploaded by cmaj (license 830)
+ patch-main-tcptls-1.8.3-rc2-open-session-crash-take3.diff.txt
+ uploaded by cmaj (license 830) Tested by: cmaj, twilson ........
+
+ * main/manager.c, /: Merged revisions 310992 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r310992 | twilson | 2011-03-16 14:23:03 -0500 (Wed, 16 Mar 2011)
+ | 4 lines Don't keep trying to write to a closed connection See
+ security advisory AST-2011-003. ........
+
+ * /, main/features.c: Merged revisions 310889 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r310889 | twilson | 2011-03-16 12:03:27 -0500
+ (Wed, 16 Mar 2011) | 36 lines Merged revisions 310888 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r310888 | twilson | 2011-03-16 11:58:42 -0500 (Wed, 16 Mar 2011)
+ | 29 lines Don't delay DTMF in core bridge while listening for
+ DTMF features This patch is mostly the work of Olle Johansson. I
+ did some cleanup and added the silence generating code if
+ transmit_silence is set. When a channel listens for DTMF in the
+ core bridge, the outbound DTMF is not sent until we have received
+ DTMF_END. For a long DTMF, this is a disaster. We send 4 seconds
+ of DTMF to Asterisk, which sends no audio for those 4 seconds.
+ Some products see this delay and the time skew on RTP packets
+ that results and start ignoring the audio that is sent afterward.
+ With this change, the DTMF_BEGIN frame is inspected and checked.
+ If it matches a feature code, we wait for DTMF_END and activate
+ the feature as before. If transmit_silence=yes in asterisk.conf,
+ silence is sent if we paritally match a multi-digit feature. If
+ it doesn't match a feature, the frame is forwarded along with the
+ DTMF_END without delay. By doing it this way, DTMF is not
+ delayed. (closes issue #15642) Reported by: jasonshugart Patches:
+ issue_15652_dtmf_ast-1.4.patch.txt uploaded by twilson (license
+ 396) Tested by: globalnetinc, jde (closes issue #16625) Reported
+ by: sharvanek Review: https://reviewboard.asterisk.org/r/1092/
+ Review: https://reviewboard.asterisk.org/r/1125/ ........
+ ................
+
+2011-03-15 01:48 +0000 [r310834] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * addons/chan_ooh323.c: Fix branch compile.
+
+2011-03-15 01:00 +0000 [r310781] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/utils.c: core show locks: display ThreadID in hexadecimal
+ Allow easier cross referencing of thread ID's with GDB backtraces
+ (closes issue #18968) Reported by: alecdavis Patches:
+ bug18968.diff.txt uploaded by alecdavis (license 585)
+
+2011-03-14 21:45 +0000 [r310734] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c, addons/ooh323c/src/ooCapability.c,
+ addons/ooh323c/src/ooCalls.h: Introduce t.38 parameters control
+ functionality not full but enough for Send/RcvFax support
+ Introduce t.38 controls between asterisk core and channel/proto
+ layers. Not all parameters are transferred from proto layers but
+ *Fax apps tested and work ok. (issue #18693) Reported by:
+ benngard2 Patches: issue-18693.patch uploaded by may213 (license
+ 454)
+
+2011-03-14 21:30 +0000 [r310726-310733] Jonathan Rose <jrose@digium.com>
+
+ * main/channel.c: Undoes 310726 for further analysis
+
+ * main/channel.c: Moves data store destruction from channel
+ destruction to hangup in channel.c This moves the data store
+ destruction and app signaling events for a call to ast_hangup so
+ that threads which wait for data store destruction don't become
+ stuck forever when attached to an application/function/etc that
+ keeps the channel open. (closes issue #18742) Reported by:
+ jkister Patches: patch.diff uploaded by jrose (license 1225)
+ Tested by: jkister, jcovert, jrose Review:
+ https://reviewboard.asterisk.org/r/1136/
+
+2011-03-14 16:50 +0000 [r310636] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/callerid.c: Merged revisions 310635 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r310635 | rmudgett | 2011-03-14 11:47:54 -0500
+ (Mon, 14 Mar 2011) | 32 lines Merged revisions 310633 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r310633 | rmudgett | 2011-03-14 11:38:24 -0500 (Mon, 14 Mar 2011)
+ | 25 lines "Caller*ID failed checksum" on Wildcard TDM2400P and
+ TDM410 The last character in the caller id message is getting a
+ framing error. The checksum is the last character in the message.
+ A framing error in the checksum could be because: 1) The sender
+ did not send a full stop bit. 2) The sender cut off the FSK
+ carrier too soon. 3) The sender opted to send zero of the
+ specified zero to 10 trailing mark bits and round-off errors in
+ the code resulted in the code not being where it thought it was
+ in the demodulated bit stream. Bit 8 of 'b' is set when parity
+ error. Bit 9 of 'b' is set when framing error. Made ignore the
+ framing and parity error bits if the errored character is the
+ checksum. We can tolerate a framing/parity error there. The
+ checksum character validates the message. (closes issue #18474)
+ Reported by: nivek Patches: callerid.c.1.patch uploaded by nivek
+ (license 636) (with modifications) Tested by: nivek ........
+ ................
+
+2011-03-14 15:27 +0000 [r310587] Jonathan Rose <jrose@digium.com>
+
+ * /, funcs/func_volume.c: Merged revisions 310585 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r310585 | jrose | 2011-03-14 08:56:22 -0500 (Mon, 14 Mar 2011) |
+ 8 lines Adds 'p' as an option to func_volume. When it is on, the
+ old behavior with DTMF controlling volume adjustment will be
+ enforced. When it is off, DTMF will not be processed by the
+ function. Programmed by Jonathan Rose Reviewed by David Vossel,
+ Leif Madsen, and Russell Bryant
+ http://reviewboard.digium.internal/r/93/ ........
+
+2011-03-12 20:27 +0000 [r310415-310462] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, pbx/pbx_ael.c: Merged revisions 310448 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r310448 | tilghman | 2011-03-12 14:24:54 -0600
+ (Sat, 12 Mar 2011) | 38 lines Recorded merge of revisions 310435
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r310435 | tilghman | 2011-03-12 14:22:07 -0600 (Sat, 12 Mar 2011)
+ | 31 lines Add AELSub, which provides a stable entry point into
+ AEL subroutines. This commit needs some explanation, given that
+ we're adding a new application into an existing release branch.
+ This is generally a violation of our release policy, except in
+ very limited circumstances, and I believe this is one of those
+ circumstances. The problem that this solves is one of the sanity
+ of using multiple dialplan languages to define a dialplan. In the
+ case of the reporter, he or she is using AEL is define
+ subroutines, while using Realtime extensions to invoke those
+ subroutines. While you can do this, it's based upon the reality
+ of AEL using actual dialplan extensions; however, there is no
+ guarantee that the details of _how_ AEL is compiled into
+ extensions will remain stable. In fact, at the time of this
+ commit, it has already changed twice, once in a fundamental way.
+ Now normally, a new application would only be added to trunk.
+ However, this application is explicitly to create a stable
+ user-level API between versions, and adding it to trunk only will
+ not solve the user's problem of switching between 1.6.2 and 1.8,
+ nor will it help anybody switching from 1.8 to 1.10. Therefore,
+ it needs to go into existing release branches. For the sake of
+ consistency, and also because one of the changes was between 1.4
+ and 1.6.x, I am also electing to commit this to 1.4. (closes
+ issue #18910) Reported by: alexandrekeller Patches:
+ 20110304__issue18919__1.6.2.diff.txt uploaded by tilghman
+ (license 14) 20110304__issue18919__1.4.diff.txt uploaded by
+ tilghman (license 14) Tested by: alexandrekeller ........
+ ................
+
+ * /, funcs/func_odbc.c: Merged revisions 310414 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r310414 | tilghman | 2011-03-12 13:51:23 -0600 (Sat, 12 Mar 2011)
+ | 7 lines Transactional handles should be used for the insertbuf,
+ if available. Also, fix a possible resource leak. (closes issue
+ #18943) Reported by: irroot ........
+
+2011-03-11 06:47 +0000 [r310287] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/rtp_engine.c: remote_bridge_loop: prevent segfault when
+ after transfer of IAX2 of DAHDI call If the channel condition is
+ one of the following after breaking out of the loop, don't try to
+ update_peer (where x = 0/1) 1). ZOMBIE 2). cx->tech_pvt != pvtx
+ 3). gluex != ast_rtp_instance_get_glue(cx->tech->type)) (closes
+ issue #18781) Reported by: alecdavis Patches: bug18781.diff3.txt
+ uploaded by alecdavis (license 585) Tested by: alecdavis, ZX81
+ Review: https://reviewboard.asterisk.org/r/1128/
+
+2011-03-10 16:05 +0000 [r310240] Terry Wilson <twilson@digium.com>
+
+ * main/manager.c, res/res_phoneprov.c: Add \r\n to remaining http
+ headers passed to ast_http_send r309204 changed the behavior of
+ ast_http_send. It now requires headers to be passed with trailing
+ \r\n. This change updates the remaining instances in the code
+ that did not pass the \r\n. (closes issue #18186) Reported by:
+ nivaldomjunior Patches: res_phoneprov.c.diff uploaded by lathama
+ (license 1028) manager.diff.txt uploaded by twilson (license 396)
+ Tested by: lathama
+
+2011-03-10 15:17 +0000 [r310231] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Be more tolerant of what URI we accept for
+ call completion PUBLISH requests. (closes issue #18946) Reported
+ by: GeorgeKonopacki Patches: 18946.patch uploaded by mmichelson
+ (license 60) Tested by: GeorgeKonopacki
+
+2011-03-10 05:53 +0000 [r310142] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * apps/app_voicemail.c, res/res_config_odbc.c, /,
+ funcs/func_odbc.c: Merged revisions 310141 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r310141 | tilghman | 2011-03-09 23:51:37 -0600
+ (Wed, 09 Mar 2011) | 12 lines Merged revisions 310140 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r310140 | tilghman | 2011-03-09 23:38:44 -0600 (Wed, 09 Mar 2011)
+ | 5 lines Initialize column size to 0 to deal with a potential
+ UnixODBC bug on 64-bit systems. (closes issue #18295) Reported
+ by: pruiz ........ ................
+
+2011-03-08 20:19 +0000 [r310088] Jonathan Rose <jrose@digium.com>
+
+ * channels/sip/dialplan_functions.c: Returns with an error notice
+ if CHANNEL function of SIP channel is read without arguments.
+ (Closes issue #18653) Reported by: wuwu Patches: diff.patch
+ uploaded by jrose (license 1225) Tested by: jrose
+
+2011-03-08 18:10 +0000 [r310039] Terry Wilson <twilson@digium.com>
+
+ * res/res_calendar.c: Spelling fix in "calendar show calendar"
+ s/Cartegories/Catagories/ (closes issue #18931) Reported by:
+ pdugas Patches: res_calendar.c.patch uploaded by pdugas (license
+ 1222)
+
+2011-03-08 16:37 +0000 [r309994] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Make pri parameter description consistent.
+
+2011-03-07 22:07 +0000 [r309858] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_mixmonitor.c, /: Merged revisions 309857 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r309857 | jrose | 2011-03-07 16:04:44 -0600
+ (Mon, 07 Mar 2011) | 15 lines Merged revisions 309856 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r309856 | jrose | 2011-03-07 16:02:12 -0600 (Mon, 07 Mar 2011) |
+ 8 lines Bug fix for MixMonitor involving filenames with '.' not
+ in the extension Closes issue #18391) Reported by: pabelanger
+ Patches: bugfix.patch uploaded by jrose (license 1225) Tested by:
+ jrose ........ ................
+
+2011-03-07 00:54 +0000 [r309808] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/ast_expr2.fl, channels/chan_dahdi.c, /, configure,
+ include/asterisk/autoconfig.h.in, main/ast_expr2f.c,
+ configure.ac: Merged revisions 309251 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r309251 | tilghman | 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011)
+ | 7 lines Revert previous 2 commits, and instead conditionally
+ redefine the same macro used in flex 2.5.35 that clashed with our
+ workaround. Not surprisingly, the workaround was exactly the same
+ code as was provided by the Flex maintainers, albeit in two
+ different places, in different macros. This should fix the
+ FreeBSD builds, which have an older version of Flex. ........
+
+2011-03-07 00:13 +0000 [r309765] Mark Michelson <mmichelson@digium.com>
+
+ * configs/sip.conf.sample: Indicate that Asterisk uses the Allow
+ header to determine if MESSAGE requests should be sent.
+
+2011-03-05 17:44 +0000 [r309720] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c: Fix caller id passed to
+ openr2_chan_make_call (closes issue #18894) Reported by: malufrj
+ Tested by: moy
+
+2011-03-05 10:29 +0000 [r309678] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/asterisk.c, /: Merged revisions 309677 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r309677 | tilghman | 2011-03-05 04:28:24 -0600 (Sat, 05 Mar 2011)
+ | 7 lines Missed part of the conversion when we started passing
+ ppid to astcanary. (closes issue #18850) Reported by: viraptor
+ Patches: canary_ppid.patch uploaded by viraptor (license 543)
+ ........
+
+2011-03-04 19:38 +0000 [r309448-309585] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, pbx/pbx_lua.c: Merged revisions 309584 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r309584 | mnicholson | 2011-03-04 13:37:13 -0600 (Fri, 04 Mar
+ 2011) | 2 lines Restore mysterious lua_pushvalue() call removed
+ in r309494. The mystery has been solved. ........
+
+ * /, pbx/pbx_lua.c: Merged revisions 309541 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r309541 | mnicholson | 2011-03-04 12:59:20 -0600 (Fri, 04 Mar
+ 2011) | 4 lines Check for errors from fseek() when loading config
+ file, properly abort on errors from fread(), and supply a
+ traceback for errors generated when loading the config file.
+ Also, prepend a newline to traceback output so that the main
+ error message is on it's own line. ........
+
+ * /, pbx/pbx_lua.c: Merged revisions 309494 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r309494 | mnicholson | 2011-03-04 11:55:57 -0600 (Fri, 04 Mar
+ 2011) | 2 lines remove mysterious lua_pushvalue() that is never
+ used ........
+
+ * pbx/pbx_lua.c: Export global symbols from pbx_lua to allow
+ modules to be loaded. Fixes a regression introduced in r278132.
+ (closes issue #18671) Reported by: Igels Patches:
+ pbx_lua_global_symbols1.diff uploaded by mnicholson (license 96)
+ Tested by: Igels
+
+2011-03-04 15:22 +0000 [r309445] Richard Mudgett <rmudgett@digium.com>
+
+ * UPGRADE.txt, channels/sig_pri.c, channels/sig_pri.h,
+ channels/chan_dahdi.c, funcs/func_channel.c: Get real channel of
+ a DAHDI call. Starting with Asterisk v1.8, the DAHDI channel name
+ format was changed for ISDN calls to:
+ DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number> There
+ were several reasons that the channel name had to change. 1) Call
+ completion requires a device state for ISDN phones. The generic
+ device state uses the channel name. 2) Calls do not necessarily
+ have B channels. Calls placed on hold by an ISDN phone do not
+ have B channels. 3) The B channel a call initially requests may
+ not be the B channel the call ultimately uses. Changes to the
+ internal implementation of the Asterisk master channel list
+ caused deadlock problems for chan_dahdi if it needed to change
+ the channel name. Chan_dahdi no longer changes the channel name.
+ 4) DTMF attended transfers now work with ISDN phones because the
+ channel name is "dialable" like the chan_sip channel names. For
+ various reasons, some people need to know which B channel a DAHDI
+ call is using. * Added CHANNEL(dahdi_span),
+ CHANNEL(dahdi_channel), and CHANNEL(dahdi_type) so the dialplan
+ can determine the B channel currently in use by the channel. Use
+ CHANNEL(no_media_path) to determine if the channel even has a B
+ channel. * Added AMI event DAHDIChannel to associate a DAHDI
+ channel with an Asterisk channel so AMI applications can
+ passively determine the B channel currently in use. Calls with
+ "no-media" as the DAHDIChannel do not have an associated B
+ channel. No-media calls are either on hold or call-waiting.
+ (closes issue #17683) Reported by: mrwho Tested by: rmudgett
+ (closes issue #18603) Reported by: arjankroon Patches:
+ issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664)
+ Tested by: stever28, rmudgett
+
+2011-03-04 01:50 +0000 [r309403] David Ruggles <thedavidfactor@gmail.com>
+
+ * apps/app_externalivr.c, /: Merged revisions 309356 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r309356 | diruggles | 2011-03-03 19:42:28 -0500
+ (Thu, 03 Mar 2011) | 16 lines Merged revisions 309355 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r309355 | diruggles | 2011-03-03 19:34:13 -0500 (Thu, 03 Mar
+ 2011) | 9 lines fix small memory leak fix small memory leak
+ caused by a string allocation that wasn't freed (closes issue
+ #18907) Reported by: andy11 Patches:
+ asterisk_trunk-app_externalivr-leak.patch uploaded by andy11
+ (license 1224) ........ ................
+
+2011-03-02 19:54 +0000 [r309204-309256] Jason Parker <jparker@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 309255 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r309255 | qwell | 2011-03-02 13:53:47 -0600 (Wed, 02 Mar 2011) |
+ 8 lines Fix usage of "hasvoicemail=yes" and "mailbox=" in
+ users.conf for SIP. Since it's a duplicate, nothing is going to
+ be done, so delme doesn't need to be set at all. Strangely, when
+ this was added, this was being set to 1 in 1.6, and 0 in trunk.
+ (issue AST-439) ........
+
+ * main/http.c: Fix consistency of CRLFs on HTTP headers that get
+ sent out. (closes issue #18186) Reported by: nivaldomjunior
+ Patches: 18186-httpheadernewline.diff uploaded by qwell (license
+ 4)
+
+2011-03-01 21:57 +0000 [r309126-309170] Richard Mudgett <rmudgett@digium.com>
+
+ * funcs/func_channel.c: Document CHANNEL(keypad_digits) and
+ CHANNEL(no_media_path). * Added XML documentation for
+ CHANNEL(keypad_digits) and CHANNEL(no_media_path). * Tweaked XML
+ documentation for CHANNEL(reversecharge).
+
+ * channels/sig_analog.c: Chan_dahdi does not retain CID when
+ detecting DTMF CID without polarity reversal. Looks like an
+ unintended change when sig_analog.c was extracted from
+ chan_dahdi.c. Removed useless conditional around needed code and
+ fixed resulting compiler warning. (closes issue #18667) Reported
+ by: enegaard Patches: issue18667.patch uploaded by enegaard
+ (license 1197) Tested by: enegaard JIRA SWP-2965
+
+2011-03-01 16:09 +0000 [r309084] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 309083 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r309083 | dvossel | 2011-03-01 10:05:25 -0600 (Tue, 01 Mar 2011)
+ | 9 lines Fixes thread blocking issue in the sip TCP/TLS
+ implementation. (closes issue #18497) Reported by: vois Patches:
+ issues_18497.diff uploaded by dvossel (license 671) Tested by:
+ vois, rossbeer, kowalma, Freddi_Fonet ........
+
+2011-02-28 11:10 +0000 [r308991-309035] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/ast_expr2.fl, /, configure,
+ include/asterisk/autoconfig.h.in, main/ast_expr2f.c,
+ configure.ac: Merged revisions 309033-309034 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r309033 | tilghman | 2011-02-28 04:43:12 -0600 (Mon, 28 Feb 2011)
+ | 4 lines A later version of flex already includes the fwrite
+ workaround code, which if used twice causes a compilation error.
+ Detect whether Flex will compile without the workaround; if so,
+ suppress our workaround code. ........ r309034 | tilghman |
+ 2011-02-28 05:07:52 -0600 (Mon, 28 Feb 2011) | 2 lines Clarify
+ meaning, removing double negative (stupid!) ........
+
+ * /, funcs/func_odbc.c: Merged revisions 308990 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r308990 | tilghman | 2011-02-28 03:32:22 -0600 (Mon, 28 Feb 2011)
+ | 7 lines Statements updating zero rows may return SQL_NO_DATA.
+ This is fine; it's handled. (closes issue #18815) Reported by:
+ irroot Patches: func_odbc.insert_nodata.patch uploaded by irroot
+ (license 52) ........
+
+2011-02-25 18:52 +0000 [r308945] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/chan_sip.c: Fix Deadlock with attended transfer of SIP
+ call Call path sip_set_rtp_peer (locks chan then pvt)
+ transmit_reinvite_with_sdp try_suggested_sip_codec
+ pbx_builtin_getvar_helper (locks p->owner) But by the time
+ p->owner lock was attempted, seems as though chan and p->owner
+ were different. So in sip_set_rtp_peer, lock pvt first then lock
+ p->owner using deadlocking methods. (closes issue #18837)
+ Reported by: alecdavis Patches: bug18837-trunk.diff3.txt uploaded
+ by alecdavis (license 585) Tested by: alecdavis, Irontec, ZX81,
+ cmaj Review: [https://reviewboard.asterisk.org/r/1126/]
+
+2011-02-24 21:38 +0000 [r308903] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Invalid read in ast_channel_set_caller_event().
+ Valgrind reported that ast_channel_set_caller_event() was reading
+ data from a freed buffer when using the pre_set structure.
+ Rearange things to pre-calculate the name and number pointer
+ before updating the caller party structure to see if the name or
+ number was changed.
+
+2011-02-24 17:57 +0000 [r308815] Terry Wilson <twilson@digium.com>
+
+ * main/manager.c, /: Merged revisions 308814 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r308814 | twilson | 2011-02-24 11:54:49 -0600
+ (Thu, 24 Feb 2011) | 19 lines Merged revisions 308813 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r308813 | twilson | 2011-02-24 11:42:16 -0600 (Thu, 24 Feb 2011)
+ | 12 lines Don't broadcast FullyBooted to every AMI connection
+ The FullyBooted event should not be sent to every AMI connection
+ every time someone connects via AMI. It should only be sent to
+ the user who just connected. (closes issue #18168) Reported by:
+ FeyFre Patches: bug0018168.patch uploaded by FeyFre (license
+ 1142) Tested by: FeyFre, twilson ........ ................
+
+2011-02-24 15:06 +0000 [r308723] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/udptl.c, /: Merged revisions 308722 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r308722 | mnicholson | 2011-02-24 08:59:41 -0600
+ (Thu, 24 Feb 2011) | 9 lines Merged revisions 308721 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r308721 | mnicholson | 2011-02-24 08:54:56 -0600 (Thu,
+ 24 Feb 2011) | 2 lines silence gcc 4.2 compiler warning ........
+ ................
+
+2011-02-24 03:41 +0000 [r308679] Terry Wilson <twilson@digium.com>
+
+ * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
+ 308678 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r308678 | twilson | 2011-02-23 21:38:22 -0600 (Wed, 23 Feb 2011)
+ | 8 lines Use remotesecret to authenticate with a remote party
+ The remotesecret option was only being used for outbound
+ registration and not for placing calls. This patch uses
+ remotesecret on outbound calls if it is set, otherwise secret is
+ still used. Review: https://reviewboard.asterisk.org/r/1107/
+ ........
+
+2011-05-09 Leif Madsen <lmadsen@digium.com>
+
+ * Asteris 1.8.4 Released.
+
+2011-04-25 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.4-rc3 Released.
+
+ * Use SSLv23_client_method instead of old SSLv2 only.
+
+ (closes issue 0019095)
+ (closes issue 0019138)
+ Reported by: tzafrir
+ Patches:
+ no_ssl2.diff uploaded by tzafrir (license 46)
+ Tested by: russell, chazzam
+
+ * Resolve crash in ast_mutex_init()
+
+2011-02-25 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.4-rc2 Released.
+
+ * Fix Deadlock with attended transfer of SIP call
+ (Closes issue #18837. Reported, patched by alecdavis. Tested by
+ alecdavid, Irontec, ZX81, cmaj)
+
+2011-02-23 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.4-rc1 Released.
+
+2011-02-23 23:38 +0000 [r308622] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: sig_pri_new_ast_channel() should return NULL
+ when new_ast_channel() fails. (closes issue #18874) Reported by:
+ cmaj Patches:
+ patch-sig_pri-crash-possible-null-channel-pointer.diff.txt
+ uploaded by cmaj (license 830) JIRA SWP-3172
+
+2011-02-22 15:31 +0000 [r308526] Andrew Latham <lathama@gmail.com>
+
+ * main/http.c: Use ast_debug for console logging Guessed the log
+ levels based on info that level 3 is the soft roof. Can we create
+ a page / document to define the levels?
+
+2011-02-21 15:02 +0000 [r308416] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/udptl.c, /: Merged revisions 308414 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r308414 | mnicholson | 2011-02-21 09:00:22 -0600
+ (Mon, 21 Feb 2011) | 12 lines Merged revisions 308413 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r308413 | mnicholson | 2011-02-21 08:57:15 -0600 (Mon, 21 Feb
+ 2011) | 5 lines Properly check the bounds of arrays when decoding
+ UDPTL packets. Also, remove broken support for receiving UDPTL
+ packets larger than 16k. That shouldn't ever happen anyway.
+ AST-2011-002 FAX-281 ........ ................
+
+2011-02-21 14:24 +0000 [r308393] Andrew Latham <lathama@gmail.com>
+
+ * main/http.c: Add HTTP URI Debug logging and update notice enable
+ reporting of the request URI / URL in debugging change funny
+ debug note to a serious note.
+
+2011-02-19 14:06 +0000 [r308330] Andrew Latham <lathama@gmail.com>
+
+ * main/http.c: Add CSS MIME Type Modern browsers are checking for
+ the MIME Type of pages and in some cases will not load a file if
+ the type is wrong.
+
+2011-02-19 11:02 +0000 [r308288] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * utils: A few more (copies of) files to ignore in this directory.
+
+2011-02-18 00:07 +0000 [r308242] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323cDriver.c, addons/ooh323cDriver.h,
+ addons/chan_ooh323.c: added g729onlyA option for announce only
+ AnnexA g.729 codec in h.323 capabilities. Option can be global or
+ per user/peer.
+
+2011-02-16 20:21 +0000 [r308150] Paul Belanger <pabelanger@digium.com>
+
+ * addons/ooh323c/src/ooSocket.c: Fix FreeBSD builds.
+
+2011-02-16 07:57 +0000 [r308098] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooSocket.c: ifdef __linux__ keepalive
+ variables also
+
+2011-02-15 23:34 +0000 [r308010] Jason Parker <jparker@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 308007 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r308007 | qwell | 2011-02-15 17:33:24 -0600
+ (Tue, 15 Feb 2011) | 17 lines Merged revisions 308002 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r308002 | qwell | 2011-02-15 17:32:20 -0600 (Tue, 15 Feb 2011) |
+ 10 lines Fix regression that changed behavior of queues when
+ ringing a queue member. This reverts r298596, which was to fix a
+ highly bizarre and contrived issue with a queue member that
+ called into his own queue being transferred back into his own
+ queue. I couldn't reproduce that issue in any way. I think one of
+ the other recent transfer fixes actually fixed this. (closes
+ issue #18747) Reported by: vrban ........ ................
+
+2011-02-15 23:08 +0000 [r307970] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooSocket.c: include tcp keepalive socket calls
+ only on linux, freebsd and others don't have these options on
+ sockets.
+
+2011-02-15 19:52 +0000 [r307879-307962] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c: Don't crash when forcing caller id.
+
+ * channels/sig_pri.c, include/asterisk/ccss.h, channels/sig_pri.h,
+ channels/chan_dahdi.c, channels/chan_sip.c, main/ccss.c: No
+ response sent for SIP CC subscribe/resubscribe request. Asterisk
+ does not send a response if we try to subscribe for call
+ completion after we have received a 180 Ringing. You can only
+ subscribe for call completion when the call has been cleared.
+ When we receive the 180 Ringing, for this call, its
+ call-completion state is 'CC_AVAILABLE'. If we then send a
+ subscribe message to Asterisk, it trys to change the
+ call-completion state to 'CC_CALLER_REQUESTED'. Because this is
+ an invalid state change, it just ignores the message. The only
+ state Asterisk will accept our subscribe message is in the
+ 'CC_CALLER_OFFERED' state. Asterisk will go into the
+ 'CC_CALLER_OFFERED' when the SIP client clears the call by
+ sending a CANCEL. Asterisk should always send a response. Even if
+ its a negative one. The fix is to allow for the CCSS core to
+ notify a CC agent that a failure has occurred when CC is
+ requested. The "ack" callback is replaced with a "respond"
+ callback. The "respond" callback has a parameter indicating
+ either a successful response or a specific type of failure that
+ may need to be communicated to the requester. (closes issue
+ #18336) Reported by: GeorgeKonopacki Tested by: mmichelson,
+ rmudgett JIRA SWP-2633 (closes issue #18337) Reported by:
+ GeorgeKonopacki Tested by: mmichelson JIRA SWP-2634
+
+2011-02-15 07:02 +0000 [r307750-307837] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, funcs/func_odbc.c: Merged revisions 307836 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r307836 | tilghman | 2011-02-15 01:01:37 -0600 (Tue, 15 Feb 2011)
+ | 8 lines Need to retrieve the rows affected before using the
+ associated variable. (closes issue #18795) Reported by: irroot
+ Patches: 20110211__issue18795.diff.txt uploaded by tilghman
+ (license 14) Tested by: tilghman ........
+
+ * res/res_odbc.c, /: Merged revisions 307792 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r307792 | tilghman | 2011-02-14 14:10:28 -0600 (Mon, 14 Feb 2011)
+ | 8 lines Increment usage count at first reference, to avoid a
+ race condition with many threads creating connections all at
+ once. (issue #18156) Reported by: asgaroth Patches:
+ 20110214__issue18156.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman ........
+
+ * apps/app_queue.c, apps/app_dial.c: Calling a gosub routine
+ defined in AEL from Dial/Queue ceased to work. A bug in AEL did
+ not distinguish between the "s" extension generated by AEL and an
+ "s" extension that was required to exist by the chan_dahdi (or
+ another channel) that was not supplied with a starting extension.
+ Therefore, AEL made incorrect assumptions about what commands
+ were permissable in the context. This was fixed by making AEL
+ generate a different extension name. However, Dial and Queue make
+ additional assumptions about the name of the default gosub
+ extension. Therefore, they needed to be brought into line with a
+ "macro" rendered by AEL (as a gosub), without breaking
+ traditional dialplans written without the aid of AEL. Related to
+ (issue #18480) Reported by: nivek (closes issue #18729) Reported
+ by: kkm Patches: 20110209__issue18729.diff.txt uploaded by
+ tilghman (license 14) 018729-dial-queue-gosub-try3.patch uploaded
+ by kkm (license 888) Tested by: kkm
+
+2011-02-10 22:39 +0000 [r307536] Jason Parker <jparker@digium.com>
+
+ * main/asterisk.c, contrib/init.d/rc.debian.asterisk, /: Merged
+ revisions 307535 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r307535 | qwell | 2011-02-10 16:35:49 -0600
+ (Thu, 10 Feb 2011) | 15 lines Merged revisions 307534 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r307534 | qwell | 2011-02-10 16:33:09 -0600 (Thu, 10 Feb 2011) |
+ 8 lines Remove color when executing commands via a remote
+ console. Essentially this makes '-x' imply '-n' on rasterisk.
+ This was done in a different and incomplete way previously, which
+ I'm reverting here. (issue #18776) Reported by: alecdavis
+ ........ ................
+
+2011-02-10 18:50 +0000 [r307509] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/oochannels.c,
+ addons/ooh323c/src/ooStackCmds.c, addons/ooh323c/src/ooq931.c,
+ addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
+ addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooCalls.h:
+ Corrections for properly work with H.323v2 (older) endpoints and
+ other small fixes. Interpret remote side H.225 version.
+ Corrections for H.323v2 endpoints: don't start TCS and MSD before
+ connect, don't start TCS and MSD by accepting H.245 connection,
+ start TCS and MSD by StartH245 facility message. Other fixes: fix
+ non zeroended remoteDisplayName issue, small fixes in call
+ clearing by closing H.245 connection, tcp keepalive introduced on
+ TCP connections (now is hardcoded, will be configurable in the
+ future), don't force H.245tunneling if FastStart is active, don't
+ send Alerting singal more than once per call. (issue 0018542)
+ Reported by: vmikhelson Patches: issue18542-final-3.patch
+ uploaded by may213 (license 454) Tested by: vmikhelson
+
+2011-02-10 17:44 +0000 [r307467] Mark Michelson <mmichelson@digium.com>
+
+ * configs/ccss.conf.sample: Fix a gaffe in the CCSS sample
+ configuration. Discovered by Philippe Lindheimer and pointed out
+ on #asterisk-dev
+
+2011-02-09 21:44 +0000 [r307314] Andrew Latham <lathama@gmail.com>
+
+ * contrib/init.d/rc.debian.asterisk: Disable color during running
+ test (closes issue #18776) Reported by: alecdavis Patches:
+ ast_deb_init.diff uploaded by lathama (license 1028) Tested by:
+ andrel, lathama
+
+2011-02-09 21:06 +0000 [r307228-307273] Jeff Peeler <jpeeler@digium.com>
+
+ * main/astobj2.c: Add missing debug info for ao2_link for use with
+ REF_DEBUG in ao2 callback. (closes issue #18758) Reported by:
+ rgagnon Patches: branch-1.8-r306540-astobj-fix.diff uploaded by
+ rgagnon (license 1202) trunk-r306540-astobj-fix.diff uploaded by
+ rgagnon (license 1202)
+
+ * /, main/features.c: Merged revisions 307227 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r307227 | jpeeler | 2011-02-09 13:52:12 -0600 (Wed, 09 Feb 2011)
+ | 11 lines Make sure to set parking dial context for non-default
+ parking lots. Since parking_con_dial isn't settable, set all
+ parking lots to "park-dial". (closes issue #17946) Reported by:
+ bluecrow76 Patches:
+ asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff uploaded by
+ bluecrow76 (license 270) modified by me ........
+
+2011-02-09 05:39 +0000 [r307142] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/lock.c: Initialize tracking variable in structure properly.
+ Fixes a memory leak. (Reported by The_Boy_Wonder on IRC, fixed by
+ me.)
+
+2011-02-08 21:24 +0000 [r307092] Jason Parker <jparker@digium.com>
+
+ * main/logger.c: Fix issue with verbose messages not showing on
+ remote console. This code was reworked recently, and since the
+ logchannel list hadn't been created yet at this point, and it was
+ a verbose message, it was being dropped on the floor. Now it'll
+ continue on to where it should be handled. (closes issue #18580)
+ Reported by: pabelanger
+
+2011-02-08 21:13 +0000 [r307065] Mark Michelson <mmichelson@digium.com>
+
+ * main/ccss.c: Add a couple of useful channel variables for the CC
+ recall macro. CC_EXTEN and CC_CONTEXT will allow you to determine
+ the channel and context that will be called when the recall
+ occurs.
+
+2011-02-08 20:22 +0000 [r306999] Andrew Latham <lathama@gmail.com>
+
+ * doc/asterisk.sgml, doc/asterisk.8, configs/asterisk.conf.sample,
+ configs/voicemail.conf.sample: Documentation Updates Note default
+ polling setting in voicemail.conf Add missing config to
+ asterisk.conf Update manpage (issue #16505) Reported by: tzafrir
+ Patches: asterisk_sgml_fixes_demo.diff uploaded by tzafrir
+ (license 46) Tested by: lathama, tzafrir
+
+2011-02-08 20:18 +0000 [r306979] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 306973 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r306973 | twilson | 2011-02-08 12:14:09 -0800
+ (Tue, 08 Feb 2011) | 9 lines Merged revisions 306972 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r306972 | twilson | 2011-02-08 12:05:13 -0800 (Tue, 08
+ Feb 2011) | 2 lines Fix comparison for REFER Replaces tags with
+ pedantic=yes ........ ................
+
+2011-02-08 19:41 +0000 [r306866-306967] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 306966 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r306966 | jpeeler | 2011-02-08 13:41:21 -0600
+ (Tue, 08 Feb 2011) | 9 lines Merged revisions 306965 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r306965 | jpeeler | 2011-02-08 13:40:58 -0600 (Tue, 08
+ Feb 2011) | 1 line fix this line again ........ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 306961 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r306961 | jpeeler | 2011-02-08 13:25:10 -0600
+ (Tue, 08 Feb 2011) | 15 lines Merged revisions 306960 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r306960 | jpeeler | 2011-02-08 13:18:50 -0600 (Tue, 08 Feb 2011)
+ | 9 lines Backup file storing message duration is not used with
+ IMAP_STORAGE, remove code. The message duration is stored in the
+ body of the email when using IMAP_STORAGE, so nothing needs to
+ happen with the backup file. (closes issue #18718) Reported by:
+ kerframil ........ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 306865 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r306865 | jpeeler | 2011-02-08 10:21:25 -0600
+ (Tue, 08 Feb 2011) | 9 lines Merged revisions 306864 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r306864 | jpeeler | 2011-02-08 10:19:17 -0600 (Tue, 08
+ Feb 2011) | 1 line make this safer and fully correct, pointed out
+ by Steve Davis ........ ................
+
+2011-02-08 01:45 +0000 [r306826] Andrew Latham <lathama@gmail.com>
+
+ * UPGRADE.txt, include/asterisk/manager.h, doc/asterisk.sgml,
+ include/asterisk/doxygen/mantisworkflow.h: Documentation Updates.
+ More updates to the removed doc folder and start updates to the
+ man page. (issue #16505) Reported by: tzafrir Tested by: lathama
+
+2011-02-07 22:43 +0000 [r306619-306674] Terry Wilson <twilson@digium.com>
+
+ * /, main/features.c: Merged revisions 306673 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r306673 | twilson | 2011-02-07 14:40:20 -0800
+ (Mon, 07 Feb 2011) | 17 lines Merged revisions 306672 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r306672 | twilson | 2011-02-07 14:35:20 -0800 (Mon, 07 Feb 2011)
+ | 10 lines Don't try to pickup a call in the middle of a
+ masquerade If A calls B which doesn't answer and C & D both try
+ to do a call pickup, it is possible for ast_pickup_call to answer
+ the call, then fail to masquerade one of the calls because the
+ other one is already in the process of masquerading. This patch
+ checks to see if the channel is in the process of masquerading
+ before call before selecting it for a pickup. Review:
+ https://reviewboard.asterisk.org/r/1094/ ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 306618 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r306618 | twilson | 2011-02-07 13:59:54 -0800
+ (Mon, 07 Feb 2011) | 17 lines Merged revisions 306617 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r306617 | twilson | 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011)
+ | 10 lines Don't allow a REFER w/replaces to replace its own
+ dialog Asterisk currently accepts a REFER with a Refer-To with an
+ embedded Replaces header that matches the dialog of the REFER.
+ This would be a situation like A calls B, A calls C, A transfers
+ B to A, which is just silly. This patch makes the transfer fail
+ instead of making Asterisk freak out and forget to hang other
+ channels up. Review: https://reviewboard.asterisk.org/r/1093/
+ ........ ................
+
+2011-02-07 17:36 +0000 [r306575] Mark Michelson <mmichelson@digium.com>
+
+ * main/ccss.c: Rearrange a bit of code in the generic CC recall
+ operation. By waiting to call the callback macro after the
+ CC_INTERFACES, extension, priority, and context have been set,
+ this information can be accessed more easily within the callback
+ macro. Reported by Philippe Lindheimer.
+
+2011-02-04 19:24 +0000 [r306356] Jason Parker <jparker@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 306346 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r306346 | qwell | 2011-02-04 13:21:43 -0600 (Fri, 04 Feb 2011) |
+ 9 lines Don't fallthrough to 'unknown' in the 'ringing' case.
+ This could cause improper exits from the queue. (closes issue
+ #18499) Reported by: zaltar Patches: app_queue.patch uploaded by
+ zaltar (license 1148) ........
+
+2011-02-04 18:53 +0000 [r306324] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_queue.c, apps/app_dial.c: Don't send redirecting updates
+ to the caller if the dialplan forked the call. Each fork in the
+ dial could be redirected and confuse the caller. For ISDN the
+ DivLeg1 and DivLeg3 messages would get confused because ISDN
+ redirects calls in sequence not in parallel. * Also fixed a
+ formatting inconsistency in app_dial.c and make a warning message
+ more useful about what frame type could not be written.
+
+2011-02-03 23:49 +0000 [r306215] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_sip.c: Fix SIP deadlock involving state changes.
+ Once again a call to pbx_builtin_getvar_helper (and
+ pbx_builtin_setvar_helper) has caused locking problems. Both of
+ these functions lock the channel when the channel argument is
+ passed in! In this case, the suspected problem (the backtrace
+ makes it impossible to tell) was the private being locked in
+ sip_set_rtp_peer and then: transmit_reinvite_with_sdp
+ try_suggested_sip_codec pbx_builtin_getvar_helper (Traced to
+ verify that the fix was only required in 1.8 and later.) (closes
+ issue #18491) Reported by: cmaj Patches:
+ chan_sip_fix_deadlocks_bug_18491.txt uploaded by cmaj (license
+ 830) Tested by: cmaj
+
+2011-02-03 21:03 +0000 [r306127] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 306126 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r306126 | twilson | 2011-02-03 12:56:00 -0800
+ (Thu, 03 Feb 2011) | 16 lines Merged revisions 306119 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r306119 | twilson | 2011-02-03 12:36:34 -0800 (Thu, 03 Feb 2011)
+ | 9 lines Set hangup cause in local_hangup When a call involves a
+ local channel (like SIP -> Local -> SIP), the hangup cause was
+ not being set. This resulted in SIP channels sometimes getting a
+ 503 error instead of a 486 when the far side sent a busy. In
+ Asterisk 1.8+ this also can cause issues with CCSS that involve a
+ local channel. This patch sets the hangupcause for one side of
+ the local channel to the other in local_hangup for outbound
+ calls. ........ ................
+
+2011-02-03 20:50 +0000 [r306124] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 306123 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r306123 | jpeeler | 2011-02-03 14:49:48 -0600 (Thu, 03 Feb 2011)
+ | 10 lines Set exception on channel in parking thread when
+ POLLPRI event detected. This is done just to make the code be
+ equivalent to the old select code. As noted in 303106 the same
+ issue was already fixed in this branch, but the exception was not
+ set on the channel in the case of POLLPRI. The reason that this
+ did not cause a problem here is because in 122923 the check in
+ __ast_read to check the exception flag was removed. (related to
+ #18637) ........
+
+2011-02-03 15:50 +0000 [r305987] Andrew Latham <lathama@gmail.com>
+
+ * phoneprov/snom-mac.xml (added), configs/phoneprov.conf.sample, /:
+ res_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 support
+ (issue #18713) Reported by: lathama Patches: snom_dir.diff
+ uploaded by lathama (license 1028) Tested by: lathama
+
+2011-02-03 00:24 +0000 [r305923] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, main/manager.c, /, channels/chan_sip.c,
+ apps/app_sendtext.c: Merged revisions 305889 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r305889 | rmudgett | 2011-02-02 18:15:07 -0600
+ (Wed, 02 Feb 2011) | 17 lines Merged revisions 305888 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011)
+ | 8 lines Minor AST_FRAME_TEXT related issues. * Include the null
+ terminator in the buffer length. When the frame is queued it is
+ copied. If the null terminator is not part of the frame buffer
+ length, the receiver could see garbage appended onto it. * Add
+ channel lock protection with ast_sendtext(). * Fixed AMI SendText
+ action ast_sendtext() return value check. ........
+ ................
+
+2011-02-02 20:05 +0000 [r305844] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * funcs/func_env.c: Eliminate a file descriptor leak when using the
+ FILE() dialplan function. (closes issue #18731) Reported by:
+ marioabajo
+
+2011-02-02 19:27 +0000 [r305753-305838] Andrew Latham <lathama@gmail.com>
+
+ * apps/app_externalivr.c, configs/sip.conf.sample,
+ configs/skinny.conf.sample, configs/h323.conf.sample,
+ configs/sla.conf.sample, apps/app_voicemail.c,
+ configs/iax.conf.sample, funcs/func_enum.c,
+ configs/dundi.conf.sample, funcs/func_callcompletion.c,
+ configs/mgcp.conf.sample, configs/iaxprov.conf.sample,
+ configs/unistim.conf.sample: Replacing doc/* and asterisk.pdf
+ with wiki links Adding links to http(s)://wiki.asterisk.org
+
+ * configs/ccss.conf.sample, configs/sip.conf.sample,
+ configs/skinny.conf.sample, main/config.c,
+ configs/h323.conf.sample, configs/sla.conf.sample,
+ main/ast_expr2.fl, res/res_srtp.c,
+ configs/chan_dahdi.conf.sample, configs/extconfig.conf.sample,
+ configs/res_snmp.conf.sample, main/ast_expr2f.c,
+ res/res_timing_dahdi.c: Replacing doc/* with wiki links Adding
+ links to http(s)://wiki.asterisk.org
+
+ * channels/chan_sip.c: Replace link to old doc with new wiki page.
+ Link to
+ https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
+
+2011-02-01 22:48 +0000 [r305692] Jason Parker <jparker@digium.com>
+
+ * channels/chan_iax2.c: Reverse sense of an error test when reading
+ from astdb. (closes issue #18545) Reported by: jcovert Patches:
+ chan_iax2.c.patch uploaded by jcovert (license 551)
+
+2011-02-01 21:14 +0000 [r305649] Andrew Latham <lathama@gmail.com>
+
+ * configs/sip.conf.sample: SIP Configuration Documentation sip show
+ settings reports qualifyfreq in milliseconds. sip.conf configures
+ qualifyfreg in seconds.
+
+2011-02-01 19:23 +0000 [r305603] Brett Bryant <bbryant@digium.com>
+
+ * cel/cel_pgsql.c: Add a possible solution to a customer problem
+ with reloading cel_pgsql.so quickly.
+
+2011-02-01 18:02 +0000 [r305560] Andrew Latham <lathama@gmail.com>
+
+ * CHANGES, Makefile, README, /: doc/tex dir removed, but
+ corresponding entries still exists Update README, CHANGES, and
+ Makefile. Direct users to http://wiki.asterisk.org for
+ documentation or to the AST.txt and AST.pdf included in the
+ tarball. (closes issue #18443) Reported by: bas Patches:
+ changes.diff uploaded by lathama (license 1028) readme.diff
+ uploaded by lathama (license 1028) Tested by: lathama bas
+
+2011-02-01 17:04 +0000 [r305473] Jason Parker <jparker@digium.com>
+
+ * res/res_musiconhold.c, /: Merged revisions 305472 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r305472 | qwell | 2011-02-01 11:02:09 -0600
+ (Tue, 01 Feb 2011) | 16 lines Merged revisions 305471 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r305471 | qwell | 2011-02-01 11:00:55 -0600 (Tue, 01 Feb 2011) |
+ 9 lines Close file descriptor for timing source when a MOH class
+ gets destroyed. (closes issue #18457) Reported by: mcallist
+ Patches: 18457-closetimer.diff uploaded by qwell (license 4)
+ 18457-closetimer_trunk.diff uploaded by qwell (license 4) Tested
+ by: qwell, loloski ........ ................
+
+2011-02-01 00:01 +0000 [r305343] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c, /: Merged revisions 305342 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r305342 | rmudgett | 2011-01-31 17:50:10 -0600
+ (Mon, 31 Jan 2011) | 14 lines Merged revisions 305341 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r305341 | rmudgett | 2011-01-31 17:45:58 -0600 (Mon, 31 Jan 2011)
+ | 7 lines Obtain the pri lock for PRI queue counters. Need to
+ obtain the pri lock when calling pri_dump_info_str() to avoid a
+ reentrancy problem when calculating the Q.921 Q count statistic.
+ JIRA AST-484 ........ ................
+
+2011-01-31 23:07 +0000 [r305131-305254] Jason Parker <jparker@digium.com>
+
+ * apps/app_dial.c, /, channels/chan_sip.c: Merged revisions 305253
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r305253 | qwell | 2011-01-31 16:59:34 -0600
+ (Mon, 31 Jan 2011) | 17 lines Merged revisions 305252 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) |
+ 10 lines Prevent a crash when dialing a technology with no
+ destination (ex: Dial(SIP/)) chan_iax2 and other channel drivers
+ already had code to prevent this. The attempt that app_dial was
+ making to prevent it was not correct, so I fixed that. (closes
+ issue #18371) Reported by: gbour Patches: 18371.patch uploaded by
+ gbour (license 1162) ........ ................
+
+ * configs/sip.conf.sample, main/tcptls.c: Add alternative name for
+ config option. The SIP sample configuration had "tlscadir" as the
+ option name, but chan_sip used the more correct "tlscapath". Now
+ both are accepted. Discovered (sort of) by a user on IRC in
+ #asterisk
+
+ * res/res_musiconhold.c: Fix compile error. pseudofd no longer
+ exists.
+
+ * res/res_musiconhold.c, /: Merged revisions 305130 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r305130 | qwell | 2011-01-31 14:59:37 -0600
+ (Mon, 31 Jan 2011) | 9 lines Merged revisions 305129 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r305129 | qwell | 2011-01-31 14:56:25 -0600 (Mon, 31 Jan
+ 2011) | 2 lines Set file descriptors to -1 on creation, so that
+ we don't see weirdness later. ........ ................
+
+2011-01-31 13:56 +0000 [r305083] Andrew Latham <lathama@gmail.com>
+
+ * main/http.c: Asterisk HTTP response Content-type Address content
+ type for BSD and other platforms (closes issue #18456) Reported
+ by: alexo Patches: asterisk18_http.patch uploaded by alexo
+ (license 1175) Tested by: alexo
+
+2011-01-31 07:51 +0000 [r304950-305040] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * include/asterisk/lock.h: Use the non-specific API aliases, to
+ avoid a problem with building the utils directory.
+
+ * apps/app_voicemail.c, /: Merged revisions 304978 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r304978 | tilghman | 2011-01-31 01:25:14 -0600
+ (Mon, 31 Jan 2011) | 9 lines Merged revisions 304952 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r304952 | tilghman | 2011-01-31 00:54:45 -0600 (Mon, 31
+ Jan 2011) | 2 lines Fix compilation when ODBC_STORAGE is defined.
+ ........ ................
+
+ * main/utils.c, include/asterisk/lock.h, .cleancount, main/lock.c,
+ main/heap.c: Change mutex tracking so that it only consumes
+ memory in the core mutex object when it's actually being used.
+ This reduces the overall size of a mutex which was 3016 bytes
+ before this back down to 216 bytes (this is on 64-bit Linux with
+ a glibc-implemented mutex). The exactness of the numbers here may
+ vary slightly based upon how mutexes are implemented on a
+ platform, but the long and short of it is that prior to this
+ commit, chan_iax2 held down 98MB of memory on a 64-bit system for
+ nothing more than a table of 32767 locks. After this commit, the
+ same table occupies a mere 7MB of memory. (closes issue #18194)
+ Reported by: job Patches: 20110124__issue18194.diff.txt uploaded
+ by tilghman (license 14) Tested by: tilghman Review:
+ https://reviewboard.asterisk.org/r/1066
+
+2011-01-30 00:11 +0000 [r304908] Andrew Latham <lathama@gmail.com>
+
+ * apps/app_externalivr.c, apps/app_queue.c, apps/app_voicemail.c,
+ funcs/func_realtime.c, res/res_calendar.c,
+ funcs/func_callcompletion.c: Add Function and Application
+ Relationships to documentation Add and extend the see-also
+ sections to the documentation for applications and functions in
+ an effort to expand the online documentation of the wiki. Also
+ check for and update any links to moved documentation in the doc
+ folder.
+
+2011-01-29 23:07 +0000 [r304638-304866] Sean Bright <sean@malleable.com>
+
+ * res/res_config_ldap.c, /: Merged revisions 304865 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r304865 | seanbright | 2011-01-29 18:05:25 -0500 (Sat,
+ 29 Jan 2011) | 7 lines Plug some memory leaks in the LDAP
+ realtime driver. (closes issue #18435) Reported by: zaltar
+ Patches: res_config_ldap.patch uploaded by zaltar (license 1148)
+ ........
+
+ * /, apps/app_meetme.c: Merged revisions 304776 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r304776 | seanbright | 2011-01-29 13:08:14 -0500 (Sat, 29 Jan
+ 2011) | 15 lines If we fail to allocate our announcement objects,
+ make sure we don't leak objects. The majority of this patch was
+ committed already in r304726 and r304729. (issue #18225) Reported
+ by: kenji (issue #18444) Reported by: junky (closes issue #18343)
+ Reported by: kobaz Patches: meetme-refs.diff uploaded by kobaz
+ (license 834) ........
+
+ * /, apps/app_meetme.c: Merged revisions 304773 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r304773 | seanbright | 2011-01-29 12:51:28 -0500 (Sat, 29 Jan
+ 2011) | 9 lines When we pass the S() or L() options to MeetMe,
+ make sure that we honor C as well. Without this patch, if the
+ user was kicked from the conference via the S() or L() mechanism,
+ we would just hang up on them even if we also passed C (continue
+ in dialplan when kicked). With this patch we honor the C flag in
+ those cases. (closes issue #17317) Reported by: var ........
+
+ * /, apps/app_meetme.c: Merged revisions 304729 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r304729 | seanbright | 2011-01-29 12:01:51 -0500 (Sat, 29 Jan
+ 2011) | 15 lines Make sure that we unref the correct object when
+ ejecting the most recent caller. Currently, when we kick the last
+ user to enter, we decrement our own reference count which results
+ in a crash when we kick another user or when we exit the
+ conference ourselves. This will fix #18225 in 1.8 and trunk, but
+ that particular bug does not exist in 1.6.2. (closes issue
+ #18225) Reported by: kenji Patches: issue18225.patch uploaded by
+ seanbright (license 71) Tested by: seanbright ........
+
+ * /, apps/app_meetme.c: Merged revisions 304726 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r304726 | seanbright | 2011-01-29 11:26:57 -0500 (Sat, 29 Jan
+ 2011) | 9 lines Fix user reference leak in MeetMe. We were
+ unlinking the user from the conferences user container, but not
+ decrementing the reference count of the user as well, resulting
+ in a leak. (closes issue #18444) Reported by: junky Tested by:
+ seanbright ........
+
+ * /, apps/app_meetme.c: Merged revisions 304659,304682 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r304659 | seanbright | 2011-01-28 16:22:09 -0500 (Fri,
+ 28 Jan 2011) | 5 lines Don't leak references if we can't create a
+ pseudo channel for mixing in MeetMe. If there was a problem
+ allocating a pseudo channel when building our meetme, we weren't
+ destroying our user container or destroying the mutexes that we
+ created. ........ r304682 | seanbright | 2011-01-28 17:38:05
+ -0500 (Fri, 28 Jan 2011) | 2 lines Revert part of the previous
+ commit that snuck in. ........
+
+ * main/acl.c: Restore some conditionals that we lost in r277814.
+ There are some cases where ast_append_ha() is called with a NULL
+ instead of a valid int pointer. So if we get a NULL, don't try to
+ dereference it. (closes issue #18162) Reported by: imcdona
+ Patches: issue0018162.patch uploaded by pabelanger (license 224)
+ Tested by: enegaard
+
+2011-01-27 19:08 +0000 [r304554] Richard Mudgett <rmudgett@digium.com>
+
+ * main/ccss.c: Warning message if CALLCOMPLETION(cc_callback_macro
+ or cc_agent_dialstring) are empty. Test if the value pointer is
+ not NULL instead of not ast_strlen_zero().
+
+2011-01-27 17:03 +0000 [r304462-304466] Jason Parker <jparker@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 304465 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r304465 | qwell | 2011-01-27 11:01:24 -0600
+ (Thu, 27 Jan 2011) | 16 lines Merged revisions 304464 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r304464 | qwell | 2011-01-27 10:57:46 -0600 (Thu, 27 Jan 2011) |
+ 9 lines Fix default prefix=/usr regression on non-Linux systems.
+ This partially reverts a change made in branches/1.4/ r267759,
+ which will cause issue #17013 to be reopened. This issue was
+ pointed out by a user on #asterisk, who helpfully discovered that
+ paths were being set incorrectly. To truly understand what was
+ wrong, one should run: svn diff --force -c<this revision>
+ configure ........ ................
+
+ * /, configure: Merged revisions 304461 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r304461 | qwell | 2011-01-27 10:48:00 -0600
+ (Thu, 27 Jan 2011) | 9 lines Merged revisions 304460 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r304460 | qwell | 2011-01-27 10:47:03 -0600 (Thu, 27 Jan
+ 2011) | 1 line Rerun bootstrap.sh with no changes, so that it is
+ more obvious what my next commit changes. ........
+ ................
+
+2011-01-26 22:27 +0000 [r304339] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 304338 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r304338 | jpeeler | 2011-01-26 16:26:37 -0600 (Wed, 26 Jan 2011)
+ | 2 lines Change delimiter used internally for GOTO_ON_BLINDXFR
+ to commas to match 76703. ........
+
+2011-01-26 21:02 +0000 [r304251] Mark Michelson <mmichelson@digium.com>
+
+ * main/udptl.c, /: Merged revisions 304250 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r304250 | mmichelson | 2011-01-26 15:02:10 -0600
+ (Wed, 26 Jan 2011) | 9 lines Merged revisions 304242 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r304242 | mmichelson | 2011-01-26 14:38:37 -0600 (Wed,
+ 26 Jan 2011) | 3 lines Get rid of unused 'verbose' field in
+ ast_udptl ........ ................
+
+2011-01-26 20:43 +0000 [r304245] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/sip/include/sip.h,
+ channels/sip/include/reqresp_parser.h, /, channels/chan_sip.c,
+ channels/sip/reqresp_parser.c: Merged revisions 304244 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r304244 | mnicholson | 2011-01-26 14:42:16 -0600
+ (Wed, 26 Jan 2011) | 13 lines Merged revisions 304241 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r304241 | mnicholson | 2011-01-26 14:38:22 -0600 (Wed, 26 Jan
+ 2011) | 6 lines This patch modifies chan_sip to route responses
+ to the address the request came from. It also modifies chan_sip
+ to respect the maddr parameter in the Via header. ABE-2664
+ Review: https://reviewboard.asterisk.org/r/1059/ ........
+ ................
+
+2011-01-26 20:23 +0000 [r304186] Sean Bright <sean@malleable.com>
+
+ * /, configs/queues.conf.sample: Merged revisions 304181 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r304181 | seanbright | 2011-01-26 15:22:47 -0500
+ (Wed, 26 Jan 2011) | 9 lines Merged revisions 304159 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r304159 | seanbright | 2011-01-26 15:18:29 -0500 (Wed,
+ 26 Jan 2011) | 1 line Make sure the sample queues.conf is
+ properly commented. ........ ................
+
+2011-01-26 19:39 +0000 [r304150] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 304149 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r304149 | rmudgett | 2011-01-26 13:38:38 -0600
+ (Wed, 26 Jan 2011) | 9 lines Merged revisions 304148 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r304148 | rmudgett | 2011-01-26 13:23:46 -0600 (Wed,
+ 26 Jan 2011) | 2 lines Update documentation for
+ DAHDISendCallreroutingFacility() application. ..........
+ ................
+
+2011-01-26 01:26 +0000 [r304097] Sean Bright <sean@malleable.com>
+
+ * /, main/file.c: Merged revisions 304096 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r304096 | seanbright | 2011-01-25 20:24:58 -0500 (Tue, 25 Jan
+ 2011) | 12 lines Per the man page, setvbuf() must be called
+ before any other operation on an open file. We use setvbuf() to
+ associate a buffer with a stream, but we have already written to
+ the open file. This works (by chance) on Linux, but fails on
+ other platforms, such as OpenSolaris. (closes issue #16610)
+ Reported by: bklang Patches: setvbuf.patch uploaded by crjw
+ (license 963) Tested by: bklang, asgaroth, efutch ........
+
+2011-01-25 23:28 +0000 [r304007] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Merged revisions 304006 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r304006 | rmudgett | 2011-01-25 17:25:32 -0600
+ (Tue, 25 Jan 2011) | 15 lines Merged revisions 304005 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r304005 | rmudgett | 2011-01-25 17:21:09 -0600 (Tue, 25 Jan 2011)
+ | 8 lines DTMF attended transfers sometimes fail for no apparent
+ reason. The loop in feature_request_and_dial() can exit when
+ Party C has answered without processing an AST_CONTROL_ANSWER.
+ Also sometimes an AST_CONTROL_ANSWER never happens even though
+ Party C has answered. Don't hangup Party C if he is up or we
+ receive an AST_CONTROL_ANSWER. ........ ................
+
+2011-01-25 22:09 +0000 [r303962] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 303960 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r303960 | twilson | 2011-01-25 16:02:42 -0600
+ (Tue, 25 Jan 2011) | 23 lines Merged revisions 303906 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r303906 | twilson | 2011-01-25 14:50:59 -0600 (Tue, 25 Jan 2011)
+ | 16 lines Guard against retransmitting BYEs indefinitely In the
+ case of an attended transfer (A calls B, A atxfers to C) where A
+ becomes unreachable before replying to Asterisk's BYE, Asterisk
+ can sometimes retransmit the BYE indefinitely. This is because
+ __sip_autodestruct tests p->refer && !ast_test_flag(&p->flags[0],
+ SIP_ALREADYGONE and will then transmit a BYE. When this BYE times
+ out, it will not ever be marked as ALREADYGONE, so when
+ __sip_autodestruct is called again, we end up starting the cycle
+ over. This patch adds a call to sip_alreadygone(pkt->owner) in
+ retrans_pkt in the case of a BYE that has timed out. This should
+ prevent Asterisk from trying to transmit new BYE messages in the
+ future. Review: https://reviewboard.asterisk.org/r/1077/ ........
+ ................
+
+2011-01-25 20:56 +0000 [r303907] Matthew Nicholson <mnicholson@digium.com>
+
+ * include/asterisk/res_fax.h, res/res_fax.c: Reimplemented fax
+ session reservation to reverse the ABI breakage introduced in
+ r297486.
+
+2011-01-25 18:55 +0000 [r303860] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, channels/chan_sip.c: Merged revisions 303858 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r303858 | tilghman | 2011-01-25 12:41:26 -0600 (Tue, 25 Jan 2011)
+ | 5 lines Fix "sip show user <tab>", so that it actually shows
+ results, instead of just completing the last entry. (closes issue
+ #16675) Reported by: pj ........
+
+2011-01-25 17:49 +0000 [r303771] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c, channels/sig_ss7.c, channels/sig_pri.h,
+ channels/chan_dahdi.c, channels/sig_ss7.h, /: Merged revisions
+ 303769 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600
+ (Tue, 25 Jan 2011) | 47 lines Merged revisions 303765 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011)
+ | 40 lines Sending out unnecessary PROCEEDING messages breaks
+ overlap dialing. Issue #16789 was a good idea. Unfortunately, it
+ breaks overlap dialing through Asterisk. There is not enough
+ information available at this point to know if dialing is
+ complete. The ast_exists_extension(), ast_matchmore_extension(),
+ and ast_canmatch_extension() calls are not adequate to detect a
+ dial through extension pattern of "_9!". Workaround is to use the
+ dialplan Proceeding() application early in non-dial through
+ extensions. * Effectively revert issue #16789. * Allow outgoing
+ overlap dialing to hear dialtone and other early media. A
+ PROGRESS "inband-information is now available" message is now
+ sent after the SETUP_ACKNOWLEDGE message for non-digital calls.
+ An AST_CONTROL_PROGRESS is now generated for incoming
+ SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of
+ the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent
+ with the cause codes. * Added better protection from sending out
+ of sequence messages by combining several flags into a single
+ enum value representing call progress level. * Added diagnostic
+ messages for deferred overlap digits handling corner cases.
+ (closes issue #17085) Reported by: shawkris (closes issue #18509)
+ Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch
+ uploaded by rmudgett (license 664) Expanded upon
+ issue18509_early_media_v1.8_v3.patch to include analog and SS7
+ because of backporting requirements. Tested by: wimpy, rmudgett
+ ........ ................
+
+2011-01-25 17:02 +0000 [r303678] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 303677 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r303677 | jpeeler | 2011-01-25 10:59:28 -0600
+ (Tue, 25 Jan 2011) | 26 lines Merged revisions 303676 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r303676 | jpeeler | 2011-01-25 10:58:29 -0600 (Tue, 25 Jan 2011)
+ | 20 lines Fix voicemail sequencing for file based storage. A
+ previous change was made to account for when the number of
+ voicemail messages exceeds the max limit to be handled properly,
+ but it caused gaps in the messages to not be properly handled.
+ This has now been resolved. In later non 1.4 branches, it appears
+ that resequencing wasn't even occurring due from what appears and
+ accidental code removal. (closes issue #18498) Reported by:
+ JJCinAZ Patches: bug18498v2.patch uploaded by jpeeler (license
+ 325) (closes issue #18486) Reported by: bluefox Patches:
+ bug18486.patch uploaded by jpeeler (license 325) ........
+ ................
+
+2011-01-24 20:51 +0000 [r303549] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/channel.h, main/channel.c, main/pbx.c, /,
+ apps/app_meetme.c, main/features.c: Merged revisions 303548 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r303548 | russell | 2011-01-24 14:49:53 -0600
+ (Mon, 24 Jan 2011) | 38 lines Merged revisions 303546 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011)
+ | 31 lines Fix channel redirect out of MeetMe() and other issues
+ with channel softhangup. Mantis issue #18585 reports that a
+ channel redirect out of MeetMe() stopped working properly. This
+ issue includes a patch that resolves the issue by removing a call
+ to ast_check_hangup() from app_meetme.c. I left that in my patch,
+ as it doesn't need to be there. However, the rest of the patch
+ fixes this problem with or without the change to app_meetme. The
+ key difference between what happens before and after this patch
+ is the effect of the END_OF_Q control frame. After END_OF_Q is
+ hit in ast_read(), ast_read() will return NULL. With the
+ ast_check_hangup() removed, app_meetme sees this which causes it
+ to exit as intended. Checking ast_check_hangup() caused
+ app_meetme to exit earlier in the process, and the target of the
+ redirect saw the condition where ast_read() returned NULL.
+ Removing ast_check_hangup() works around the issue in app_meetme,
+ but doesn't solve the issue if another application did the same
+ thing. There are also other edge cases where if an application
+ finishes at the same time that a redirect happens, the target of
+ the redirect will think that the channel hung up. So, I made some
+ changes in pbx.c to resolve it at a deeper level. There are
+ already places that unset the SOFTHANGUP_ASYNCGOTO flag in an
+ attempt to abort the hangup process. My patch extends this to
+ remove the END_OF_Q frame from the channel's read queue, making
+ the "abort hangup" more complete. This same technique was used in
+ every place where a softhangup flag was cleared. (closes issue
+ #18585) Reported by: oej Tested by: oej, wedhorn, russell Review:
+ https://reviewboard.asterisk.org/r/1082/ ........
+ ................
+
+2011-01-24 17:20 +0000 [r303467] Jason Parker <jparker@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 303285 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r303285 | qwell | 2011-01-21 15:48:09 -0600
+ (Fri, 21 Jan 2011) | 15 lines Merged revisions 303284 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) |
+ 8 lines Reset configuration before parsing users.conf. Some
+ values configured in chan_dahdi.conf were able to leak in to
+ users.conf configuration. This was surprising users, and
+ potentially setting non-sane "defaults". ASTNOW-125 ........
+ ................
+
+2011-01-21 23:11 +0000 [r303286-303375] Jason Parker <jparker@digium.com>
+
+ * channels/chan_dahdi.c, /: Temporarily revert r303286
+
+ * channels/chan_dahdi.c, /: Merged revisions 303285 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r303285 | qwell | 2011-01-21 15:48:09 -0600
+ (Fri, 21 Jan 2011) | 15 lines Merged revisions 303284 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) |
+ 8 lines Reset configuration before parsing users.conf. Some
+ values configured in chan_dahdi.conf were able to leak in to
+ users.conf configuration. This was surprising users, and
+ potentially setting non-sane "defaults". ASTNOW-125 ........
+ ................
+
+2011-01-20 20:31 +0000 [r303153] Richard Mudgett <rmudgett@digium.com>
+
+ * main/ccss.c: Merged revision 303098 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r303098 | rmudgett | 2011-01-20 12:11:45 -0600 (Thu,
+ 20 Jan 2011) | 15 lines CC_INTERFACES does not get built
+ correctly with local channels. If local channels are used with
+ CCSS, CC_INTERFACES gets garbage prepended to it so the CC recall
+ fails. Also CC_INTERFACES gets "&(null)" appended to it. *
+ Initialize the buffer to eliminate the prepended garbage. *
+ Filter out the empty interface strings to eliminate the latter. *
+ Added a diagnostic message if the CC_INTERFACES is ever empty.
+ JIRA ABE-2740 JIRA SWP-2848 ..........
+
+2011-01-20 19:57 +0000 [r303107] Shaun Ruffell <sruffell@digium.com>
+
+ * /, main/features.c: Merged revisions 303106 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r303106 | sruffell | 2011-01-20 13:56:34 -0600 (Thu, 20 Jan 2011)
+ | 15 lines main/features: Use POLLPRI when waiting for events on
+ parked channels. This change resolves a regression in the 1.6.2
+ when converting from select to poll. The DAHDI timers use POLLPRI
+ to indicate that the timer fired, but features was not waiting
+ for that flag. The result was no audio for MOH when a call was
+ parked and res_timing_dahdi was in use. This patch is slightly
+ modified from the one on the mantis issue. It does not set an
+ exception on the channel if the POLLPRI flag is set. (closes
+ issue #18262) Reported by: francesco_r Patches:
+ patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029)
+ Tested by: francesco_r, rfrantik, one47 ........
+
+2011-01-20 17:10 +0000 [r303009] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_queue.c, /, configs/queues.conf.sample: Merged revisions
+ 303008 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r303008 | jpeeler | 2011-01-20 11:07:44 -0600
+ (Thu, 20 Jan 2011) | 14 lines Merged revisions 303007 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r303007 | jpeeler | 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011)
+ | 8 lines Add new queue strategy to preserve behavior for when
+ queue members moved to ao2. Add queue strategy called "rrordered"
+ to mimic old behavior from when queue members were stored in a
+ linked list. ABE-2707 ........ ................
+
+2011-01-20 16:12 +0000 [r302921] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_privacy.c: Merged revisions 302920 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r302920 | russell | 2011-01-20 10:11:58 -0600 (Thu, 20 Jan 2011)
+ | 2 lines Resolve a compiler warning. ........
+
+2011-01-20 15:45 +0000 [r302918] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 302917 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r302917 | lmadsen | 2011-01-20 09:42:05 -0600 (Thu, 20 Jan 2011)
+ | 8 lines Option L() is milliseconds, not seconds. > Change the
+ verbose output of option L() to say milliseconds and not seconds
+ > as the value is in milliseconds. > > (closes issue #18264) >
+ Reported by: jacco > Patches: > app_dial_patch.txt uploaded by
+ lmadsen (license 10) ........
+
+2011-01-19 23:56 +0000 [r302837] Russell Bryant <russell@digium.com>
+
+ * main/manager.c: Only check container count if it exists.
+
+2011-01-19 23:49 +0000 [r302834] Sean Bright <sean@malleable.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 302833 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r302833 | seanbright | 2011-01-19 18:47:22 -0500 (Wed,
+ 19 Jan 2011) | 7 lines Support greetingsfolder as documented in
+ voicemail.conf.sample. (closes issue #17870) Reported by:
+ edhorton Patches:
+ __20100816-app_voicemail-greetingsfolder-support.txt uploaded by
+ lmadsen (license 10) ........
+
+2011-01-19 23:29 +0000 [r302831] Paul Belanger <pabelanger@digium.com>
+
+ * contrib/scripts/install_prereq: Add binutils-dev for
+ BETTER_BACKTRACES
+
+2011-01-19 23:06 +0000 [r302785-302789] Russell Bryant <russell@digium.com>
+
+ * main/manager.c, /: Merged revisions 302788 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r302788 | russell | 2011-01-19 17:06:14 -0600 (Wed, 19 Jan 2011)
+ | 4 lines Turn a noisy verbose message into a debug message. This
+ can drown your console if you're using the AMI over HTTP.
+ ........
+
+ * main/manager.c: Resolve a memory leak with the manager interface
+ is disabled. The intent of this check as it stands in previous
+ versions of Asterisk was to check if there are any active
+ sessions. If there were no sessions, then the function would
+ return immediately and not bother with queueing up the manager
+ event to be processed. Since the conversion of storing sessions
+ in an astobj2 container, this check will always pass. I changed
+ it to go back to checking what was intended. The side effect of
+ this was that if the AMI is disabled, the manager event queue is
+ populated anyway, but the code that runs to clear out the queue
+ never runs. A producer with no consumer is a bad thing. Reported
+ internally by kmorgan.
+
+2011-01-19 21:29 +0000 [r302713] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Merged revisions 302693 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r302693 | rmudgett | 2011-01-19 15:25:41 -0600
+ (Wed, 19 Jan 2011) | 22 lines Merged revisions 302671 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r302671 | rmudgett | 2011-01-19 15:21:56 -0600 (Wed, 19 Jan 2011)
+ | 15 lines DTMF transfer plays the wrong sounds for wrong number
+ or other call failure. * Set the default for features.conf.sample
+ xferfailsound option to "beeperr" as documented instead of
+ "pbx-invalid" and corrected the use of it in DTMF blind transfer
+ (#1). * Improved DTMF blind transfer handling of wrong numbers.
+ Most of the concerns in this issue were taken care of by the
+ patch for issue 17999: Issues with DTMF triggered attended
+ transfers. (closes issue #18379) Reported by: gincantalupo Tested
+ by: rmudgett ........ ................
+
+2011-01-19 21:23 +0000 [r302634-302680] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * include/asterisk/astdb.h, /: Merged revisions 302675 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r302675 | tilghman | 2011-01-19 15:22:45 -0600
+ (Wed, 19 Jan 2011) | 9 lines Merged revisions 302663 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r302663 | tilghman | 2011-01-19 15:20:28 -0600 (Wed, 19
+ Jan 2011) | 2 lines Add some API documentation ........
+ ................
+
+ * main/app.c, /: Merged revisions 302599 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r302599 | tilghman | 2011-01-19 14:13:24 -0600 (Wed, 19 Jan 2011)
+ | 15 lines Kill zombies. When we ast_safe_fork() with a non-zero
+ argument, we're expected to reap our own zombies. On a zero
+ argument, however, the zombies are only reaped when there aren't
+ any non-zero forked children alive. At other times, we accumulate
+ zombies. This code is forward ported from res_agi in 1.4, so that
+ forked children are always reaped, thus preventing an
+ accumulation of zombie processes. (closes issue #18515) Reported
+ by: ernied Patches: 20101221__issue18515.diff.txt uploaded by
+ tilghman (license 14) Tested by: ernied ........
+
+2011-01-19 20:14 +0000 [r302600] Jason Parker <jparker@digium.com>
+
+ * res/res_fax.c: Fix typo pointed out on asterisk-users list.
+
+2011-01-19 19:03 +0000 [r302505-302555] Sean Bright <sean@malleable.com>
+
+ * main/utils.c, /: Merged revisions 302554 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r302554 | seanbright | 2011-01-19 14:02:29 -0500 (Wed, 19 Jan
+ 2011) | 7 lines Don't call strlen() when we only need to look at
+ the next character or two. (closes issue #18042) Reported by:
+ wdoekes Patches: astsvn-inefficient-ast-uri-decode.patch uploaded
+ by wdoekes (license 717) ........
+
+ * /, main/features.c: Merged revisions 302551 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r302551 | seanbright | 2011-01-19 13:54:03 -0500 (Wed, 19 Jan
+ 2011) | 7 lines Remove an extraneous \r\n at the end of a parking
+ manager events. (closes issue #18363) Reported by:
+ clegall_proformatique Patches:
+ asterisk_1.8_295998_parking_manager_events_format.patch uploaded
+ by clegall proformatique (license 1139) ........
+
+ * /, res/res_agi.c: Merged revisions 302548 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r302548 | seanbright | 2011-01-19 13:37:09 -0500 (Wed, 19 Jan
+ 2011) | 10 lines Properly handle partial reads from fgets() when
+ handling AGIs. When fgets() failed with EAGAIN, we were
+ continually decrementing the available space left in our buffer,
+ resulting in botched command handling. (closes issue #16032)
+ Reported by: notahat Patches: agi_buffer_patch2.diff uploaded by
+ fnordian (license 110) ........
+
+ * main/utils.c, /: Merged revisions 302504 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r302504 | seanbright | 2011-01-19 12:56:32 -0500 (Wed, 19 Jan
+ 2011) | 7 lines Make sure that h_length is set when we
+ short-circuit out of ast_gethostbyname. (closes issue #16135)
+ Reported by: thedavidfactor Patches: utils.patch uploaded by
+ thedavidfactor (license 903) ........
+
+2011-01-19 17:09 +0000 [r302462] Paul Belanger <pabelanger@digium.com>
+
+ * /, res/res_timing_timerfd.c: Merged revisions 302461 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r302461 | pabelanger | 2011-01-19 12:08:01 -0500 (Wed,
+ 19 Jan 2011) | 2 lines Handle 'Resource temporarily unavailable'
+ error more gracefully. ........
+
+2011-01-19 15:53 +0000 [r302412-302417] Sean Bright <sean@malleable.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 302416 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r302416 | seanbright | 2011-01-19 10:52:44 -0500 (Wed, 19 Jan
+ 2011) | 9 lines Remove references to priorityjumping from the
+ sample extensions.conf. Priority jumping was removed from
+ pbx_config in r68970. (closes issue #18622) Reported by: kshumard
+ Patches: extensions.conf.sample.patch uploaded by kshumard
+ (license 92) ........
+
+ * channels/chan_sip.c: Initialize an uninitialized variable.
+ (closes issue #18640) Reported by: jcovert Patches:
+ chan_sip.c.patch uploaded by jcovert (license 551)
+
+ * channels/chan_local.c: Use appropriate type for requested format
+ in chan_local. We were passing and storing the requested format
+ as an int instead of format_t resulting in truncation. (closes
+ issue #18238) Reported by: whizemen Patches:
+ 0018238_speex16.patch uploaded by whizemen (license 1143)
+
+2011-01-18 22:04 +0000 [r302318] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: Use the expanded format type instead of plain
+ int.
+
+2011-01-18 21:43 +0000 [r302314] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 302313 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r302313 | mnicholson | 2011-01-18 15:40:03 -0600
+ (Tue, 18 Jan 2011) | 11 lines Merged revisions 302311 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r302311 | mnicholson | 2011-01-18 15:35:03 -0600 (Tue, 18 Jan
+ 2011) | 4 lines URI encode the user part of the contact header.
+ ABE-2705 ........ ................
+
+2011-01-18 20:19 +0000 [r302267] Russell Bryant <russell@digium.com>
+
+ * main/astobj2.c: Don't enable AO2_DEBUG by default if AST_DEVMODE
+ is on. AO2_DEBUG is not important and is causing a false compiler
+ warning to be generated on my Ubuntu Natty dev box.
+
+2011-01-18 20:19 +0000 [r302266] Jeff Peeler <jpeeler@digium.com>
+
+ * main/pbx.c, /: Merged revisions 302265 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r302265 | jpeeler | 2011-01-18 14:13:52 -0600 (Tue, 18 Jan 2011)
+ | 27 lines Convert device state callbacks to ao2 objects to fix a
+ deadlock in chan_sip. Lock scenario presented here: Thread 1
+ holds ast_rdlock_contexts &conlock holds handle_statechange hints
+ holds handle_statechange hint waiting for cb_extensionstate
+ Locked Here: chan_sip.c line 7428 (find_call) Thread 2 holds
+ handle_request_do &netlock holds find_call sip_pvt_ptr waiting
+ for ast_rdlock_contexts &conlock Locked Here: pbx.c line 9911
+ (ast_rdlock_contexts) Chan_sip has an established locking order
+ of locking the sip_pvt and then getting the context lock. So the
+ as stated by the summary, the operations in thread 2 have been
+ modified to no longer require the context lock. (closes issue
+ #18310) Reported by: one47 Patches: statecbs_ao2.mk2.patch
+ uploaded by one47 (license 23), modified by me Review:
+ https://reviewboard.asterisk.org/r/1072/ ........
+
+2011-01-18 18:11 +0000 [r302174] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Merged revisions 302173 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r302173 | rmudgett | 2011-01-18 12:07:15 -0600
+ (Tue, 18 Jan 2011) | 95 lines Merged revisions 302172 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011)
+ | 88 lines Issues with DTMF triggered attended transfers. Issue
+ #17999 1) A calls B. B answers. 2) B using DTMF dial *2 (code in
+ features.conf for attended transfer). 3) A hears MOH. B dial
+ number C 4) C ringing. A hears MOH. 5) B hangup. A still hears
+ MOH. C ringing. 6) A hangup. C still ringing until
+ "atxfernoanswertimeout" expires. For v1.4 C will ring forever
+ until C answers the dead line. (Issue #17096) Problem: When A and
+ B hangup, C is still ringing. Issue #18395 SIP call limit of B is
+ 1 1. A call B, B answered 2. B *2(atxfer) call C 3. B hangup, C
+ ringing 4. Timeout waiting for C to answer 5. Recall to B fails
+ because B has reached its call limit. Because B reached its call
+ limit, it cannot do anything until the transfer it started
+ completes. Issue #17273 Same scenario as issue 18395 but party B
+ is an FXS port. Party B cannot do anything until the transfer it
+ started completes. If B goes back off hook before C answers, B
+ hears ringback instead of the expected dialtone. ********** Note
+ for the issue #17273 and #18395 fix: DTMF attended transfer works
+ within the channel bridge. Unfortunately, when either party A or
+ B in the channel bridge hangs up, that channel is not completely
+ hung up until the transfer completes. This is a real problem
+ depending upon the channel technology involved. For chan_dahdi,
+ the channel is crippled until the hangup is complete. Either the
+ channel is not useable (analog) or the protocol disconnect
+ messages are held up (PRI/BRI/SS7) and the media is not released.
+ For chan_sip, a call limit of one is going to block that endpoint
+ from any further calls until the hangup is complete. For party A
+ this is a minor problem. The party A channel will only be in this
+ condition while party B is dialing and when party B and C are
+ conferring. The conversation between party B and C is expected to
+ be a short one. Party B is either asking a question of party C or
+ announcing party A. Also party A does not have much incentive to
+ hangup at this point. For party B this can be a major problem
+ during a blonde transfer. (A blonde transfer is our term for an
+ attended transfer that is converted into a blind transfer. :))
+ Party B could be the operator. When party B hangs up, he assumes
+ that he is out of the original call entirely. The party B channel
+ will be in this condition while party C is ringing, while
+ attempting to recall party B, and while waiting between call
+ attempts. WARNING: The ATXFER_NULL_TECH conditional is a hack to
+ fix the problem. It will replace the party B channel technology
+ with a NULL channel driver to complete hanging up the party B
+ channel technology. The consequences of this code is that the 'h'
+ extension will not be able to access any channel technology
+ specific information like SIP statistics for the call.
+ ATXFER_NULL_TECH is not defined by default. ********** (closes
+ issue #17999) Reported by: iskatel Tested by: rmudgett JIRA
+ SWP-2246 (closes issue #17096) Reported by: gelo Tested by:
+ rmudgett JIRA SWP-1192 (closes issue #18395) Reported by:
+ shihchuan Tested by: rmudgett (closes issue #17273) Reported by:
+ grecco Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1047/ ........
+ ................
+
+2011-02-22 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.3 Released.
+
+ * Merged changes related to AST-2011-002
+
+2011-02-16 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.3-rc3 Released.
+
+ ------------------------------------------------------------------------
+ r301790 | jpeeler | 2011-01-14 11:32:53 -0600 (Fri, 14 Jan 2011) | 42 lines
+
+ Resolve deadlock involving REFER.
+
+ (closes issue 0018403)
+ Reported by: jthurman
+ Patches:
+ 20110103-blind_deadlock.diff uploaded by jthurman (license 614)
+ issue18403.patch uploaded by jpeeler (license 325)
+ Tested by: jthurman
+
+ ------------------------------------------------------------------------
+
+ ------------------------------------------------------------------------
+ r308002 | qwell | 2011-02-15 17:32:21 -0600 (Tue, 15 Feb 2011) | 10
+ lines
+
+ Fix regression that changed behavior of queues when ringing a queue
+ member.
+
+ This reverts r298596, which was to fix a highly bizarre and contrived
+ issue with a queue member that called into his own queue being
+ transferred back into his own queue. I couldn't reproduce that issue in
+ any way. I think one of the other recent transfer fixes actually fixed
+ this.
+
+ (closes issue 0018747)
+ Reported by: vrban
+
+ ------------------------------------------------------------------------
+
+2011-01-20 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.3-rc2 Released.
+
+ ------------------------------------------------------------------------
+ r303907 | mnicholson | 2011-01-25 14:56:12 -0600 (Tue, 25 Jan 2011) | 2
+ lines
+
+ Reimplemented fax session reservation to reverse the ABI breakage
+ introduced in r297486.
+ ------------------------------------------------------------------------
+
+ ------------------------------------------------------------------------
+ r303106 | sruffell | 2011-01-20 13:56:35 -0600 (Thu, 20 Jan 2011) | 15
+ lines
+
+ main/features: Use POLLPRI when waiting for events on parked channels.
+
+ This change resolves a regression in the 1.6.2 when converting from
+ select to poll. The DAHDI timers use POLLPRI to indicate that the
+ timer
+ fired, but features was not waiting for that flag. The result was no
+ audio for MOH when a call was parked and res_timing_dahdi was in use.
+
+ This patch is slightly modified from the one on the mantis issue. It
+ does
+ not set an exception on the channel if the POLLPRI flag is set.
+
+ (closes issue 0018262)
+ Reported by: francesco_r
+ Patches:
+ patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029)
+ Tested by: francesco_r, rfrantik, one47
+ ------------------------------------------------------------------------
+
+ ------------------------------------------------------------------------
+ r302785 | russell | 2011-01-19 16:35:15 -0600 (Wed, 19 Jan 2011) | 15
+ lines
+
+ Resolve a memory leak with the manager interface is disabled.
+
+ The intent of this check as it stands in previous versions of Asterisk
+ was to
+ check if there are any active sessions. If there were no sessions,
+ then the
+ function would return immediately and not bother with queueing up the
+ manager
+ event to be processed. Since the conversion of storing sessions in an
+ astobj2
+ container, this check will always pass. I changed it to go back to
+ checking
+ what was intended.
+
+ The side effect of this was that if the AMI is disabled, the manager
+ event
+ queue is populated anyway, but the code that runs to clear out the
+ queue
+ never runs. A producer with no consumer is a bad thing.
+
+ Reported internally by kmorgan.
+
+ ------------------------------------------------------------------------
+
+ ------------------------------------------------------------------------
+ r302837 | russell | 2011-01-19 17:56:48 -0600 (Wed, 19 Jan 2011) | 2
+ lines
+
+ Only check container count if it exists.
+
+ ------------------------------------------------------------------------
+
+2011-01-17 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.3-rc1 Released.
+
+2011-01-18 18:11 +0000 [r302174] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Merged revisions 302173 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r302173 | rmudgett | 2011-01-18 12:07:15 -0600
+ (Tue, 18 Jan 2011) | 95 lines Merged revisions 302172 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011)
+ | 88 lines Issues with DTMF triggered attended transfers. Issue
+ #17999 1) A calls B. B answers. 2) B using DTMF dial *2 (code in
+ features.conf for attended transfer). 3) A hears MOH. B dial
+ number C 4) C ringing. A hears MOH. 5) B hangup. A still hears
+ MOH. C ringing. 6) A hangup. C still ringing until
+ "atxfernoanswertimeout" expires. For v1.4 C will ring forever
+ until C answers the dead line. (Issue #17096) Problem: When A and
+ B hangup, C is still ringing. Issue #18395 SIP call limit of B is
+ 1 1. A call B, B answered 2. B *2(atxfer) call C 3. B hangup, C
+ ringing 4. Timeout waiting for C to answer 5. Recall to B fails
+ because B has reached its call limit. Because B reached its call
+ limit, it cannot do anything until the transfer it started
+ completes. Issue #17273 Same scenario as issue 18395 but party B
+ is an FXS port. Party B cannot do anything until the transfer it
+ started completes. If B goes back off hook before C answers, B
+ hears ringback instead of the expected dialtone. ********** Note
+ for the issue #17273 and #18395 fix: DTMF attended transfer works
+ within the channel bridge. Unfortunately, when either party A or
+ B in the channel bridge hangs up, that channel is not completely
+ hung up until the transfer completes. This is a real problem
+ depending upon the channel technology involved. For chan_dahdi,
+ the channel is crippled until the hangup is complete. Either the
+ channel is not useable (analog) or the protocol disconnect
+ messages are held up (PRI/BRI/SS7) and the media is not released.
+ For chan_sip, a call limit of one is going to block that endpoint
+ from any further calls until the hangup is complete. For party A
+ this is a minor problem. The party A channel will only be in this
+ condition while party B is dialing and when party B and C are
+ conferring. The conversation between party B and C is expected to
+ be a short one. Party B is either asking a question of party C or
+ announcing party A. Also party A does not have much incentive to
+ hangup at this point. For party B this can be a major problem
+ during a blonde transfer. (A blonde transfer is our term for an
+ attended transfer that is converted into a blind transfer. :))
+ Party B could be the operator. When party B hangs up, he assumes
+ that he is out of the original call entirely. The party B channel
+ will be in this condition while party C is ringing, while
+ attempting to recall party B, and while waiting between call
+ attempts. WARNING: The ATXFER_NULL_TECH conditional is a hack to
+ fix the problem. It will replace the party B channel technology
+ with a NULL channel driver to complete hanging up the party B
+ channel technology. The consequences of this code is that the 'h'
+ extension will not be able to access any channel technology
+ specific information like SIP statistics for the call.
+ ATXFER_NULL_TECH is not defined by default. ********** (closes
+ issue #17999) Reported by: iskatel Tested by: rmudgett JIRA
+ SWP-2246 (closes issue #17096) Reported by: gelo Tested by:
+ rmudgett JIRA SWP-1192 (closes issue #18395) Reported by:
+ shihchuan Tested by: rmudgett (closes issue #17273) Reported by:
+ grecco Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1047/ ........
+ ................
+
+2011-01-17 15:04 +0000 [r302005] Terry Wilson <twilson@digium.com>
+
+ * configs/sip.conf.sample: Document "encryption" option in
+ sip.conf.sample
+
+2011-01-14 21:09 +0000 [r301946] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Deadlock between dahdi_request() and
+ pri_dchannel() processing an incomming call. The
+ sig_pri_new_ast_channel() is called with the channel private lock
+ held when pri_dchannel() calls it and no channel private lock
+ held when dahdi_request() calls it. The use of pri_grab() in
+ sig_pri_new_ast_channel() could leave the channel private lock
+ held when it returns if the lock was not held before calling it.
+ Make sig_pri_new_ast_channel() just lock the PRI span lock
+ instead of using pri_grab(). It is safe to do this because
+ dahdi_request() does not have the channel private lock and the
+ deadlock potential with the PRI span lock is only between
+ pri_dchannel() and other threads.
+
+2011-01-14 20:11 +0000 [r301851] Brett Bryant <bbryant@digium.com>
+
+ * channels/chan_multicast_rtp.c: Changing previous revisions
+ 301845/301847 to use ast_sockaddr_setnull() instead of setting
+ the field manually to avoid uninitialized data. Review:
+ https://reviewboard.asterisk.org/r/1076/
+
+2011-01-14 20:05 +0000 [r301849] Andrew Latham <lathama@gmail.com>
+
+ * funcs/func_base64.c, /, funcs/func_aes.c: Add relationships to
+ function documentation. Fix amatuer type mistake
+
+2011-01-14 19:35 +0000 [r301845] Brett Bryant <bbryant@digium.com>
+
+ * channels/chan_multicast_rtp.c: Fix for a consistent MulticastRTP
+ channel driver crash due to use of unitilized data. (closes issue
+ #18290) (closes issue #18602) Reported by: voipgate, wybecom
+ Review: https://reviewboard.asterisk.org/r/1076/
+
+2011-01-14 19:35 +0000 [r301844] Andrew Latham <lathama@gmail.com>
+
+ * funcs/func_base64.c, /, funcs/func_aes.c: Add relationships to
+ function documentation.
+
+2011-01-14 17:32 +0000 [r301790] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_sip.c: Resolve deadlock involving REFER. Two fixes:
+ 1) One must always have the private unlocked before calling
+ pbx_builtin_setvar_helper to not invalidate locking order since
+ it locks the channel. 2) Unlock the channel before calling
+ pbx_find_extension, which starts and stops autoservice during the
+ lookup. The problem scenario as illustrated by the reporter:
+ Thread: do_monitor ----------------------- handle_request_do
+ handle_incoming handle_request_refer ast_parking_ext_valid
+ pbx_find_extension ast_autoservice_stop while (chan_list_state ==
+ as_chan_list_state) { usleep(1000); } Thread: autoservice_run
+ ----------------------- autoservice_run chan = ast_waitfor_n
+ ast_waitfor_nandfds ast_waitfor_nandfds_classic / simple /
+ complex (depending on your system) ast_channel_lock(c[x]);
+ handle_request_do and schedule_process_request_queue locks the
+ owner if it exists. The autoservice thread is waiting for the
+ channel lock, which wasn't ever released since the do_monitor
+ thread was waiting for autoservice operations to complete. Solved
+ by unlocking the channel but keeping a reference to guarantee
+ safety. (closes issue #18403) Reported by: jthurman Patches:
+ 20110103-blind_deadlock.diff uploaded by jthurman (license 614)
+ issue18403.patch uploaded by jpeeler (license 325) Tested by:
+ jthurman
+
+2011-01-13 17:01 +0000 [r301731] Leif Madsen <lmadsen@digium.com>
+
+ * configs/phoneprov.conf.sample, /: Merged revisions 301730 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r301730 | lmadsen | 2011-01-13 11:01:11 -0600 (Thu, 13 Jan 2011)
+ | 7 lines Add static entry for split Polycom 332 firmware.
+ (closes issue #18607) Reported by: cjacobsen Patches:
+ polycom_331.diff uploaded by cjacobsen (license 1029) Tested by:
+ lathama ........
+
+2011-01-12 21:19 +0000 [r301683] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 301682 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r301682 | twilson | 2011-01-12 15:05:02 -0600 (Wed, 12 Jan 2011)
+ | 9 lines Don't reject all SUBSCRIBE auth requests When merging
+ another SUBSCRIBE fix from 1.4, some braces were put in the wrong
+ place. This patch fixes that. (closes issue #18597) Reported by:
+ thsgmbh ........
+
+2011-01-12 18:51 +0000 [r301595] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/manager.c, /: Merged revisions 301594 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r301594 | mnicholson | 2011-01-12 12:50:31 -0600
+ (Wed, 12 Jan 2011) | 15 lines Removed a usleep(1) that shouldn't
+ be necessary in session_do, and removed the ms_t member from the
+ mansession_session structure. Merged revisions 301591 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r301591 | mnicholson | 2011-01-12 12:39:03 -0600 (Wed, 12 Jan
+ 2011) | 5 lines Don't store the thread id for the manager session
+ in the structure we pass to the thread for the manager session.
+ ABE-2543 ........ ................
+
+2011-01-12 18:12 +0000 [r301504] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 301503 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r301503 | jpeeler | 2011-01-12 12:11:49 -0600
+ (Wed, 12 Jan 2011) | 19 lines Merged revisions 301502 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r301502 | jpeeler | 2011-01-12 12:10:42 -0600 (Wed, 12 Jan 2011)
+ | 12 lines Fix CPU spike when pressing DTMF after agent login.
+ The problem here is that DTMF was being continuously deferred and
+ requeued since ast_safe_sleep is called in a loop. There are
+ serveral other places in the code that sleeps and then loops in a
+ similar fashion. Because of this fact I opted to not defer DTMF
+ any more, which will not affect the original fix:
+ https://reviewboard.asterisk.org/r/674 (closes issue #18130)
+ Reported by: rgj ........ ................
+
+2011-01-12 16:05 +0000 [r301446] David Vossel <dvossel@digium.com>
+
+ * main/file.c: Removal of unused variables so Asterisk will
+ compile.
+
+2011-01-12 15:57 +0000 [r301444] Stefan Schmidt <sst@sil.at>
+
+ * Makefile: fix wrong text of rerun menuselect after user interface
+ warning the warning, if no user interface for menuselect warning
+ was found is not right. you have to rerun configure before make
+ menuselect after installing a proper user interface. (closes
+ issue #18594) Reported by: Dovid
+
+2011-01-12 00:26 +0000 [r301402] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/file.c: Call execl() directly for a better solution for
+ paths with spaces. (closes issue #18600) Reported by: ebroad
+ Patches: 20110111__issue18600__2.diff.txt uploaded by tilghman
+ (license 14)
+
+2011-01-11 19:16 +0000 [r301311] Paul Belanger <pabelanger@digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 301310 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r301310 | pabelanger | 2011-01-11 14:14:31 -0500 (Tue, 11 Jan
+ 2011) | 2 lines Fix a logic issue when passing context ARG
+ ........
+
+2011-01-11 18:51 +0000 [r301308] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/utils.c, /: Merged revisions 301307 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r301307 | mnicholson | 2011-01-11 12:42:05 -0600
+ (Tue, 11 Jan 2011) | 11 lines Merged revisions 301305 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r301305 | mnicholson | 2011-01-11 12:34:40 -0600 (Tue, 11 Jan
+ 2011) | 4 lines Prevent buffer overflows in ast_uri_encode()
+ ABE-2705 ........ ................
+
+2011-01-10 22:39 +0000 [r301263] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/strcompat.c: Little endian machines were not converted
+ properly. (closes issue #18583) Reported by: jcovert Patches:
+ 20110110__issue18583.diff.txt uploaded by tilghman (license 14)
+ Tested by: jcovert
+
+2011-01-09 21:40 +0000 [r301177-301221] Paul Belanger <pabelanger@digium.com>
+
+ * autoconf/ast_ext_lib.m4, /, configure, configure.ac: Merged
+ revisions 301220 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r301220 | pabelanger | 2011-01-09 16:38:24 -0500 (Sun, 09 Jan
+ 2011) | 14 lines SOUND_CACHE_DIR now defaults to empty Sounds
+ files included in the Asterisk tarball were being ignored and
+ re-downloaded. Users wanting to cache the files can still
+ override the setting using the --with-sounds-cache option.
+ (closes issue #18589) Reported by: pabelanger Patches:
+ issue18589.patch uploaded by pabelanger (license 224) Tested by:
+ pabelanger Review: https://reviewboard.asterisk.org/r/1074/
+ ........
+
+ * apps/app_verbose.c, /: Merged revisions 301176 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r301176 | pabelanger | 2011-01-08 16:58:24 -0500 (Sat, 08 Jan
+ 2011) | 7 lines Indicate log level argument for Log() is not
+ optional (closes issue #18586) Reported by: kshumard Patches:
+ app_verbose.c.patch uploaded by kshumard (license 92) ........
+
+2011-01-08 01:11 +0000 [r301134] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: The DTMF attended transfer feature cannot
+ callback a chan_dahdi BRI phone. The DAHDI ISDN channel name is
+ not dialable. Make a channel name like DAHDI/i3/400-12 dialable
+ when the sequence number is stripped off of the name.
+
+2011-01-07 20:53 +0000 [r301090] Jason Parker <jparker@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 301089 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r301089 | qwell | 2011-01-07 14:52:00 -0600 (Fri, 07 Jan 2011) |
+ 8 lines Initialize useropts/adminopts in case there is no column
+ in the realtime DB. (closes issue #18182) Reported by: dimas
+ Patches: v1-18182.patch uploaded by dimas (license 88) Tested by:
+ dimas ........
+
+2011-01-07 19:58 +0000 [r300955-301047] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 301046 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r301046 | jpeeler | 2011-01-07 13:57:42 -0600 (Fri, 07
+ Jan 2011) | 8 lines Fix regression causing forwarding voicemails
+ to not work with file storage. I had actually already fixed this
+ in 295200 in 1.4 and thought it wasn't missing in the other
+ branches for some reason. (closes issue #18358) Reported by:
+ cabal95 ........
+
+ * apps/app_voicemail.c, /: Merged revisions 300951 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r300951 | jpeeler | 2011-01-07 11:23:37 -0600
+ (Fri, 07 Jan 2011) | 14 lines Merged revisions 300918 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r300918 | jpeeler | 2011-01-07 11:13:21 -0600 (Fri, 07 Jan 2011)
+ | 7 lines Ensure good bye prompt in voicemail is played at the
+ correct time. Specifically in the case of timing out but not
+ leaving voicemail nothing should be heard. And when leaving
+ voicemail it should be heard. ABE-2647 ........ ................
+
+2011-01-06 06:28 +0000 [r300798] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * addons/res_config_mysql.c: Don't destroy handle not created by
+ use (because the caller will). (closes issue #18526) Reported by:
+ makoto Patches: res-config-mysql-include.patch uploaded by makoto
+ (license 38) Tested by: makoto
+
+2011-01-05 20:54 +0000 [r300714] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Merged revision 300711 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r300711 | rmudgett | 2011-01-05 13:43:55 -0600 (Wed,
+ 05 Jan 2011) | 14 lines A call retrieved from hold may wind up
+ with no audio. If the retrieved call is natively bridged then the
+ call may not have any audio path. The following warning message
+ is given: "Failed to add <dfd> to conference <chan>/<chan>:
+ Invalid argument". * Open the media on a B channel when
+ pri_fixup_principle() moves the call from a no_b_channel channel
+ to a real channel. * Added lock protection while
+ pri_fixup_principle() moves a call from one private structure to
+ another. * Made some pri_fixup_principle() messages more
+ meaningful. ..........
+
+2011-01-05 18:56 +0000 [r300623] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * res/res_odbc.c, /: Merged revisions 300622 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r300622 | tilghman | 2011-01-05 12:54:58 -0600
+ (Wed, 05 Jan 2011) | 17 lines Merged revisions 300621 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r300621 | tilghman | 2011-01-05 12:47:46 -0600 (Wed, 05 Jan 2011)
+ | 10 lines Use the sanity check in place of the
+ disconnect/connect cycle. The disconnect/connect cycle has the
+ potential to cause random crashes. (closes issue #18243) Reported
+ by: ks3 Patches: res_odbc.patch uploaded by ks3 (license 1147)
+ Tested by: ks3 ........ ................
+
+2011-01-05 16:29 +0000 [r300575] Paul Belanger <pabelanger@digium.com>
+
+ * /, cdr/cdr_sqlite.c: Merged revisions 300574 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r300574 | pabelanger | 2011-01-05 11:28:07 -0500 (Wed, 05 Jan
+ 2011) | 6 lines Change deprecated message to LOG_WARNING Also
+ removed latter part of message Discussed on #asterisk-dev
+ ........
+
+2011-01-04 21:53 +0000 [r300433-300521] Leif Madsen <lmadsen@digium.com>
+
+ * channels/chan_iax2.c, main/xmldoc.c, /, channels/chan_sip.c,
+ channels/chan_agent.c: Merged revisions 300520 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r300520 | lmadsen | 2011-01-04 15:52:41 -0600 (Tue, 04 Jan 2011)
+ | 9 lines Fix backwards and broken XML documentation. (closes
+ issue #18547) Reported by: jcovert Patches: xmldoc.c.patch
+ uploaded by jcovert (license 551) chan_iax2.c.doc.patch uploaded
+ by jcovert (license 551) chan_sip.c.patch uploaded by jcovert
+ (license 551) chan_agent.c.patch uploaded by jcovert (license
+ 551) ........
+
+ * configs/users.conf.sample, /: Merged revisions 300431 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r300431 | lmadsen | 2011-01-04 15:00:29 -0600 (Tue, 04 Jan 2011)
+ | 7 lines Add some documentation to users.conf.sample. (closes
+ issue #18531) Reported by: lathama Patches:
+ users.conf.sample2.diff uploaded by lathama (license 1028) Tested
+ by: lathama ........
+
+2011-01-04 21:00 +0000 [r300430] Russell Bryant <russell@digium.com>
+
+ * contrib/scripts/autosupport, /, contrib/scripts/autosupport.8:
+ Merged revisions 300429 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r300429 | russell | 2011-01-04 14:59:56 -0600
+ (Tue, 04 Jan 2011) | 11 lines Merged revisions 300428 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r300428 | russell | 2011-01-04 14:56:04 -0600 (Tue, 04 Jan 2011)
+ | 4 lines Update the autosupport script from Digium support.
+ (closes AST-395) ........ ................
+
+2011-01-04 19:45 +0000 [r300384] Leif Madsen <lmadsen@digium.com>
+
+ * phoneprov/000000000000.cfg: Update STAT() to use the comma
+ instead of the pipe. (closes issue #18503) Reported by: cjacobsen
+ Patches: old_separator.diff uploaded by cjacobsen (license 1029)
+ Tested by: lathama
+
+2011-01-04 17:54 +0000 [r300301] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 300298 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r300298 | twilson | 2011-01-04 11:37:26 -0600
+ (Tue, 04 Jan 2011) | 22 lines Merged revisions 300216 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r300216 | twilson | 2011-01-04 11:11:48 -0600 (Tue, 04 Jan 2011)
+ | 15 lines Don't authenticate SUBSCRIBE re-transmissions This
+ only skips authentication on retransmissions that are already
+ authenticated. A similar method is already used for INVITES. This
+ is the kind of thing we end up having to do when we don't have a
+ transaction layer... (closes issue #18075) Reported by: mdu113
+ Patches: diff.txt uploaded by twilson (license 396) Tested by:
+ twilson, mdu113 Review: https://reviewboard.asterisk.org/r/1005/
+ ........ ................
+
+2011-01-04 17:01 +0000 [r300214] Jan Kalab <pitlicek@gmail.com>
+
+ * res/res_calendar_exchange.c, res/res_calendar_icalendar.c: Memory
+ leaking in calendars ne_request_destroy() was missing in
+ icalendar and exchange calendar modules, causing memory leak.
+ (closes issue #18521) Review:
+ https://reviewboard.asterisk.org/r/1068/
+
+2011-01-03 23:14 +0000 [r300166] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Merged revisions 300165 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r300165 | rmudgett | 2011-01-03 17:02:13 -0600 (Mon, 03 Jan 2011)
+ | 4 lines Use correct variable for atxfercallbackretries config
+ option. * Misc formatting changes. ........
+
+2011-01-03 13:14 +0000 [r300082] Leif Madsen <lmadsen@digium.com>
+
+ * pbx/pbx_dundi.c: Increase side of mapping response field. I've
+ increased the size of the response field in a DUNDi mapping
+ because of some documentation I'm writing. Previously it was set
+ to AST_MAX_EXTENSION which is only 80 characters, which is far
+ too small when you're using some dialplan functions to craft a
+ response. The example I'm using is: extensions =>
+ RegisteredDevices,0,SIP,dundi:very_awesome_password/${IF($[${DB_EXISTS(phones/${NUMBER}/device)}]?${DB(phones/${NUMBER}/device)}:None)},nopartial
+
+2010-12-29 22:02 +0000 [r299989] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * apps/app_voicemail.c, main/file.c: Quote arguments, just in case
+ there's a space in a pathname. (Diagnosed by pabelanger on
+ #asterisk-dev, fixed by me.)
+
+2010-12-29 19:28 +0000 [r299865-299948] Paul Belanger <pabelanger@digium.com>
+
+ * sounds/Makefile: Only remove /tmp/astdatadir, not
+ /var/lib/asterisk
+
+ * build_tools/make_sample_voicemail, sounds/Makefile, Makefile:
+ Properly quote varibles for MAC OS X
+
+ * apps/app_chanspy.c, /: Merged revisions 299864 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r299864 | pabelanger | 2010-12-28 13:51:13 -0500 (Tue, 28 Dec
+ 2010) | 2 lines Documentation typo ........
+
+2010-12-27 21:23 +0000 [r299752-299820] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * sounds/Makefile: More space-in-pathname issues.
+
+ * sounds/Makefile, Makefile, Makefile.moddir_rules: Mac OS X
+ spaces-in-pathnames fix.
+
+ * configure: Regen configure
+
+ * configure.ac: Properly quote path on Darwin.
+
+2010-12-25 16:12 +0000 [r299711] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooq931.c,
+ addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooh245.c: Change
+ order of sending TCS and MSD packets Change order of sending
+ Terminal Capability Set and MasterSlave Determination packets,
+ MSD send when TCS exchange procedure is done (we send tcs ack to
+ remote and we have remote tcs ack already or we receive tcs ack
+ from remote and we have send our tcs ack to remote already). Some
+ endpoints can work in this sequence only, i suggest they can't
+ work with both (tcs and msd) exchange procedures simultaneously.
+ Also changed StartH245 facility message sending. It send on
+ incoming calls only due to some endpoints can't proccess properly
+ this facility messages on their incoming calls. (issue #18433)
+ Reported by: MrHanMan Patches: tcs-msd-h245-3.patch uploaded by
+ may213 (license 454) Tested by: MrHanMan, may213
+
+2010-12-25 10:07 +0000 [r299583-299626] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * channels/chan_local.c, /: Merged revisions 299625 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r299625 | tilghman | 2010-12-25 04:05:00 -0600
+ (Sat, 25 Dec 2010) | 12 lines Merged revisions 299624 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r299624 | tilghman | 2010-12-25 04:04:06 -0600 (Sat, 25 Dec 2010)
+ | 5 lines Move check for extension existence below variable
+ inheritance, due to the possible use of an eswitch. (closes issue
+ #16228) Reported by: jlaguilar ........ ................
+
+ * addons/res_config_mysql.c: Reset 'first' variable after usage.
+ (closes issue #18525) Reported by: makoto Patches:
+ res-config-mysql-update2.patch uploaded by makoto (license 38)
+
+2010-12-23 02:53 +0000 [r299531] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 299530 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r299530 | moy | 2010-12-22 21:28:37 -0500 (Wed, 22 Dec
+ 2010) | 7 lines Enqueue AST_CONTROL_PROGRESS after
+ AST_CONTROL_RINGING when MFC-R2 calls are accepted (closes issue
+ #18438) Reported by: mariner7 Tested by: moy ........
+
+2010-12-22 20:05 +0000 [r299449] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * pbx/ael/ael-test/ref.ael-test19,
+ pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c,
+ pbx/ael/ael-test/ref.ael-vtest25,
+ pbx/ael/ael-test/ref.ael-vtest17, /,
+ pbx/ael/ael-test/ref.ael-test3: Merged revisions 299448 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r299448 | tilghman | 2010-12-22 14:03:30 -0600 (Wed, 22 Dec 2010)
+ | 8 lines Resolve warnings by disambiguating the "s" extension as
+ used by chan_dahdi from the "s" extension as used by the AEL
+ macros. (closes issue #18480) Reported by: nivek Patches:
+ 20101215__issue18480__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: nivek ........
+
+2010-12-22 02:10 +0000 [r299405] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Chan_dahdi sends an empty COLP on the bridged
+ channel. Chan_dahdi always inserts a connected party IE when you
+ call from one dahdi channel to another dahdi channel, even if no
+ such information was received on the 2nd channel. This clears the
+ display of many phones. * Removed leftover artifact from before
+ the valid flag was added. * Updated all of the channel's caller
+ id information with the new connected line information instead of
+ just the string parts. (closes issue #18508) Reported by: wimpy
+ Patches: issue18508_trunk.patch uploaded by rmudgett (license
+ 664) Tested by: wimpy, rmudgett
+
+2010-12-21 15:25 +0000 [r299353] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 299242 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r299242 | mnicholson | 2010-12-20 15:25:35 -0600
+ (Mon, 20 Dec 2010) | 23 lines Merged revisions
+ 299194,299198,299220 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r299194 | mnicholson | 2010-12-20 14:45:38 -0600 (Mon, 20 Dec
+ 2010) | 6 lines Respond as soon as possible with a 202 Accepted
+ to refer requests. This change also plugs a few memory leaks that
+ can occur when parking sip calls. ABE-2656 ........ r299198 |
+ mnicholson | 2010-12-20 15:00:44 -0600 (Mon, 20 Dec 2010) | 2
+ lines Remove changes to via processing that were not supposed to
+ go into the last commit. ........ r299220 | mnicholson |
+ 2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) | 4 lines Use
+ ast_free() instead of free() ABE-2656 ........ ................
+
+2010-12-21 00:44 +0000 [r299312] Paul Belanger <pabelanger@digium.com>
+
+ * configs/cel.conf.sample: Correct typo with USER_DEFINED event.
+ (closes issue #18461) Reported by: joscas Patches:
+ cel.conf.sample.diff uploaded by lathama (license 1028) Tested
+ by: lathama, joscas
+
+2010-12-20 21:38 +0000 [r299248] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix a couple of CCSS issues. * Make sure to
+ allocate a cc_params structure when creating autopeers. * Use
+ sip_uri_cmp when retrieving SIP CC agents and monitors in case
+ parameters appear in the URI. (closes issue #18504) Reported by:
+ kkm (closes issue #18338) Reported by: GeorgeKonopacki Patches:
+ 18338.diff uploaded by mmichelson (license 60) Tested by:
+ GeorgeKonopacki
+
+2010-12-20 18:17 +0000 [r299131-299138] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * sample.call, /: Merged revisions 299136 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r299136 | tilghman | 2010-12-20 12:16:37 -0600 (Mon, 20 Dec 2010)
+ | 2 lines Documentation fix ........
+
+ * cdr/cdr_pgsql.c, /: Merged revisions 299130 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r299130 | tilghman | 2010-12-20 11:41:24 -0600 (Mon, 20 Dec 2010)
+ | 11 lines If a call was not answered, then the billsec was
+ calculated unusually large. Also, due to a copy and paste error,
+ a request for the answer field would have given the start value,
+ instead. (closes issue #18460) Reported by: joscas Patches:
+ 20101215__issue18460.diff.txt uploaded by tilghman (license 14)
+ Tested by: joscas ........
+
+2010-12-20 16:18 +0000 [r299088] Leif Madsen <lmadsen@digium.com>
+
+ * /, main/features.c: Merged revisions 299087 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r299087 | lmadsen | 2010-12-20 10:18:03 -0600 (Mon, 20 Dec 2010)
+ | 5 lines Note that Park() timeout is milliseconds. (closes issue
+ #15758) Reported by: mmurdock Tested by: mmurdock, seanbright
+ ........
+
+2010-12-20 09:14 +0000 [r299004] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * main/aoc.c, channels/sig_pri.h, channels/chan_sip.c: Typos:
+ recieved => received
+
+2010-12-18 00:09 +0000 [r298818-298963] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, main/say.c: Merged revisions 298962 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r298962 | tilghman | 2010-12-17 18:08:57 -0600 (Fri, 17 Dec 2010)
+ | 2 lines Remove backtrace used for testing merge process
+ ........
+
+ * main/utils.c, main/astobj2.c, utils/conf2ael.c,
+ include/asterisk/logger.h, configure,
+ build_tools/menuselect-deps.in, main/logger.c, utils/ael_main.c,
+ utils/hashtest2.c, makeopts.in, utils/check_expr.c,
+ utils/refcounter.c, include/asterisk/utils.h,
+ build_tools/cflags-devmode.xml, /,
+ include/asterisk/autoconfig.h.in, main/Makefile, main/say.c,
+ configure.ac, utils/hashtest.c: Merged revisions 298957 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r298957 | tilghman | 2010-12-17 17:30:55 -0600
+ (Fri, 17 Dec 2010) | 13 lines Merged revisions 298905 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r298905 | tilghman | 2010-12-17 15:40:56 -0600 (Fri, 17 Dec 2010)
+ | 6 lines Let Asterisk find better backtrace information with
+ libbfd. The menuselect option BETTER_BACKTRACES, if enabled, will
+ use libbfd to search for better symbol information within both
+ the Asterisk binary, as well as loaded modules, to assist when
+ using inline backtraces to track down problems. ........
+ ................
+
+ * contrib/init.d/rc.debian.asterisk: -v implies -f, so override
+ with -F. (closes issue #18446) Reported by: lathama Patches:
+ rc.debian.asterisk.diff uploaded by lathama (license 1028) Tested
+ by: lathama
+
+ * /, configure, configure.ac: Merged revisions 298817 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r298817 | tilghman | 2010-12-17 15:03:06 -0600 (Fri, 17
+ Dec 2010) | 8 lines Also include PTHREAD_LIBS and PTHREAD_CFLAGS
+ for SQLite 3, as it's needed on some platforms. (closes issue
+ #18493) Reported by: pprindeville Patches:
+ asterisk-1.8-sqlite3.patch uploaded by pprindeville (license 347)
+ Tested by: pprindeville ........
+
+2010-12-17 17:26 +0000 [r298773] Brad Watkins <Marquis42@gmail.com>
+
+ * configs/sip.conf.sample, channels/chan_sip.c: Fix parsing of mwi
+ => lines in sip.conf Reworking parsing of mwi => lines to resolve
+ a segfault. Also add a set of unit tests for the function that
+ does the parsing. (closes issue #18350) Reported by: gbour Tested
+ by: Marquis, gbour Review:
+ https://reviewboard.asterisk.org/r/1053/
+
+2010-12-16 23:31 +0000 [r298598-298685] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 298684 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r298684 | jpeeler | 2010-12-16 17:30:59 -0600
+ (Thu, 16 Dec 2010) | 9 lines Merged revisions 298683 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r298683 | jpeeler | 2010-12-16 17:29:30 -0600 (Thu, 16
+ Dec 2010) | 2 lines After recording only silence for a voicemail
+ prepending, restore backup files. ........ ................
+
+ * apps/app_queue.c, /: Merged revisions 298597 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r298597 | jpeeler | 2010-12-16 14:49:33 -0600
+ (Thu, 16 Dec 2010) | 14 lines Merged revisions 298596 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r298596 | jpeeler | 2010-12-16 14:46:52 -0600 (Thu, 16 Dec 2010)
+ | 7 lines Fix improper hangup when doing an attended transfer to
+ queue. Had to indicate ringing in wait_for_answer so the attended
+ transfer code would not try and hang up the local channel it
+ created, which would kill the call. ABE-2624 ........
+ ................
+
+2010-12-16 09:28 +0000 [r298394-298539] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * channels/chan_sip.c: Ensure the ipaddr field in realtime is large
+ enough to handle IPv6 addresses. (closes issue #18464) Reported
+ by: IgorG Patches: realtime_ipv6store.diff uploaded by IgorG
+ (license 20) (plus a few additional lines by tilghman)
+
+ * res/res_config_odbc.c, /: Merged revisions 298481 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r298481 | tilghman | 2010-12-16 03:04:38 -0600
+ (Thu, 16 Dec 2010) | 21 lines Merged revisions 298480 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r298480 | tilghman | 2010-12-16 03:03:40 -0600 (Thu, 16 Dec 2010)
+ | 14 lines Only increment the pointer once per loop, otherwise we
+ corrupt the value. (closes issue #18251) Reported by: bcnit
+ Patches: 20101110__issue18251.diff.txt uploaded by tilghman
+ (license 14) Tested by: trev, jthurman, elguero (closes issue
+ #18279) Reported by: zerohalo Patches:
+ 20101109__issue18279.diff.txt uploaded by tilghman (license 14)
+ Tested by: zerohalo ........ ................
+
+ * /, funcs/func_dialgroup.c: Merged revisions 298477 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r298477 | tilghman | 2010-12-16 02:54:23 -0600 (Thu, 16
+ Dec 2010) | 8 lines Eliminate duplicates from container. (closes
+ issue #18091) Reported by: bunny Patches:
+ 20101006__issue18091.diff.txt uploaded by tilghman (license 14)
+ Tested by: bunny ........
+
+ * /, cdr/cdr_sqlite.c: Merged revisions 298393 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r298393 | tilghman | 2010-12-15 18:29:10 -0600
+ (Wed, 15 Dec 2010) | 15 lines Merged revisions 298392 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r298392 | tilghman | 2010-12-15 18:28:04 -0600 (Wed, 15 Dec 2010)
+ | 8 lines Unregister before shutting down the connection, to
+ avoid a race. (closes issue #18481) Reported by: pabelanger
+ Patches: 20101215__issue18481.diff.txt uploaded by tilghman
+ (license 14) Tested by: pabelanger ........ ................
+
+2010-12-13 17:11 +0000 [r298195] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c, channels/chan_dahdi.c, /: Merged revisions
+ 298194 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r298194 | rmudgett | 2010-12-13 11:04:41 -0600
+ (Mon, 13 Dec 2010) | 26 lines Merged revisions 298193 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r298193 | rmudgett | 2010-12-13 10:56:07 -0600 (Mon, 13 Dec 2010)
+ | 19 lines Outgoing PRI/BRI calls cannot do DTMF triggered
+ transfers. Outgoing PRI/BRI calls cannot do DTMF triggered
+ transfers if a PROCEEDING message is not received. The debug
+ output shows that the DTMF begin event is seen, but the DTMF end
+ event is missing. When the DTMF begin happens, the call is muted
+ so we now have one way audio (until a DTMF end event is somehow
+ seen). * Made set the proceeding flag when the PRI_EVENT_ANSWER
+ event is received. * Made absorb the DTMF begin and DTMF end
+ events if we are overlap dialing and have not seen a PROCEEDING
+ message. * Added a debug message when absorbing a DTMF event.
+ JIRA SWP-2690 JIRA ABE-2697 ........ ................
+
+2011-01-12 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.2 Released.
+
+ * Merge in a change in the configure script to fix an issue for
+ Debian packagers.
+
+ ------------------------------------------------------------------------
+ r301221 | pabelanger | 2011-01-09 15:40:35 -0600 (Sun, 09 Jan 2011)
+ | 21 lines
+
+ Merged revisions 301220 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 [^]
+
+ ........
+ r301220 | pabelanger | 2011-01-09 16:38:24 -0500 (Sun, 09 Jan
+ 2011) | 14 lines
+ SOUND_CACHE_DIR now defaults to empty
+
+ Sounds files included in the Asterisk tarball were being
+ ignored and
+ re-downloaded. Users wanting to cache the files can
+ still override the setting
+ using the --with-sounds-cache option.
+
+ (closes issue 0018589)
+ Reported by: pabelanger
+ Patches:
+ issue18589.patch uploaded by
+ pabelanger (license 224)
+ Tested by: pabelanger
+
+ Review:
+ https://reviewboard.asterisk.org/r/1074/
+
+ ------------------------------------------------------------------------
+
+2010-12-13 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.2-rc1 Released.
+
+2010-12-11 21:45 +0000 [r298099] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooGkClient.c: Correction to work with
+ gatekeeper which don't send GK ID Don't use GK ID if it's not
+ presented in GK replies Extract GK ID not only in GK confirm but
+ in GK register confirm also (issue #18401) Reported by: MrHanMan
+ Patches: no-gkid-2.patch uploaded by may213 (license 454) Tested
+ by: may213, MrHanMan
+
+2010-12-10 16:52 +0000 [r298054] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: Prevent a memcpy overlap in
+ GENERIC_FAX_EXEC_SET_VARS
+
+2010-12-10 16:26 +0000 [r298051] Tilghman Lesher <tlesher@digium.com>
+
+ * main/netsock.c, /, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Merged revisions 298050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r298050 | tilghman | 2010-12-10 10:24:13 -0600 (Fri, 10 Dec 2010)
+ | 11 lines Portability issue on OpenSolaris. Also detect the
+ required structure element, because OpenSolaris defines
+ SIOCGIFHWADDR, but without support for IP sockets. (closes issue
+ #18442) Reported by: ranjtech Patches:
+ 20101209__issue18442.diff.txt uploaded by tilghman (license 14)
+ Tested by: ranjtech ........
+
+2010-12-09 22:18 +0000 [r297965] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 297960 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297960 | twilson | 2010-12-09 16:10:31 -0600
+ (Thu, 09 Dec 2010) | 21 lines Merged revisions 297959 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297959 | twilson | 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010)
+ | 14 lines Ignore spurious REGISTER requests If a REGISTER
+ request with a Call-ID matching an existing transaction is
+ received it was possible that the REGISTER request would
+ overwrite the initreq of the private structure. This info is used
+ to generate messages for other responses in the transaction. This
+ patch ignores REGISTER requests that match non-REGISTER
+ transactions. (closes issue #18051) Reported by: eeman Tested by:
+ twilson Review: https://reviewboard.asterisk.org/r/1050/ ........
+ ................
+
+2010-12-09 21:32 +0000 [r297957] David Vossel <dvossel@digium.com>
+
+ * channels/chan_gtalk.c: Fixes issue with outbound google voice
+ calls not working. Thanks to az1234 and nevermind_quack for their
+ input in helping debug the issue. (closes issue #18412) Reported
+ by: nevermind_quack Patches: fix uploaded by dvossel (license
+ 671)
+
+2010-12-09 20:48 +0000 [r297952] Terry Wilson <twilson@digium.com>
+
+ * main/features.c: Don't crash after Set(CDR(userfield)=...) in
+ ast_bridge_call Instead of setting peer->cdr = NULL, set it to
+ not post. (closes issue #18415) Reported by: macbrody Patches:
+ patch-18415 uploaded by jsolares (license 1167) Tested by:
+ jsolares, twilson
+
+2010-12-08 18:06 +0000 [r297909] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 297908 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r297908 | tilghman | 2010-12-08 12:04:38 -0600 (Wed, 08 Dec 2010)
+ | 4 lines Use inheritance to get correct results for
+ SIPFROMDOMAIN. (from an internal Digium discussion) ........
+
+2010-12-08 16:12 +0000 [r297905] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: Display the capabilities requested when requesting
+ a fax session fails instead of displaying a hex value. Tweak the
+ way fax stats are calculated so that all fax attempts and
+ faliures are logged. Also make ensure faxes are either counted as
+ completed or falied and never both. FAX-210
+
+2010-12-07 22:59 +0000 [r297825] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 297824 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297824 | jpeeler | 2010-12-07 16:58:54 -0600
+ (Tue, 07 Dec 2010) | 19 lines Merged revisions 297823 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297823 | jpeeler | 2010-12-07 16:57:48 -0600 (Tue, 07 Dec 2010)
+ | 12 lines Revert code that changed SSRC for DTMF. Some previous
+ behavior was attempted to be restored, but mistakingly I did not
+ realize that the previous behavior was incorrect. This fixes DTMF
+ not being detected since DTMF shouldn't cause the SSRC to change.
+ (related to issue #17404) (closes issue #18189) (closes issue
+ #18352) Reported by: marcbou Tested by: cmbaker82 ........
+ ................
+
+2010-12-07 22:51 +0000 [r297733-297821] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/org.asterisk.muted.plist (added), Makefile,
+ contrib/init.d/org.asterisk.asterisk.plist, utils/muted.c, /:
+ Merged revisions 297819 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297819 | tilghman | 2010-12-07 16:40:45 -0600
+ (Tue, 07 Dec 2010) | 11 lines Merged revisions 297818 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297818 | tilghman | 2010-12-07 16:35:50 -0600 (Tue, 07 Dec 2010)
+ | 4 lines Use non-deprecated APIs for CoreAudio Review:
+ https://reviewboard.asterisk.org/r/1040/ ........
+ ................
+
+ * apps/app_followme.c, /: Merged revisions 297713 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297713 | tilghman | 2010-12-06 18:21:50 -0600
+ (Mon, 06 Dec 2010) | 15 lines Merged revisions 297689 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010)
+ | 8 lines Don't create a Local channel if the target extension
+ does not exist. (closes issue #18126) Reported by: junky Patches:
+ followme.diff uploaded by junky (license 177) (partially
+ restructured by me to avoid a possible memory leak) ........
+ ................
+
+2010-12-06 22:06 +0000 [r297607] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 297605 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297605 | jpeeler | 2010-12-06 16:03:04 -0600
+ (Mon, 06 Dec 2010) | 18 lines Merged revisions 297603 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297603 | jpeeler | 2010-12-06 15:57:15 -0600 (Mon, 06 Dec 2010)
+ | 12 lines Improve handling of REGISTER requests with multiple
+ contact headers. The changes here attempt to more strictly follow
+ RFC 3261 section 10.3. Basically the following will now cause a
+ 400 Bad Response to be returned, if: - multiple Contact headers
+ are present with one set to expire all bindings ("*") - wildcard
+ parameter is specified for Contact without Expires header or
+ Expires header is not set to zero. ABE-2442 ABE-2443 ........
+ ................
+
+2010-12-03 17:41 +0000 [r297535] Sean Bright <sean@malleable.com>
+
+ * channels/chan_console.c, /: Merged revisions 297534 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r297534 | seanbright | 2010-12-03 12:40:52 -0500 (Fri,
+ 03 Dec 2010) | 3 lines The CLI command should not contain
+ <placeholder>s, these are for descriptions. ........
+
+2010-12-03 15:21 +0000 [r297486-297495] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: Print a DEBUG message instead of a WARNING message
+ when the selected fax tech does not support reserving sessions.
+ Answer the channel before quering it for t.38 support. This is
+ necessary for the query to work properly over local channels.
+
+ * include/asterisk/res_fax.h, res/res_fax.c: Add support for
+ reserving a fax session before answering the channel. Note: this
+ change breaks ABI compatibility. FAX-217
+
+2010-12-02 20:09 +0000 [r297406] Paul Belanger <pabelanger@digium.com>
+
+ * Makefile, /: Merged revisions 297405 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297405 | pabelanger | 2010-12-02 15:06:43 -0500
+ (Thu, 02 Dec 2010) | 14 lines Merged revisions 297404 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297404 | pabelanger | 2010-12-02 15:01:08 -0500 (Thu, 02 Dec
+ 2010) | 7 lines Resolve compile error under FreeBSD We now set
+ _ASTCFLAGS+=-march=i686 for i386 processors, still allowing
+ ASTCFLAGS to override the setting. Review:
+ https://reviewboard.asterisk.org/r/1043/ ........
+ ................
+
+2010-12-02 18:13 +0000 [r297312] Terry Wilson <twilson@digium.com>
+
+ * /, main/abstract_jb.c: Merged revisions 297311 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297311 | twilson | 2010-12-02 12:07:39 -0600
+ (Thu, 02 Dec 2010) | 21 lines Merged revisions 297310 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297310 | twilson | 2010-12-02 12:00:27 -0600 (Thu, 02 Dec 2010)
+ | 12 lines Initialize offset for adaptive jitter buffer When the
+ adaptive jitter buffer is enabled in sip.conf, the first frame
+ placed in the jitter buffer fails with something like:
+ jb_warning_output: Resyncing the jb. last_delay 0, this delay
+ -215886466, threshold 1000, new offset 215886466 This happens
+ because the offset is not initialized before calling jb_put().
+ This patch modifies jb_put_first_adaptive() to set the offset to
+ the frame's timestamp. Review:
+ https://reviewboard.asterisk.org/r/1041/ ........
+ ................
+
+2010-12-02 13:20 +0000 [r297245] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 297229 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297229 | russell | 2010-12-02 07:16:47 -0600
+ (Thu, 02 Dec 2010) | 13 lines Merged revisions 297228 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010)
+ | 6 lines Add "DAHDI" to a couple of app_meetme error messages.
+ This is in response to some questions on IRC. To the user, there
+ was nothing that made it obvious that this error had anything to
+ do with DAHDI not being loaded. ........ ................
+
+2010-12-01 19:47 +0000 [r297157] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: Changed some NOTICE and WARNING messages to DEBUG
+ messages.
+
+2010-12-01 17:53 +0000 [r297075] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 297073 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297073 | jpeeler | 2010-12-01 11:52:46 -0600
+ (Wed, 01 Dec 2010) | 30 lines Merged revisions 297072 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010)
+ | 23 lines Fix not stopping MOH when transfered local channel
+ queue member is answered. The problem here is only present when
+ local channels are used with the MOH passthru option as well as
+ no optimization (/nm). I will describe the slightly bizarre
+ scenario that was used to test, where phones B and C are queue
+ members: Phone A dials into a queue with two members using local
+ channels and the above options. Phone B answers. Phone A blind
+ transfers phone B into the same queue. Phone A hangs up. Phone C
+ answers, but phone B didn't stop playing MOH. In this scenario,
+ the unhold frame that should have gotten to phone B never arrived
+ due to the masquerade from the blind transfer. This is usually
+ fine since app_queue manages the starting and stopping of MOH.
+ However, with the passthrough option enabled when app_queue
+ attempts to stop MOH it tries to do so on the local channel
+ rather than the real channel. The easiest solution was to just
+ make sure to send an unhold frame during the transfer since it
+ wouldn't make sense to have MOH playing after a transfer anyway.
+ This only modifies SIP transfers, but the other transfers did not
+ seem to be a problem. If DTMF based transfers were a problem it
+ might be okay to add ast_moh_stop to finishup, but I didn't want
+ to have to add that unless required. ABE-2624 ........
+ ................
+
+2010-12-01 17:01 +0000 [r296951-296992] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/frame.h, /: Merged revisions 296991 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296991 | tilghman | 2010-12-01 11:01:00 -0600
+ (Wed, 01 Dec 2010) | 12 lines Merged revisions 296990 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296990 | tilghman | 2010-12-01 10:59:26 -0600 (Wed, 01 Dec 2010)
+ | 5 lines Clarify documentation on how we store codec preference
+ lists. (closes issue #18397) Reported by: birgita ........
+ ................
+
+ * channels/chan_iax2.c, /: Merged revisions 296950 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r296950 | tilghman | 2010-11-30 19:38:19 -0600 (Tue, 30
+ Nov 2010) | 2 lines Missed initializations caused startup errors
+ on Mac OS X (and possibly others, too). ........
+
+2010-12-01 00:28 +0000 [r296870] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 296869 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296869 | jpeeler | 2010-11-30 18:24:58 -0600
+ (Tue, 30 Nov 2010) | 11 lines Merged revisions 296868 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296868 | jpeeler | 2010-11-30 18:23:19 -0600 (Tue, 30 Nov 2010)
+ | 4 lines Properly restore backup information file when hanging
+ up during message prepending. ABE-2654 ........ ................
+
+2010-11-30 19:12 +0000 [r296787] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c: DOC: Conference number can be omitted; if
+ omitted, all users in a meetme are listed.
+
+2010-11-29 23:05 +0000 [r296673] Paul Belanger <pabelanger@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 296671 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296671 | pabelanger | 2010-11-29 17:54:14 -0500
+ (Mon, 29 Nov 2010) | 12 lines Merged revisions 296670 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296670 | pabelanger | 2010-11-29 17:49:39 -0500 (Mon, 29 Nov
+ 2010) | 5 lines Make sure nothing else is needed before
+ destroying the scheduler. (closes issue #18398) Reported by:
+ pabelanger ........ ................
+
+2010-11-29 21:26 +0000 [r296628] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Complete some error handling in
+ transmit_publish() in chan_sip.c. This error handling block
+ caught my eye. It was missing a couple of things, but it should
+ be safe now. Thanks to mmichelson for the quick peer review on
+ IRC.
+
+2010-11-29 20:46 +0000 [r296582] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_msg_parser.c, channels/chan_misdn.c: Merged
+ revision 296575 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r296575 | rmudgett | 2010-11-29 14:27:37 -0600 (Mon,
+ 29 Nov 2010) | 13 lines Invalid mISDN PTMP redirecting signaling
+ as TE towards NT. The mISDN PTMP redirection signaling (NOTIFY
+ redirecting number and notification code, SETUP redirecting
+ number) is also sent in PTMP/TE mode. It should only apply in
+ PTMP/NT mode. The call setup proceeds but the network (Deutsche
+ Telekom) reacts with ugly ISDN STATUS messages. Also don't send
+ the redirecting number ie when PTP is also sending the
+ DivertingLegInformation2 facility. The redirecting number ie is
+ redundant and the network (Deutsche Telekom) complains about it.
+ Patches: abe_2651_v4.patch uploaded by rmudgett (license 664)
+ JIRA ABE-2651 JIRA SWP-2537 ..........
+
+2010-11-29 07:28 +0000 [r296534] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Merged revisions 296533 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r296533 | tilghman | 2010-11-29 01:27:09 -0600 (Mon, 29 Nov 2010)
+ | 13 lines I love standards. There are so many to choose from.
+ Except when there isn't one. Linux and *BSD disagree on the
+ elements within the ucred structure. Detect which one is in use
+ on the system. (closes issue #18384) Reported by: bjm Patches:
+ cred-diffs uploaded by bjm (license 473)
+ 20101127__issue18384__1.6.2.diff.txt uploaded by tilghman
+ (license 14) 20101127__issue18384__1.8.diff.txt uploaded by
+ tilghman (license 14) Tested by: tilghman, bjm ........
+
+2010-11-27 10:40 +0000 [r296429-296467] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 296466 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r296466 | tilghman | 2010-11-27 04:39:01 -0600 (Sat, 27 Nov 2010)
+ | 5 lines 18 characters is too short for most date/times (20 is
+ the usual, but we add more in case of greater precision). (closes
+ issue #18369) Reported by: tnakonz ........
+
+ * include/asterisk.h: Also don't build DEBUG_FD_LEAKS when
+ STANDALONE2 is defined. (closes issue #18385) Reported by: cmaj
+
+2010-11-26 21:37 +0000 [r296391] Olle Johansson <oej@edvina.net>
+
+ * main/say.c: Merged revisions 296351 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296351 | oej | 2010-11-26 13:23:03 +0100 (Fre,
+ 26 Nov 2010) | 17 lines Merged revisions 296309 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296309 | oej | 2010-11-26 10:53:31 +0100 (Fre, 26 Nov 2010) | 11
+ lines Fix bugs in saying numbers using the Swedish language
+ syntax (closes issue #18355) Reported by: oej Patch by: oej Much
+ help from Peter Lindahl. Testing by the ClearIT team during a
+ coffee break. Review: https://reviewboard.asterisk.org/r/1033/
+ ........ ................
+
+2010-11-26 18:31 +0000 [r296352-296354] Brad Watkins <Marquis42@gmail.com>
+
+ * res/res_jabber.c: Fix XMPP PubSub-based distributed device state.
+ Initialize pubsubflags to 0 so res_jabber doesn't think there is
+ already an XMPP connection sending device state. Also clean up
+ CLI commands a bit. (closes issue #18272) Reported by: klaus3000
+ Patches: res_jabber_fix_pubsubflags_and_CLI-patch.txt uploaded by
+ klaus3000 (license 65) Tested by: klaus3000, Marquis Review:
+ https://reviewboard.asterisk.org/r/1030/
+
+ * channels/chan_sip.c: Fix reloading of peer when a user is
+ requested. Prevent peer reloading from causing multiple MWI
+ subscriptions to be created when using realtime. This had the
+ effect of sending one NOTIFY for every time a sip peer made a
+ call, in one case eventually overwhelming the phone and causing
+ it to reboot. (closes issue #18342) Reported by: nivek Patches:
+ issue0018342p1.patch uploaded by nivek (license 636) Tested by:
+ nivek Review: https://reviewboard.asterisk.org/r/1029/
+
+2010-11-24 23:29 +0000 [r296230] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 296221 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296221 | russell | 2010-11-24 17:28:19 -0600
+ (Wed, 24 Nov 2010) | 13 lines Merged revisions 296213 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010)
+ | 6 lines Make Asterisk less crashy. Since we might not put a new
+ translation path on the channel, go ahead and set it to NULL
+ right after destroying the old one to ensure we don't try to free
+ an invalid translation path later on. ........ ................
+
+2010-11-24 22:49 +0000 [r296167] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
+ /, channels/sig_analog.h: Merged revisions 296166 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296166 | rmudgett | 2010-11-24 16:42:45 -0600
+ (Wed, 24 Nov 2010) | 50 lines Merged revisions 296165 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24 Nov 2010)
+ | 43 lines Oneway audio to SIP phone from FXS port after FXS port
+ gets a CallWaiting pip. The FXS connected phone has to have
+ CW/CID support to fail, as it will send back a DTMF 'A' or 'D'
+ when it's ready to receive CallerID. A normal phone with no CID
+ never fails. Also the SIP phone does not hear MOH when the CW
+ call is answered. The DTMF end frame is suppressed when the phone
+ acknowledges the CW signal for CID. The problem is the DTMF begin
+ frame needs to be suppressed as well. The DTMF begin frame is
+ causing SIP to start sending the DTMF RTP frames. Since the DTMF
+ end frame is suppressed, SIP will not stop sending those DTMF RTP
+ packets. * Suppress the DTMF begin and end frames when the
+ channel driver is looking for DTMF digits. * Fixed a couple
+ issues caused by not cleaning up the CID spill if you answer the
+ CW call while it is sending the CID spill. * Fixed not sending
+ CW/CID spill to the phone when the call is natively bridged.
+ (Fixed by not using native bridge if CW/CID is possible.) *
+ Suppress received audio when sending CW/CID spills. The other
+ parties involved do not need to hear the CW/CID spills and may be
+ confused if the CW call is for them. (closes issue #18129)
+ Reported by: alecdavis Patches: issue_18129_v1.8_v3.patch
+ uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett
+ NOTE: * v1.4 does not have the main problem fixed by suppressing
+ the DTMF start frames. The other three items fixed are relevant.
+ * If you really must restore native bridging between analog
+ ports, you need to disable CW/CID either by configuring
+ chan_dahdi.conf callwaitingcallerid=no or dialing *70 before
+ dialing the number to temporarily disable CW. ........
+ ................
+
+2010-11-24 20:23 +0000 [r296002-296084] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 296083 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296083 | russell | 2010-11-24 14:23:11 -0600
+ (Wed, 24 Nov 2010) | 19 lines Merged revisions 296082 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010)
+ | 12 lines Fix false reporting of an error by set_format(). In
+ the case that the native format was able to be changed to match
+ the new requested format, the code proceeded to attempt to build
+ a translation path, anyway. The result would be NULL, since no
+ translation path is necessary and resulted in this function
+ thinking an error has occurred. This case is now specifically
+ caught and no attempt to build a translation path is attempted.
+ Thanks to our automated tests and bamboo.asterisk.org for
+ catching this problem and making a whole lot of noise when things
+ started failing. :-) ........ ................
+
+ * apps/app_dial.c, main/channel.c, /: Merged revisions 296001 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296001 | russell | 2010-11-24 11:03:16 -0600
+ (Wed, 24 Nov 2010) | 45 lines Merged revisions 296000 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010)
+ | 38 lines Handle failures building translation paths more
+ effectively. The problem scenario occurred on a heavily loaded
+ system that was using the codec_dahdi module and exceeded the
+ hardware transcoding capacity. The failure mode at that point was
+ not good. The report came in to us as an Asterisk lock-up. The
+ "core show locks" shows a ton of threads locked up (but no
+ obvious deadlock). Upon deeper investigation, when the system is
+ in this state, the CPU was maxed out. The CPU was being consumed
+ by the Asterisk logger spewing messages on every audio frame for
+ calls set up after transcoder capacity was reached. The purpose
+ of this patch is to make Asterisk handle failures to create a
+ translation path in a more graceful manner. If we can't
+ translate, then the call just needs to be dropped, as it's not
+ going to work. These are the changes: 1) In set_format() of
+ channel.c (which is called by set_read_format() and
+ set_write_format()), it was ignoring if
+ ast_translator_build_path() failed and returned NULL. It now pays
+ attention to that case and returns a result reflecting failure.
+ With this change in place, the bridging code will immediately
+ detect a failure and end the bridge instead of proceeding to try
+ to bridge frames that can't be translated and making channel
+ drivers freak out by sending them frames in a format they weren't
+ expecting. 2) In ast_indicate_data() of channel.c, failure of
+ ast_playtones_start() was ignored. It is now reflected in the
+ return value of the function. This didn't turn out to have any
+ affect on the bug, but seemed like a good change to leave in. 3)
+ In app_dial(), when only sending a call to a single endpoint, it
+ will attempt to do some bridging of its own of early audio. It
+ uses make_compatible() when it's going to do this. However, it
+ ignored failure from make compatible. So, even with the fix from
+ #1, if there was early audio going through app_dial, there would
+ still be a period of invalid frames passing through. After
+ detecting failure here, Dial() exits. ABE-2658 ........
+ ................
+
+2010-11-23 10:30 +0000 [r295949] Olle Johansson <oej@edvina.net>
+
+ * /, main/say.c: Merged revisions 295907 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r295907 | oej | 2010-11-23 10:36:38 +0100 (Tis,
+ 23 Nov 2010) | 14 lines Merged revisions 295906 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r295906 | oej | 2010-11-23 10:28:14 +0100 (Tis, 23 Nov 2010) | 8
+ lines Fix support of saynumber(1,n) in the Swedish language
+ (closes issue #18353) Reported by: oej Review:
+ https://reviewboard.asterisk.org/r/1031/ ........
+ ................
+
+2010-11-22 20:03 +0000 [r295869] Sean Bright <sean@malleable.com>
+
+ * configs/chan_dahdi.conf.sample, /: Merged revisions 295868 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r295868 | seanbright | 2010-11-22 15:02:37 -0500 (Mon, 22 Nov
+ 2010) | 2 lines Change some documentation to suggest
+ dahdi_monitor instead of ztmonitor. ........
+
+2010-11-22 19:36 +0000 [r295866] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_macro.c, include/asterisk/channel.h,
+ include/asterisk/frame.h, main/channel.c, main/pbx.c, /: Merged
+ revisions 295843 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r295843 | rmudgett | 2010-11-22 13:28:23 -0600
+ (Mon, 22 Nov 2010) | 53 lines Merged revisions 295790 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010)
+ | 46 lines The channel redirect function (CLI or AMI) hangs up
+ the call instead of redirecting the call. To recreate the
+ problem: 1) Party A calls Party B 2) Invoke CLI "channel
+ redirect" command to redirect channel call leg associated with A.
+ 3) All associated channels are hung up. Note that if the CLI
+ command were done on the channel call leg associated with B it
+ works. This regression was a result of the fix for issue #16946
+ (https://reviewboard.asterisk.org/r/740/). The regression affects
+ all features that use an async goto to execute the dialplan
+ because of an external event: Channel redirect, AMI redirect, SIP
+ REFER, and FAX detection. The struct ast_channel._softhangup code
+ is a mess. The variable is used for several purposes that do not
+ necessarily result in the call being hung up. I have added
+ doxygen comments to describe how the various _softhangup bits are
+ used. I have corrected all the places where the variable was
+ tested in a non-bit oriented manner. The primary fix is the new
+ AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so
+ the soft hangup requests that do not normally result in a hangup
+ do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171)
+ Reported by: SantaFox (closes issue #18185) Reported by:
+ kwemheuer (closes issue #18211) Reported by: zahir_koradia
+ (closes issue #18230) Reported by: vmarrone (closes issue #18299)
+ Reported by: mbrevda (closes issue #18322) Reported by: nerbos
+ Review: https://reviewboard.asterisk.org/r/1013/ ........
+ ................
+
+2010-11-20 03:11 +0000 [r295747] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h: One way audio before answering call
+ waiting call on analog port. * Analog call waiting Caller ID
+ spills could get stuck resulting in one way audio until the
+ waiting call is answered. This only happens on the second (and
+ later) call waiting call if the active call is not the first
+ call. * The CLI/AMI "dahdi show channel" command could report the
+ wrong channel information. Must keep the struct analog_pvt.owner
+ and struct dahdi_pvt.owner pointer in sync.
+
+2010-11-20 00:50 +0000 [r295711] Russell Bryant <russell@digium.com>
+
+ * main/event.c, include/asterisk/event.h, /: Merged revisions
+ 295710 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r295710 | russell | 2010-11-19 18:45:51 -0600 (Fri, 19 Nov 2010)
+ | 29 lines Fix cache of device state changes for multiple
+ servers. This patch addresses a regression where device states
+ across multiple servers were not being processing completely
+ correctly. The code works to determine the overall state by
+ looking at the last known state of a device on each server.
+ However, there was a regression due to some invasive rewrites of
+ how the cache works that led to the cache only storing the last
+ device state change for a device, regardless of which server it
+ was on. The code is set up to cache device state change events by
+ ensuring that each event in the cache has a unique device name +
+ entity ID (server ID). The code that was responsible for
+ comparing raw information elements (which EID is) always returned
+ a match due to a memcmp() with a length of 0. There isn't much
+ code to fix the actual bug. This patch also introduces a new CLI
+ command that was very useful for debugging this problem. The
+ command allows you to dump the contents of the event cache.
+ (closes issue #18284) Reported by: klaus3000 Patches:
+ issue18284.rev1.txt uploaded by russell (license 2) Tested by:
+ russell, klaus3000 (closes issue #18280) Reported by: klaus3000
+ Review: https://reviewboard.asterisk.org/r/1012/ ........
+
+2010-11-19 22:06 +0000 [r295673] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 295672 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r295672 | twilson | 2010-11-19 13:55:48 -0800
+ (Fri, 19 Nov 2010) | 15 lines Merged revisions 295628 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r295628 | twilson | 2010-11-19 12:53:36 -0800 (Fri, 19 Nov 2010)
+ | 8 lines Discard responses with more than one Via This is not a
+ perfect solution as headers that are joined via commas are not
+ detected. This is a parsing issue that to fix "correctly" would
+ necessitate a new SIP parser. Review:
+ https://reviewboard.asterisk.org/r/1019/ ........
+ ................
+
+2010-11-19 21:40 +0000 [r295670] Brett Bryant <bbryant@digium.com>
+
+ * apps/app_queue.c: Patch for deadlock from ordering issue between
+ channel/queue locks in app_queue (set_queue_variables). (closes
+ issue #18031) Reported by: rain Review:
+ https://reviewboard.asterisk.org/r/1018/
+
+2010-11-19 16:47 +0000 [r295516] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h: Bring sig_analog extraction more into
+ alignment with orig-trunk/v1.6.2 chan_dahdi. * Restore SMDI
+ support. * Fixed initial value of struct analog_pvt.use_callerid.
+ It may get forced on depending upon other config options. * Call
+ analog_dnd() instead of manual inlined code. * Removed unused
+ struct analog_pvt.usedistinctiveringdetection. * Removed the
+ struct analog_pvt.unknown_alarm flag. It was really the struct
+ analog_pvt.inalarm flag. * Use ast_debug() instead of
+ ast_log(LOG_DEBUG). * Rename several function's index variable to
+ idx. * Some formatting tweaks.
+
+2010-11-18 20:30 +0000 [r295477] Leif Madsen <lmadsen@digium.com>
+
+ * configs/sip_notify.conf.sample: 'sip notify clear-mwi' needs
+ terminating CRLF. (closes issue #18275) Reported by: klaus3000
+ Patches: fix_body_CRLF_patch.txt uploaded by klaus3000 (license
+ 65)
+
+2010-11-18 18:02 +0000 [r295361-295441] Paul Belanger <pabelanger@digium.com>
+
+ * res/res_jabber.c, /, include/asterisk/jabber.h: Merged revisions
+ 295440 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r295440 | pabelanger | 2010-11-18 12:51:34 -0500 (Thu, 18 Nov
+ 2010) | 4 lines Fix compiler warnings when using openssl-dev
+ 1.0.0+ Review: https://reviewboard.asterisk.org/r/1016/ ........
+
+ * contrib/scripts/install_prereq: Add RedHat specific dependencies
+
+ * configs/res_curl.conf.sample: Uncomment settings under [global],
+ to surpress warning when loading Asterisk.
+
+2010-11-16 23:02 +0000 [r295282] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /: Merged revisions 295281 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r295281 | rmudgett | 2010-11-16 16:57:07 -0600
+ (Tue, 16 Nov 2010) | 9 lines Merged revisions 295280 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r295280 | rmudgett | 2010-11-16 16:52:06 -0600 (Tue, 16
+ Nov 2010) | 1 line Dead code elimination in
+ channel.c:ast_channel_bridge() variable who. ........
+ ................
+
+2010-11-16 22:41 +0000 [r295164-295278] Russell Bryant <russell@digium.com>
+
+ * build_tools/prep_tarball: Check for pdftotext and give a useful
+ error if not found.
+
+ * build_tools/prep_tarball: Remove intentional typo I had added
+ when testing the check. oops.
+
+ * build_tools/prep_tarball: Check for wikiexport.py in PATH and
+ give a useful error message if not found.
+
+2010-12-02 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.1 Released.
+
+2010-11-16 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.1-rc1 Released.
+
+2010-11-15 18:30 +0000 [r294989-295078] Tilghman Lesher <tlesher@digium.com>
+
+ * tests/test_expr.c (added), /: Merged revisions 295062 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r295062 | tilghman | 2010-11-15 12:24:02 -0600
+ (Mon, 15 Nov 2010) | 9 lines Merged revisions 295026 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r295026 | tilghman | 2010-11-15 11:58:37 -0600 (Mon, 15
+ Nov 2010) | 2 lines Create test verifying results of expression
+ parser ........ ................
+
+ * funcs/func_curl.c, /: Merged revisions 294988 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r294988 | tilghman | 2010-11-15 01:42:39 -0600 (Mon, 15 Nov 2010)
+ | 8 lines It is possible to crash Asterisk by feeding the curl
+ engine invalid data. (closes issue #18161) Reported by: wdoekes
+ Patches: 20101029__issue18161.diff.txt uploaded by tilghman
+ (license 14) Tested by: tilghman ........
+
+2010-11-12 21:14 +0000 [r294905-294911] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 294910 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r294910 | jpeeler | 2010-11-12 15:14:23 -0600 (Fri, 12
+ Nov 2010) | 4 lines Return correct error code if lock path fails.
+ The recent changes to open_mailbox actually caused it to be
+ fixed, but let's be consistent. Reported by alecdavis in
+ asterisk-dev. ........
+
+ * apps/app_voicemail.c, /: Merged revisions 294904 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r294904 | jpeeler | 2010-11-12 14:51:15 -0600
+ (Fri, 12 Nov 2010) | 23 lines Merged revisions 294903 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12 Nov 2010)
+ | 16 lines Fix regression causing abort in voicemail after
+ opening a mailbox with no mesgs. In order to be more safe, some
+ error handling code was changed to respect more error conditions
+ including the potential memory allocation failure for deleted and
+ heard message tracking introduced in 293004. However,
+ last_message_index returns -1 for zero messages (perhaps as
+ expected) and was triggering the stricter error checking. Because
+ last_message_index is only called directly in one place, just
+ return 0 from open_mailbox (for file based storage) when no
+ messages are detected unless a real error has occurred. (closes
+ issue #18240) Reported by: leobrown Patches:
+ bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585)
+ Tested by: pabelanger ........ ................
+
+2010-11-12 02:45 +0000 [r294823] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c, channels/sig_pri.h, /: Merged revisions
+ 294822 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r294822 | rmudgett | 2010-11-11 20:44:12 -0600
+ (Thu, 11 Nov 2010) | 18 lines Merged revisions 294821 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11 Nov 2010)
+ | 11 lines Asterisk is getting a "No D-channels available!"
+ warning message every 4 seconds. Asterisk is just whining too
+ much with this message: "No D-channels available! Using Primary
+ channel XXX as D-channel anyway!". Filtered the message so it
+ only comes out once if there is no D channel available without an
+ intervening D channel available period. (closes issue #17270)
+ Reported by: jmls ........ ................
+
+2010-11-11 22:17 +0000 [r294740-294745] Russell Bryant <russell@digium.com>
+
+ * doc/CCSS_architecture.pdf (removed): Remove CCSS architecture
+ PDF. It has been moved to:
+ https://wiki.asterisk.org/wiki/display/AST/CCSS+Architecture
+
+ * doc/digium-mib.txt (removed), doc/followme.txt (removed),
+ doc/building_queues.txt (removed), doc/timing.txt (removed),
+ doc/advice_of_charge.txt (removed), doc/unistim.txt (removed),
+ doc/video_console.txt (removed), doc/macroexclusive.txt
+ (removed), doc/google-soc2009-ideas.txt (removed), doc/README.txt
+ (added), doc/callfiles.txt (removed), doc/externalivr.txt
+ (removed), doc/codec-64bit.txt (removed),
+ build_tools/prep_tarball, doc/video.txt (removed), doc/jingle.txt
+ (removed), doc/modules.txt (removed), doc/manager_1_1.txt
+ (removed), doc/PEERING (removed), doc/snmp.txt (removed),
+ doc/siptls.txt (removed), doc/HOWTO_collect_debug_information.txt
+ (removed), doc/ldap.txt (removed), doc/sip-retransmit.txt
+ (removed), doc/distributed_devstate.txt (removed),
+ doc/voicemail_odbc_postgresql.txt (removed), doc/tex (removed),
+ doc/queue.txt (removed), doc/jabber.txt (removed),
+ doc/chan_sip-perf-testing.txt (removed), Makefile,
+ doc/asterisk-mib.txt (removed), doc/database_transactions.txt
+ (removed), doc/smdi.txt (removed), doc/janitor-projects.txt
+ (removed), doc/hoard.txt (removed), doc/res_config_sqlite.txt
+ (removed), doc/osp.txt (removed), doc/speechrec.txt (removed),
+ doc/sms.txt (removed), doc/distributed_devstate-XMPP.txt
+ (removed), doc/valgrind.txt (removed), doc/realtimetext.txt
+ (removed), doc/cli.txt (removed), doc/rtp-packetization.txt
+ (removed), doc/datastores.txt (removed), doc/CODING-GUIDELINES
+ (removed), doc/ss7.txt (removed), doc/backtrace.txt (removed),
+ doc/India-CID.txt (removed): Remove most of the contents of the
+ doc dir in favor of the wiki content. This merge does the
+ following things: * Removes most of the contents from the doc/
+ directory in favor of the wiki - http://wiki.asterisk.org/ *
+ Updates the build_tools/prep_tarball script to know how to export
+ the contents of the wiki in both PDF and plain text formats so
+ that the documentation is still included in Asterisk release
+ tarballs.
+
+2010-11-11 21:58 +0000 [r294640-294734] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 294733 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r294733 | jpeeler | 2010-11-11 15:57:22 -0600
+ (Thu, 11 Nov 2010) | 25 lines Merged revisions 294688 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010)
+ | 18 lines Fix problem with qualify option packets for realtime
+ peers never stopping. The option packets not only never stopped,
+ but if a realtime peer was not in the peer list multiple options
+ dialogs could accumulate over time. This scenario has the
+ potential to progress to the point of saturating a link just from
+ options packets. The fix was to ensure that the poke scheduler
+ checks to see if a peer is in the peer list before continuing to
+ poke. The reason a peer must be in the peer list to be able to
+ properly manage an options dialog is because otherwise the call
+ pointer is lost when the peer is regenerated from the database,
+ which is how existing qualify dialogs are detected. (closes issue
+ #16382) (closes issue #17779) Reported by: lftsy Patches:
+ bug16382-3.patch uploaded by jpeeler (license 325) Tested by:
+ zerohalo ........ ................
+
+ * main/asterisk.c, include/asterisk.h, main/pbx.c, /: Merged
+ revisions 294639 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r294639 | jpeeler | 2010-11-11 13:31:00 -0600
+ (Thu, 11 Nov 2010) | 53 lines Merged revisions 294384 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r294384 | jpeeler | 2010-11-09 11:37:59 -0600 (Tue, 09 Nov 2010)
+ | 47 lines Fix a deadlock in device state change processing.
+ Copied from some notes from the original author (Russell):
+ Deadlock scenario: Thread 1: device state change thread Holds -
+ rdlock on contexts Holds - hints lock Waiting on channels
+ container lock Thread 2: SIP monitor thread Holds the "iflock"
+ Holds a sip_pvt lock Holds channel container lock Waiting for a
+ channel lock Thread 3: A channel thread (chan_local in this case)
+ Holds 2 channel locks acquired within app_dial Holds a 3rd
+ channel lock it got inside of chan_local Holds a local_pvt lock
+ Waiting on a rdlock of the contexts lock A bunch of other threads
+ waiting on a wrlock of the contexts lock To address this
+ deadlock, some locking order rules must be put in place and
+ enforced. Existing relevant rules: 1) channel lock before a pvt
+ lock 2) contexts lock before hints lock 3) channels container
+ before a channel What's missing is some enforcement of the order
+ when you involve more than any two. To fix this problem, I put in
+ some code that ensures that (at least in the code paths involved
+ in this bug) the locks in (3) come before the locks in (2). To
+ change the operation of thread 1 to comply, I converted the
+ storage of hints to an astobj2 container. This allows processing
+ of hints without holding the hints container lock. So, in the
+ code path that led to thread 1's state, it no longer holds either
+ the contexts or hints lock while it attempts to lock the channels
+ container. (closes issue #18165) Reported by: antonio ABE-2583
+ ........ ................
+
+2010-11-10 23:26 +0000 [r294569-294605] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_spool.c: Fixing the Mac OS X build (bamboo warning)
+
+ * pbx/pbx_spool.c: Properly queue files with inotify(7). (closes
+ issue #18089) Reported by: abelbeck Patches:
+ 20101021__issue18089.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman
+
+2010-11-10 14:14 +0000 [r294501-294535] Russell Bryant <russell@digium.com>
+
+ * UPGRADE.txt, res/ais/clm.c, res/ais/evt.c: Tweak a couple of CLI
+ commands back to their original form. The "module" in this case
+ is two parts, so there are two words before the verb of the CLI
+ command.
+
+ * main/devicestate.c, /: Merged revisions 294500 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r294500 | russell | 2010-11-10 06:41:41 -0600 (Wed, 10 Nov 2010)
+ | 7 lines Improve a debug message to be more readable and
+ consistent. (closes issue #18282) Reported by: klaus3000 Patches:
+ ast_devstate2str-patch.txt uploaded by klaus3000 (license 65)
+ ........
+
+2010-11-09 22:46 +0000 [r294466] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Allow ast_do_masquerade() failure to be reported
+ again.
+
+2010-11-09 20:33 +0000 [r294430] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 294429 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r294429 | tilghman | 2010-11-09 14:27:23 -0600 (Tue, 09 Nov 2010)
+ | 8 lines Detect GMime properly on systems where gmime flags and
+ libs are configured with pkg-config. (closes issue #16155)
+ Reported by: jcollie Patches: 20100917__issue16155.diff.txt
+ uploaded by tilghman (license 14) Tested by: tilghman ........
+
+2010-11-09 16:55 +0000 [r294349] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/channel.h, channels/sig_pri.c, main/channel.c,
+ channels/chan_misdn.c, channels/sig_analog.c: Analog lines do not
+ transfer CONNECTED LINE or execute the interception macros. Add
+ connected line update for sig_analog transfers and simplify the
+ corresponding sig_pri and chan_misdn transfer code. Note that if
+ you create a three-way call in sig_analog before transferring the
+ call, the distinction of the caller/callee interception macros
+ make little sense. The interception macro writer needs to be
+ prepared for either caller/callee macro to be executed. The
+ current implementation swaps which caller/callee interception
+ macro is executed after a three-way call is created. Review:
+ https://reviewboard.asterisk.org/r/996/ JIRA ABE-2589 JIRA
+ SWP-2372
+
+2010-11-08 22:32 +0000 [r294278-294313] Jeff Peeler <jpeeler@digium.com>
+
+ * /, res/res_timing_timerfd.c: Merged revisions 294312 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r294312 | jpeeler | 2010-11-08 16:30:49 -0600 (Mon, 08
+ Nov 2010) | 1 line add missing unlock not present in 294277
+ ........
+
+ * include/asterisk/timing.h, main/timing.c, main/channel.c, /,
+ res/res_timing_timerfd.c: Merged revisions 294277 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08
+ Nov 2010) | 16 lines Fix playback failure when using IAX with the
+ timerfd module. To fix this issue the alert pipe will now be used
+ when the timerfd module is in use. There appeared to be a race
+ that was not solved by adding locking in the timerfd module, but
+ needed to be there anyway. The race was between the timer being
+ put in non-continuous mode in ast_read on the channel thread and
+ the IAX frame scheduler queuing a frame which would enable
+ continuous mode before the non-continuous mode event was read.
+ This race for now is simply avoided. (closes issue #18110)
+ Reported by: tpanton Tested by: tpanton I put tested by tpanton
+ because it was tested on his hardware. Thanks for the remote
+ access to debug this issue! ........
+
+2010-11-08 20:56 +0000 [r294243] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 294242 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r294242 | mnicholson | 2010-11-08 14:50:21 -0600 (Mon, 08 Nov
+ 2010) | 8 lines Go off hold when we get an empty reinvite telling
+ us to. (closes issue 0014448) Reported by: frawd (closes issue
+ #17878) Reported by: frawd ........
+
+2010-11-08 19:56 +0000 [r294207] Terry Wilson <twilson@digium.com>
+
+ * configs/calendar.conf.sample, res/res_calendar.c: Set a default
+ waittime, and make sure to convert it to milliseconds
+
+2010-11-08 17:16 +0000 [r294125] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: valgrind reported references to freed
+ memory during a mISDN hangup collision. Bad things have been
+ happening in chan_misdn because the chan_misdn channel private
+ struct chan_list is not protected from reentrancy. Hangup
+ collisions have be causing read and write accesses to freed
+ memory. Converted chan_misdn struct chan_list to an ao2 object
+ for its reference counting feature. ********** Removed an
+ impediment to converting chan_list to an ao2 object. The use of
+ the other_ch member in chan_list is shaky at best. It is set if
+ the incoming and outgoing call legs are mISDN. The use of the
+ other_ch member goes against the Asterisk architecture and can
+ even cause problems. 1) It is used to disable echo cancellation.
+ This could be bad if the call is forked and the winning call leg
+ is not mISDN or the winning call leg is not the last mISDN
+ channel called by the fork. The other_ch would become a dangling
+ pointer. 2) It is used when the far end is alerting to hear the
+ far end's inband audio instead of Asterisk's generated ringback
+ tone. This is bad if the call is forked. You would only hear the
+ last forked mISDN channel and it may not be ringing yet. The
+ other_ch would become a dangling pointer if the call is later
+ transferred. ********** JIRA SWP-2423 JIRA ABE-2614
+
+2010-11-05 22:03 +0000 [r294084] Brett Bryant <bbryant@digium.com>
+
+ * channels/chan_sip.c: Fixed deadlock avoidance issues while
+ locking channel when adding the Max-Forwards header to a request.
+ (closes issue #17949) (closes issue #18200) Reported by: bwg
+ Review: https://reviewboard.asterisk.org/r/997/
+
+2010-11-05 16:05 +0000 [r294047-294049] Terry Wilson <twilson@digium.com>
+
+ * contrib/scripts/ast_tls_cert: Corret spelling and example
+
+ * contrib/scripts/ast_tls_cert: Tell people to use the correct
+ common name for clients as well
+
+2010-11-05 00:07 +0000 [r293970] Shaun Ruffell <sruffell@digium.com>
+
+ * codecs/codec_dahdi.c, /: Merged revisions 293969 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293969 | sruffell | 2010-11-04 19:06:02 -0500
+ (Thu, 04 Nov 2010) | 25 lines Merged revisions 293968 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293968 | sruffell | 2010-11-04 19:02:53 -0500 (Thu, 04 Nov 2010)
+ | 17 lines codecs/codec_dahdi: Prevent "choppy" audio when
+ receiving unexpected frame sizes. dahdi-linux 2.4.0 (specifically
+ commit 9034) added the capability for the wctc4xxp to return more
+ than a single packet of data in response to a read. However, when
+ decoding packets, codec_dahdi was still assuming that the default
+ number of samples was in each read. In other words, each packet
+ your provider sent you, regardless of size, would result in 20 ms
+ of decoded data (30 ms if decoding G723). If your provider was
+ sending 60 ms packets then codec_dahdi would end up stripping 40
+ ms of data from each transcoded frame resulting in "choppy"
+ audio. This would only affect systems where G729 packets are
+ arriving in sizes greater than 20ms or G723 packets arriving in
+ sizes greater than 30ms. DAHDI-744. ........ ................
+
+2010-11-04 21:39 +0000 [r293924] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Fixes ringback tone on sip semi-attended
+ transfer. ABE-2168
+
+2010-11-04 13:27 +0000 [r293887] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * channels/chan_sip.c: Do not output port in IPaddress for AMI
+ sippeers. (closes issue #18248) Reported by: orn Patches:
+ ami_sippeers.patch uploaded by pabelanger (license 224) Tested
+ by: orn
+
+2010-11-03 18:35 +0000 [r293807] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+ 293806 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293806 | rmudgett | 2010-11-03 13:31:57 -0500
+ (Wed, 03 Nov 2010) | 27 lines Merged revisions 293805 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03 Nov 2010)
+ | 20 lines Party A in an analog 3-way call would continue to hear
+ ringback after party C answers. All parties are analog FXS ports.
+ 1) A calls B. 2) A flash hooks to call C. 3) A flash hooks to
+ bring C into 3-way call before C answers. (A and B hear ringback)
+ 4) C answers 5) A continues to hear ringback during the 3-way
+ call. (All parties can hear each other.) * Fixed use of wrong
+ variable in dahdi_bridge() that stopped ringback on the wrong
+ subchannel. * Made several debug messages have more information.
+ A similar issue happens if B and C are SIP channels. B continues
+ to hear ringback. For some reason this only affects v1.8 and
+ trunk. * Don't start ringback on the real and 3-way subchannels
+ when creating the 3-way conference. Removing this code is benign
+ on v1.6.2 and earlier. ........ ................
+
+2010-11-03 18:05 +0000 [r293803] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/rtp_engine.h, main/rtp_engine.c,
+ channels/chan_sip.c: Avoid valgrind warnings for
+ ast_rtp_instance_get_xxx_address The documentation for
+ ast_rtp_instance_get_(local/remote)_address stated that they
+ returned 0 for success and -1 on failure. Instead, they returned
+ 0 if the address structure passed in was already equivalent to
+ the address instance local/remote address or 1 otherwise. 90% of
+ the calls to these functions completely ignored the return
+ address and passed in an uninitialized struct, which would make
+ valgrind complain even though the operation was technically safe.
+ This patch fixes the documentation and converts the
+ get_xxx_address functions to void since all they really do is
+ copy the address and cannot fail. Additionally two new functions
+ (ast_rtp_instance_get_and_cmp_(local/remote)_address) are created
+ for the 3 times where the return value was actually checked. The
+ get_and_cmp_local_address function is currently unused, but
+ exists for the sake of symmetry. The only functional change as a
+ result of this change is that we will not do an
+ ast_sockaddr_cmp() on (mostly uninitialized) addresses before
+ doing the ast_sockaddr_copy() in the get_*_address functions. So,
+ even though it is an API change, it shouldn't have a noticeable
+ change in behavior. Review:
+ https://reviewboard.asterisk.org/r/995/
+
+2010-11-02 23:09 +0000 [r293724] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 293723 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293723 | jpeeler | 2010-11-02 18:07:13 -0500
+ (Tue, 02 Nov 2010) | 15 lines Merged revisions 293722 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010)
+ | 8 lines Add enabled/disabled information for rtautoclear sip
+ show settings output. When setting to zero/"no", the numeric
+ default was shown making it not obvious the disabled setting was
+ respected. (closes issue #18123) Reported by: zerohalo ........
+ ................
+
+2010-11-02 21:29 +0000 [r293648] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+ 293647 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293647 | rmudgett | 2010-11-02 16:26:30 -0500
+ (Tue, 02 Nov 2010) | 13 lines Merged revisions 293639 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293639 | rmudgett | 2010-11-02 16:24:13 -0500 (Tue, 02 Nov 2010)
+ | 6 lines Make warning message have more useful information in
+ it. Change "Unable to get index, and nullok is not asserted" to
+ "Unable to get index for '<channel-name>' on channel <number>
+ (<function>(), line <number>)". ........ ................
+
+2010-11-02 20:45 +0000 [r293611] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * main/manager.c: If manager and tls are disabled, do not display
+ TCP/TLS Bindaddress.
+
+2010-11-01 17:29 +0000 [r293530] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h: Analog 3-way call would not connect all
+ parties if one was using sig_pri. Also the "dahdi show channel"
+ would not show the correct 3-way call status. * Synchronized the
+ inthreeway flag between chan_dahdi and sig_analog. * Fixed a
+ my_set_linear_mode() sign error and made take an analog sub
+ channel enum.
+
+2010-11-01 16:09 +0000 [r293496] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * channels/chan_iax2.c: Use ast_sockaddr_from_sin function not
+ memcpy This resolves some IAX2 registration issue report on the
+ asterisk-users mailing list. (closes issue #18202) Reported by:
+ pabelanger Patches: update_registry.patch.v2 uploaded by
+ pabelanger (license 224) Tested by: pabelanger, Nic Colledge
+ (mailing list) Review: https://reviewboard.asterisk.org/r/993
+
+2010-11-01 14:58 +0000 [r293493] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Only offer codecs both sides support for
+ directmedia When using directmedia, Asterisk needs to limit the
+ codecs offered to just the ones that both sides recognize,
+ otherwise they may end up sending audio that the other side
+ doesn't understand. (closes issue #17403) Reported by: one47
+ Patches: sip_codecs_simplified4 uploaded by one47 (license 23)
+ Tested by: one47, falves11 Review:
+ https://reviewboard.asterisk.org/r/967/
+
+2010-10-30 01:53 +0000 [r293341-293418] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+ 293417 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293417 | rmudgett | 2010-10-29 20:49:15 -0500
+ (Fri, 29 Oct 2010) | 9 lines Merged revisions 293416 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293416 | rmudgett | 2010-10-29 20:45:49 -0500 (Fri, 29
+ Oct 2010) | 1 line Remove some more code that serves no purpose.
+ ........ ................
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+ 293340 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293340 | rmudgett | 2010-10-29 19:40:10 -0500
+ (Fri, 29 Oct 2010) | 9 lines Merged revisions 293339 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293339 | rmudgett | 2010-10-29 19:34:12 -0500 (Fri, 29
+ Oct 2010) | 1 line Remove some code that serves no purpose.
+ ........ ................
+
+2010-10-29 21:48 +0000 [r293305] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_sip.c: Modify sip_setoption to not complain about
+ unknown options. This now behaves just like the other setoption
+ callbacks. For the curious the offending option for the reporter
+ was AST_OPTION_CHANNEL_WRITE which was getting passed due to a
+ fix for chan_local in 286189. (closes issue #17985) Reported by:
+ globalnetinc
+
+2010-10-28 20:00 +0000 [r293197] Tilghman Lesher <tlesher@digium.com>
+
+ * res/ael/ael.tab.h, main/ast_expr2.c, /, main/ast_expr2.h,
+ res/ael/ael.tab.c, main/ast_expr2.y, res/ael/ael_lex.c: Merged
+ revisions 293195-293196 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293195 | tilghman | 2010-10-28 14:52:52 -0500
+ (Thu, 28 Oct 2010) | 12 lines Merged revisions 293194 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010)
+ | 5 lines "!00" evaluated as false, which is incorrect. Fixing.
+ Reported (though the reporter did not understand he was reporting
+ a bug) on the asterisk-users list:
+ http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
+ ........ ................ r293196 | tilghman | 2010-10-28
+ 14:54:34 -0500 (Thu, 28 Oct 2010) | 12 lines Merged revisions
+ 293194 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010)
+ | 5 lines "!00" evaluated as false, which is incorrect. Fixing.
+ Reported (though the reporter did not understand he was reporting
+ a bug) on the asterisk-users list:
+ http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
+ ........ ................
+
+2010-10-28 16:11 +0000 [r293159] Jeff Peeler <jpeeler@digium.com>
+
+ * /, funcs/func_strings.c: Merged revisions 293158 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r293158 | jpeeler | 2010-10-28 11:09:40 -0500 (Thu, 28
+ Oct 2010) | 11 lines Fix infinite loop in FILTER(). Specifically
+ when you're using characters above \x7f or invalid character
+ escapes (e.g. \xgg). (closes issue #18060) Reported by: wdoekes
+ Patches: issue18060_func_strings_filter_infinite_loop.patch
+ uploaded by wdoekes (license 717) Tested by: wdoekes ........
+
+2010-10-26 18:49 +0000 [r293119] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 293118 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r293118 | jpeeler | 2010-10-26 13:33:24 -0500
+ (Tue, 26 Oct 2010) | 36 lines Merged revisions 293004 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25 Oct 2010)
+ | 29 lines Fix inprocess_container in voicemail to correctly
+ restrict max messages. The comparison function logic was off, so
+ the number of sessions for a given mailbox were not being
+ incremented properly. This problem caused the maximum number of
+ messages per folder to not be respected when simultaneously
+ leaving multiple voicemails just below the threshold. These
+ problems should be fixed by the above, but just in case: Fixed
+ resequence_mailbox to rely on the actual number of detected
+ number of files in a directory rather than just assuming only 10
+ messages more than the maximum had been left. Also if more
+ messages than the maximum are deleted they are actually removed
+ now. The second purpose of this commit should have been separated
+ out probably, but is related to the above. Again, if the number
+ of messages in a given voicemail folder exceeds the maximum set
+ limit make sure to allocate enough space for the deleted and
+ heard index tracking array. A few random fixes: There was a
+ forgotten decrement of the inprocess count in imap_store_file.
+ When using IMAP storage, do not look in the directory where file
+ based storage messages may still reside and influence the message
+ count. Ensure to use only the first format in sendmail. ABE-2516
+ ........ ................
+
+2010-10-26 16:32 +0000 [r293046-293081] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: No need to define the struct if there are no
+ users.
+
+ * channels/sig_pri.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Allow the DAHDI driver to compile, even with a
+ sufficiently older version of libpri. Fixes our Bamboo builds.
+
+2010-10-25 21:15 +0000 [r292906-292969] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/sig_pri.c: Several more defines that need to be altered
+ for compiling against an older version of libpri
+
+ * channels/sig_pri.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Allow the DAHDI driver to compile, even with a
+ sufficiently older version of libpri. Fixes our Bamboo builds.
+
+2010-10-25 19:07 +0000 [r292868] David Vossel <dvossel@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 292867 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r292867 | dvossel | 2010-10-25 14:06:21 -0500
+ (Mon, 25 Oct 2010) | 32 lines Merged revisions 292866 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r292866 | dvossel | 2010-10-25 14:05:07 -0500 (Mon, 25 Oct 2010)
+ | 27 lines This patch turns chan_local pvts into astobj2 objects.
+ chan_local does some dangerous things involving deadlock
+ avoidance. tech_pvt functions like hangup and queue_frame are
+ provided with a locked channel upon entry. Those functions are
+ completely safe as long as you don't attempt to give up that
+ channel lock, but that is impossible to guarantee due to the
+ required deadlock avoidance necessary to lock both the tech_pvt
+ and both channels involved. In the past, we have tried to account
+ for this by doing things like setting a "glare" flag that
+ indicates what function should destroy the pvt. This was used in
+ local_hangup and local_queue_frame to decided who should destroy
+ the pvt if they collided in separate threads. I have removed the
+ need to do this by converting all chan_local tech_pvts to
+ astobj2. This means we can ref a pvt before deadlock avoidance
+ and not have to worry about that pvt possibly getting destroyed
+ under us. It also cleans up where we destroy the tech_pvt. The
+ only unlink from the tech_pvt container occurs in local_hangup
+ now, which is where it should occur. Since there still may be
+ thread collisions on some functions like local_hangup after
+ deadlock avoidance, I have added some checks to detect those
+ collisions and exit appropriately. I think this patch is going to
+ solve quite a bit of weirdness we have had with local channels in
+ the past. ........ ................
+
+2010-10-22 22:35 +0000 [r292794-292825] Terry Wilson <twilson@digium.com>
+
+ * contrib/scripts/ast_tls_cert: Don't create directories without at
+ least o+x Also, making files that you are going to modify
+ read-only is dumb.
+
+ * contrib/scripts/ast_tls_cert: Make files readable only by the
+ owner
+
+2010-10-22 21:28 +0000 [r292787] Leif Madsen <lmadsen@digium.com>
+
+ * configs/res_ldap.conf.sample, contrib/scripts/asterisk.ldif, /,
+ channels/chan_sip.c: Merged revisions 292786 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r292786 | lmadsen | 2010-10-22 16:16:12 -0500 (Fri, 22 Oct 2010)
+ | 13 lines Update the LDIF file for LDAP. The LDIF file
+ asterisk.ldif was quite a bit out of date from the
+ asterisk.ldap-schema file, so I've now updated that to be in
+ sync. The asterisk.ldif file being out of sync was a problem on
+ my systems where I was doing an ldapadd to import the schema into
+ the LDAP database, and the existing file would cause problems and
+ ERROR messages when registering. Additional documention has been
+ added based on feedback in the issue I'm closing. (closes issue
+ #13861) Reported by: scramatte Patches: ldap-update.txt uploaded
+ by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec,
+ rgenthner ........
+
+2010-10-22 17:09 +0000 [r292741] Mark Michelson <mmichelson@digium.com>
+
+ * tests/test_event.c: Prevent multiple runs of event_sub_test from
+ producing false failure results. The array of test subscriptions
+ was declared "static," meaning that the data.count field would
+ retain its value between runs of the test. After the first test
+ run, this would result in false reports of test failures. I chose
+ to just remove the "static" keyword from the structure since it's
+ not a huge deal to construct this structure during each run of
+ the test. Another alternative would have been to zero out the
+ data.count fields of each test subscription instead.
+
+2010-10-22 16:49 +0000 [r292740] Terry Wilson <twilson@digium.com>
+
+ * contrib/scripts/ast_tls_cert (added): Add TLS cert helper script
+ This script is useful for quickly generating self-signed CA,
+ server, and client certificates for use with Asterisk. It is
+ still recommended to obtain certificates from a recognized
+ Certificate Authority and to develop an understanding how SSL
+ certificates work. Real security is hard work. OPTIONS: -h Show
+ this message -m Type of cert "client" or "server". Defaults to
+ server. -f Config filename (openssl config file format) -c CA
+ cert filename (creates new CA cert/key as ca.crt/ca.key if not
+ passed) -k CA key filename -C Common name (cert field) For a
+ server cert, this should be the same address that clients attempt
+ to connect to. Usually this will be the Fully Qualified Domain
+ Name, but might be the IP of the server. For a CA or client cert,
+ it is merely informational. Make sure your certs have unique
+ common names. -O Org name (cert field) An informational string
+ (company name) -o Output filename base (defaults to asterisk) -d
+ Output directory (defaults to the current directory) Example: To
+ create a CA and a server (pbx.mycompany.com) cert with output in
+ /tmp: ast_tls_cert -C pbx.mycompany.com -O "My Company" -d /tmp
+ This will create a CA cert and key as well as asterisk.pem and
+ the the two files that it is made from: asterisk.crt and
+ asterisk.key. Copy asterisk.pem and ca.crt somewhere (like
+ /etc/asterisk) and set tlscertfile=/etc/asterisk.pem and
+ tlscafile=/etc/ca.crt. Since this is a self-signed key, many
+ devices will require you to import the ca.crt file as a trusted
+ cert. To create a client cert using the CA cert created by the
+ example above: ast_tls_cert -m client -c /tmp/ca.crt -k
+ /tmp/ca.key -C "Joe User" -O \ "My Company" -d /tmp -o joe_user
+ This will create client.crt/key/pem in /tmp. Use this if your
+ device supports a client certificate. Make sure that you have the
+ ca.crt file set up as a tlscafile in the necessary Asterisk
+ configs. Make backups of all .key files in case you need them
+ later.
+
+2010-10-22 15:47 +0000 [r292704] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c, main/channel.c, channels/chan_misdn.c:
+ Connected line is not updated when chan_dahdi/sig_pri or
+ chan_misdn transfers a call. When a call is transfered by ECT or
+ implicitly by disconnect in sig_pri or implicitly by disconnect
+ in chan_misdn, the connected line information is not exchanged.
+ The connected line interception macros also need to be executed
+ if defined. The CALLER interception macro is executed for the
+ held call. The CALLEE interception macro is executed for the
+ active/ringing call. JIRA ABE-2589 JIRA SWP-2296 Patches:
+ abe_2589_c3bier.patch uploaded by rmudgett (license 664)
+ abe_2589_v1.8_v2.patch uploaded by rmudgett (license 664) Review:
+ https://reviewboard.asterisk.org/r/958/
+
+2010-10-21 22:09 +0000 [r292667] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/misdn/ie.c: Compile correctly on Linux
+ (asterisk/localtime.h depends upon asterisk/autoconfig.h loading
+ first).
+
+2010-10-21 18:13 +0000 [r292628] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * contrib/init.d/rc.suse.asterisk: Fix typo in SUSE init script.
+ Reported by: Dave Cotton on asterisk-users list.
+
+2010-10-21 16:14 +0000 [r292595] David Vossel <dvossel@digium.com>
+
+ * main/manager.c: Fixes recursive lock problem in manager.c It is
+ possible for a AMI session to freeze because of invalid use of
+ recursive locks during the EVENT processing. This patch removes
+ the unnecessary locks. (closes issue #18167) Reported by: sustav
+ Patches: manager_locking_v1.diff uploaded by dvossel (license
+ 671) Tested by: sustav
+
+2010-10-21 13:12 +0000 [r292557] Leif Madsen <lmadsen@digium.com>
+
+ * configs/res_ldap.conf.sample, /: Merged revisions 292556 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r292556 | lmadsen | 2010-10-21 08:11:52 -0500 (Thu, 21 Oct 2010)
+ | 6 lines Change res_ldap.sample.conf to match the schema.
+ (closes issue #17376) Reported by: jcovert Patches:
+ res_ldap.conf.sample.patch uploaded by jcovert (license 551)
+ ........
+
+2010-10-21 11:36 +0000 [r292523] Russell Bryant <russell@digium.com>
+
+ * res/res_config_ldap.c: Add var=value to log message on update
+ failure, and add newline. ... just for you, Leif.
+
+2010-10-21 01:02 +0000 [r292489] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Send CONNECT_ACKNOWLEDGE for CIS calls too.
+ The originator of the Q.SIG call completion signaling link was
+ not changed to the active state when the CONNECT message came in.
+ The T309 processing would immediately kill the signaling link
+ because it was not in the active state.
+
+2010-10-21 00:21 +0000 [r292413-292436] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * apps/app_voicemail.c: Application not properly unregister in
+ voicemail (closes issue #18128) Reported by: junky Patches:
+ vm_unregister.diff uploaded by junky (license 177) Tested by:
+ pabelanger, lmadsen
+
+ * apps/app_dial.c, /: Merged revisions 292412 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r292412 | pabelanger | 2010-10-20 20:05:45 -0400
+ (Wed, 20 Oct 2010) | 17 lines Merged revisions 292411 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r292411 | pabelanger | 2010-10-20 20:00:51 -0400 (Wed, 20 Oct
+ 2010) | 10 lines Record priv-recordintro as sln, not gsm This
+ removes the gsm->sln step when transcoding priv-recordintro.
+ (closes issue #18176) Reported by: pabelanger Patches:
+ chan_sip.diff uploaded by pabelanger (license 224) ........
+ ................
+
+2010-10-20 00:40 +0000 [r292376] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c: Oops. This module uses the generic timer
+ and no longer uses DAHDI. This causes a problem with the Solaris
+ and other system builds that have gcc 4.1 (where optional_api is
+ non-optional).
+
+2010-10-19 22:14 +0000 [r292343] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * contrib/scripts/install_prereq: Add resample and imap_tk
+ dependencies.
+
+2010-10-19 19:27 +0000 [r292309] Terry Wilson <twilson@digium.com>
+
+ * res/res_srtp.c, channels/chan_sip.c: Add sip show peer info about
+ crypto and remove dated comment This patch adds information about
+ the encryption setting to 'sip show peers' and removes an
+ out-of-date comment from res_srtp.c and instead directs users to
+ the proper documentation. (closes issue #18140) Reported by:
+ chodorenko
+
+2010-10-21 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.0 Released.
+
+2010-10-18 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.0-rc5 Released.
+
+2010-10-18 22:02 +0000 [r292230] Leif Madsen <lmadsen@digium.com>
+
+ * sounds/Makefile, /: Merged revisions 292229 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r292229 | lmadsen | 2010-10-18 17:01:16 -0500 (Mon, 18 Oct 2010)
+ | 3 lines Fix typo in the sounds/Makefile. (Issue #17426)
+ ........
+
+2010-10-18 21:55 +0000 [r292227] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 292226 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r292226 | jpeeler | 2010-10-18 16:54:38 -0500
+ (Mon, 18 Oct 2010) | 18 lines Merged revisions 292223 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r292223 | jpeeler | 2010-10-18 16:50:30 -0500 (Mon, 18 Oct 2010)
+ | 11 lines Fix improper operator key acceptance and clean up temp
+ recording files. This is a fix for when pressing the operator key
+ after recording an unavailable, busy, name, or temporary message
+ in mailbox options. The operator key should not be accepted here,
+ but should be allowed during the message recording. If the
+ operator key is pressed during ensure the file is saved or
+ deleted as apporopriate. Also, ensure removal of temporary
+ recorded files after an early hang up or when message acceptance
+ confirmation times out. ABE-2518 ........ ................
+
+2010-10-18 21:51 +0000 [r292225] Leif Madsen <lmadsen@digium.com>
+
+ * sounds/sounds.xml, sounds/Makefile, /: Merged revisions 292224
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r292224 | lmadsen | 2010-10-18 16:50:47 -0500
+ (Mon, 18 Oct 2010) | 17 lines Merged revisions 292222 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r292222 | lmadsen | 2010-10-18 16:47:25 -0500 (Mon, 18 Oct 2010)
+ | 9 lines Add support for the new English (Australian Accent)
+ sound files. (closes issue #17426) Reported by: camsown Patches:
+ core-sounds-en_AU.txt uploaded by camsown (license 1050)
+ add_AU_sounds.patch.txt uploaded by lmadsen (license 10) Tested
+ by: camsown, lmadsen, jtodd, qwell ........ ................
+
+2010-10-18 19:50 +0000 [r292188] Russell Bryant <russell@digium.com>
+
+ * main/netsock2.c: Resolve some compiler errors in
+ ast_sockaddr_is_any(). These errors came up once this function
+ was used from within netsock2.c. The errors were like the
+ following: netsock2.c:393: error: dereferencing pointer
+ ‘({anonymous})’ does break strict-aliasing rules The usage of a
+ union here avoids this problem.
+
+2010-10-18 19:16 +0000 [r292155] David Vossel <dvossel@digium.com>
+
+ * main/netsock2.c: Fixes build error for systems not supporting
+ IPV6_TCLASS.
+
+2010-10-18 17:15 +0000 [r292122] Matthew Nicholson <mnicholson@digium.com>
+
+ * addons/chan_mobile.c: Fix the cmgr parser. (closes issue 0018152)
+ Reported by: menschentier
+
+2010-10-18 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.0-rc4 Released
+
+2010-10-18 16:02 +0000 [r292085] David Vossel <dvossel@digium.com>
+
+ * main/netsock2.c: Fixes qos settings for sockets bound to any IPv6
+ or IPv4 address. (closes issue #18099) Reported by: jamesnet
+ Patches: issues_18099_v3.diff uploaded by dvossel (license 671
+
+2010-10-18 15:32 +0000 [r292083] Jeff Peeler <jpeeler@digium.com>
+
+ * pbx/pbx_spool.c: Disable use of inotify for call file handling as
+ it is not working properly. (related to #18089)
+
+2010-10-16 10:47 +0000 [r292050] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * res/res_musiconhold.c, /, configs/musiconhold.conf.sample: Merged
+ revisions 292049 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r292049 | tzafrir | 2010-10-16 12:03:04 +0200 (ש', 16 אוק 2010) |
+ 15 lines Base directory for MOH should be ASTDATADIR If the
+ directive 'directory' is relative, make it relative to the
+ datadir, rather than to the varlibdir. In the sample
+ configuration it is relative ('moh'). This has no effect unless
+ you have actively set the datadir explicitly (at build time or at
+ run time). (closes issue #16906) Patches: moh_datadir uploaded by
+ tzafrir (license 46) Review:
+ https://reviewboard.asterisk.org/r/974/ ........
+
+2010-10-15 21:40 +0000 [r292016] Terry Wilson <twilson@digium.com>
+
+ * res/res_srtp.c: Ref/unref res_srtp when we create/destroy a
+ session This avoids unhappy crashing when we try to 'core stop
+ gracefully' and res_srtp tries to unload before chan_sip does.
+ Thanks, Russell! (closes issue #18085) Reported by: st
+
+2010-10-15 20:12 +0000 [r291942] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Fixes peer's host port information being
+ lost on sip reload. (closes issue #18135) Reported by: lmadsen
+ Patches: crazy_ports_v2.diff uploaded by dvossel (license 671)
+ Tested by: lmadsen
+
+2010-10-15 19:50 +0000 [r291940] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * configs/gtalk.conf.sample, /: Merged revisions 291939 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291939 | pabelanger | 2010-10-15 15:35:20 -0400
+ (Fri, 15 Oct 2010) | 9 lines Merged revisions 291938 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r291938 | pabelanger | 2010-10-15 15:30:41 -0400 (Fri,
+ 15 Oct 2010) | 2 lines Clean up formatting. ........
+ ................
+
+2010-10-15 16:39 +0000 [r291905] Terry Wilson <twilson@digium.com>
+
+ * res/res_jabber.c, /: Merged revisions 291904 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r291904 | twilson | 2010-10-15 09:16:57 -0700 (Fri, 15 Oct 2010)
+ | 7 lines Don't crash or deadlock on module unload We can't hold
+ the lock while pthread_join is called since aji_log_hook will
+ attempt to lock from the other therad. We reorder the
+ pthread_join and ast_aji_disconnect so that we don't do an
+ SSL_read() while SSL_shutdown is running, causing a crash.
+ ........
+
+2010-10-14 22:09 +0000 [r291827-291829] David Vossel <dvossel@digium.com>
+
+ * main/netsock2.c: Set TCLASS field of IPv6 header when sip qos
+ options are set. (closes issue #18099) Reported by: jamesnet
+ Patches: issues_18099_v2.diff uploaded by dvossel (license 671)
+ Tested by: dvossel, jamesnet
+
+ * channels/chan_gtalk.c: Safer xml parsing, treat all clients the
+ same, and better local candidate selection. The gtalk channel
+ driver was doing several unsafe operations in regards to how it
+ parsed incoming XML messages. I have cleaned that code up so it
+ should be much safer now. We now treat all clients types the
+ same. We have no reason to distinguish between GMAIL and GOOGLE
+ VOICE clients anymore because they all work the same way. I also
+ modified how the local ip is found. If no bindaddress is provided
+ in the config file, we attempt to determine the local ip we would
+ use to connect to google.com. If that fails, then we fall back to
+ the ast_find_ourip() function as a last resort. Using the new
+ method makes it much less likely that we would ever advertise a
+ local RTP candidate as a loopback address.
+
+2010-10-14 18:45 +0000 [r291791] Jeff Peeler <jpeeler@digium.com>
+
+ * main/stdtime/localtime.c: Add missing ifdefs for test framework
+ and new locale code. (closes issue #18137) Reported by: ovi
+ Patches: 18137_test_framework_ifdef.patch uploaded by wdoekes
+ (license 717) 18137_localelist_warning.patch uploaded by wdoekes
+ (license 717) Tested by: ovi
+
+2010-10-14 15:15 +0000 [r291758] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * channels/chan_gtalk.c, channels/chan_jingle.c,
+ include/asterisk/acl.h, channels/chan_sip.c,
+ channels/chan_h323.c, main/acl.c: Add the ability for
+ ast_find_ourip to return IPv4, IPv6 or both. While testing
+ chan_gtalk I noticed jabber was using my IPv6 address and not
+ IPv4. When using bindaddr=0.0.0.0 it is possible for
+ ast_find_ourip() to return both IPv6 and IPv4 results. Adding a
+ family parameter gives you the ablility to choose. Since
+ jabber/gtalk/h323 do not support IPv6, we should only return IPv4
+ results. Review: https://reviewboard.asterisk.org/r/973/
+
+2010-10-14 12:08 +0000 [r291725] Russell Bryant <russell@digium.com>
+
+ * doc/tex/secure-calls.tex: Fix a typo - s/seucre/secure/
+
+2010-10-13 23:45 +0000 [r291656] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/sig_analog.h: Merged revisions 291655 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291655 | rmudgett | 2010-10-13 18:36:50 -0500
+ (Wed, 13 Oct 2010) | 27 lines Merged revisions 291643 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291643 | rmudgett | 2010-10-13 18:29:58 -0500 (Wed, 13 Oct 2010)
+ | 20 lines Deadlock between dahdi_exception() and
+ dahdi_indicate(). There is a deadlock between dahdi_exception()
+ and dahdi_indicate() for analog ports. The call-waiting and
+ three-way-calling feature can experience deadlock if these
+ features are trying to do something and an event from the bridged
+ channel happens at the same time. Deadlock avoidance code added
+ to obtain necessary channel locks before attemting an operation
+ with call-waiting and three-way-calling. (closes issue #16847)
+ Reported by: shin-shoryuken Patches: issue_16847_v1.4.patch
+ uploaded by rmudgett (license 664) issue_16847_v1.6.2.patch
+ uploaded by rmudgett (license 664) issue_16847_v1.8_v2.patch
+ uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett
+ Review: https://reviewboard.asterisk.org/r/971/ ........
+ ................
+
+2010-10-13 23:01 +0000 [r291581] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, /: Merged revisions 291580 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291580 | twilson | 2010-10-13 15:58:43 -0700
+ (Wed, 13 Oct 2010) | 28 lines Merged revisions 291577 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010)
+ | 21 lines Don't ignore frames that have been queued when
+ softhangup'd When an outgoing call is answered and hung up by the
+ far end *very* quickly, we may not read any frames and therefor
+ end up with a call that displays the wrong
+ disposition/DIALSTATUS. The reason is because ast_queue_hangup()
+ immediately sets the _softhangup flag on the channel and then
+ queues the HANGUP control frame, but __ast_read refuses to read
+ any frames if ast_check_hangup() indicates that a hangup request
+ has been made (which it will if _softhangup is set). So, we end
+ up losing control frames. This change makes __ast_read continue
+ to read frames even if a soft hangup has been requested. It
+ queues a hangup frame to make sure that __ast_read() will still
+ eventually return NULL. Much thanks to David Vossel for all of
+ the reviews, discussion, and help! (closes issue #16946) Reported
+ by: davidw Review: https://reviewboard.asterisk.org/r/740/
+ ........ ................
+
+2010-10-13 22:46 +0000 [r291578] David Vossel <dvossel@digium.com>
+
+ * channels/chan_gtalk.c: More fixup for chan_gtalk. This patch
+ makes the xml parsing safer.
+
+2010-10-13 22:24 +0000 [r291575] Terry Wilson <twilson@digium.com>
+
+ * Makefile, static-http/mantest.html (added): Add a simple AMI
+ client web page This patch uses the XML docs to parse all of the
+ available AMI commands and allows you to enter the command name
+ and be presented with a form with the available fields. You can
+ then rapidly tab through the fields and submit the command and
+ view the response. It is much faster/easier than having to use
+ telnet for testing purposes.
+
+2010-10-13 20:21 +0000 [r291469-291541] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: The chan_dahdi faxdetect option only works
+ for the first FAX call. The chan_dahdi faxdetect option only
+ works for the first call. After that the option no longer works.
+ The struct dahdi_pvt.callprogress member is the encoded user
+ config setting for the callprogress and faxdetect config options.
+ Changing this value alters the configuration for all following
+ calls until the chan_dahdi.conf file is reloaded. * Fixed the
+ chan_dahdi ast_channel_setoption callback to not change the users
+ faxdetect config setting except for the current call. * Fixed the
+ chan_dahdi ast_channel_queryoption callback to read the active
+ DSP setting of the faxdetect option. * Made actually disable the
+ active faxdetect DSP setting for the current call on the analog
+ port. my_handle_dtmfup() is used for normal analog ports.
+ dahdi_handle_dtmfup() is the legacy code and is no longer used
+ unless in a radio mode. (closes issue #18116) Reported by:
+ seandarcy Patches: issue18116_v1.8.patch uploaded by rmudgett
+ (license 664) Review: https://reviewboard.asterisk.org/r/972/
+
+ * channels/chan_misdn.c: Merged revision 291504 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r291504 | rmudgett | 2010-10-13 13:30:21 -0500 (Wed,
+ 13 Oct 2010) | 11 lines Hold off ast_hangup() from destroying the
+ ast_channel. Must get the ast_channel lock before proceeding with
+ release_chan() and release_chan_early() to hold off ast_hangup()
+ from destroying the ast_channel. Missed this change for -r291468.
+ JIRA ABE-2598 JIRA SWP-2317 ..........
+
+ * channels/chan_misdn.c: Merge revision 291468 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r291468 | rmudgett | 2010-10-13 12:39:02 -0500 (Wed,
+ 13 Oct 2010) | 16 lines Memory overwrites when releasing mISDN
+ call. Phone <--> Asterisk <-- ALERTING --> DISCONNECT <-- RELEASE
+ --> RELEASE_COMPLETE * Add lock protection around channel list
+ for find/add/delete operations. * Protect misdn_hangup() from
+ release_chan() and vise versa using the release_lock. JIRA
+ ABE-2598 JIRA SWP-2317 ..........
+
+2010-10-13 15:46 +0000 [r291394] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 291393 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291393 | russell | 2010-10-13 10:29:21 -0500
+ (Wed, 13 Oct 2010) | 13 lines Merged revisions 291392 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010)
+ | 6 lines Lock pvt so pvt->owner can't disappear when queueing up
+ a frame. This fixes a crash due to a hangup race condition.
+ ABE-2601 ........ ................
+
+2010-10-12 17:20 +0000 [r291284] Leif Madsen <lmadsen@digium.com>
+
+ * configs/phoneprov.conf.sample, /: Merged revisions 291280 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r291280 | lmadsen | 2010-10-12 12:20:02 -0500 (Tue, 12 Oct 2010)
+ | 7 lines Add undocumented variables to phoneprov.conf.sample
+ (closes issue #18107) Reported by: lathama Patches:
+ phoneprov.conf.sample.diff uploaded by lathama (license 1028)
+ ........
+
+2010-10-12 17:06 +0000 [r291265] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/acl.c: Merged revisions 291264 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291264 | tilghman | 2010-10-12 12:05:31 -0500
+ (Tue, 12 Oct 2010) | 9 lines Merged revisions 291263 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r291263 | tilghman | 2010-10-12 11:55:30 -0500 (Tue, 12
+ Oct 2010) | 2 lines Oops, incorrect range (although unallocated
+ at ARIN) ........ ................
+
+2010-10-12 16:08 +0000 [r291230] Leif Madsen <lmadsen@digium.com>
+
+ * configs/manager.conf.sample, /: Merged revisions 291229 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r291229 | lmadsen | 2010-10-12 11:07:28 -0500 (Tue, 12 Oct 2010)
+ | 2 lines Add documention that mentions options are defined but
+ not used. (Issue #18101) ........
+
+2010-10-12 15:58 +0000 [r291192-291227] David Vossel <dvossel@digium.com>
+
+ * main/manager.c: Fixes manager.c crash. This issue was caused by
+ improper use of the mansession lock and manession_session lock.
+ These two structures are confusing to begin with so I'm not
+ surprised this occurred. I fixed this by consistently making sure
+ we use each of these locks only to protect the data in the
+ corresponding structure. We had mismatched usage of these locks
+ which resulted in no mutual exclusivity occurring at all. (closes
+ issue #17994) Reported by: vrban Patches:
+ mansession_locking_fix.diff uploaded by dvossel (license 671)
+ Tested by: vrban
+
+ * CHANGES: Update CHANGES to reflect new gtalk.conf options.
+
+ * channels/chan_gtalk.c, include/asterisk/stun.h,
+ configs/gtalk.conf.sample, res/res_stun_monitor.c: Gtalk
+ enhancements and general code cleanup. This patch includes
+ several chan_gtalk enhancements. Two new gtalk.conf options have
+ been added, externip and stunadd. Setting externip allows us to
+ manually specify what the external IP address is outside of a NAT
+ environment. Setting the stunaddr option to a valid stun server
+ allows for that external ip to be retrieved via a STUN server
+ automatically. This external IP is then advertised during call
+ setup as a possible candidate. I have also attempted to clean up
+ chan_gtalk's code so it meets our coding guidelines. During this
+ cleanup I noticed several things that need to be done in the code
+ and made a TODO section at the top of the file.
+
+2010-10-11 18:51 +0000 [r291075-291113] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Move declaration closer to where now used.
+
+ * /, channels/chan_sip.c: Merged revisions 291110-291111 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291110 | rmudgett | 2010-10-11 13:34:22 -0500
+ (Mon, 11 Oct 2010) | 9 lines Merged revisions 291109 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11
+ Oct 2010) | 1 line Add missing unlock to an exception condition
+ in reload_config(). ........ ................ r291111 | rmudgett
+ | 2010-10-11 13:39:06 -0500 (Mon, 11 Oct 2010) | 1 line Make exit
+ from handle_request_do() consistent. ................
+
+ * main/cli.c, /: Merged revisions 291073 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r291073 | rmudgett | 2010-10-11 11:39:17 -0500 (Mon, 11 Oct 2010)
+ | 15 lines Fixed infinite loop in verbose/debug message output.
+ Setting the module/filename specific message level and then
+ changing it resulted in the linked list being looped on itself.
+ Traversing this linked list is an infinite loop if what you are
+ looking for is not in the list. Also plugged some CLI parsing
+ holes in the associated CLI command: * Removing a nonexistent
+ module from the list actually added it with a level of zero. *
+ Setting the non-module specific level to zero is now equivalent
+ to setting it to "off" as documented. ........
+
+2010-10-09 23:25 +0000 [r291038] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample: Add missing
+ option to set calls to be logged in GMT/UTC.
+
+2010-10-09 15:00 +0000 [r291005-291037] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/oochannels.c: small correction for verbose
+ print h.323 packets
+
+ * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
+ addons/ooh323c/src/ooh245.c: Added fast start and h.245 tunneling
+ options per user and peer. Added options for faststart/h.245
+ tunneling per user/peer, properly handle these and global
+ options, correction of handling fs/tunneling fields in signalling
+ responses (issue #17972) Reported by: salecha Patches:
+ fs-tunnel-per-point-3.patch uploaded by may213 (license 454)
+ Tested by: may213, salecha
+
+2010-10-08 20:44 +0000 [r290973] David Vossel <dvossel@digium.com>
+
+ * channels/chan_gtalk.c: Make outbound Google Voice calls. This
+ patch allows for outbound Google Voice calls to be dialed from
+ Asterisk using chan_gtalk. Below is an example dialstring. exten
+ -> blah,1,Dial(Gtalk/asterisk/+15552225555@voice.google.com,,) In
+ this example, 'asterisk' is the jabber.conf profile configured to
+ connect to your gmail account. In order to receive Google Voice
+ calls make sure to enable 'allowguest=yes' in gtalk.conf.
+
+2010-10-08 15:49 +0000 [r290937-290938] Erin Spiceland <erin@thespicelands.com>
+
+ * addons/res_config_mysql.c: Parentheses around assignment used as
+ truth value, introduced in r290937.
+
+ * addons/res_config_mysql.c, addons/app_mysql.c,
+ configs/res_config_mysql.conf.sample: Add option to
+ res_config_mysql and app_mysql to specify a character set that
+ MySQL should use. (closes issue 17948) Reported by qmax.
+
+2010-10-08 02:56 +0000 [r290864] Jeff Peeler <jpeeler@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 290863 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290863 | jpeeler | 2010-10-07 21:45:44 -0500
+ (Thu, 07 Oct 2010) | 16 lines Merged revisions 290862 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r290862 | jpeeler | 2010-10-07 21:35:29 -0500 (Thu, 07 Oct 2010)
+ | 9 lines Ensure editline cleanup occurs when Ctrl-C is pressed
+ at control console. A recent change was made to avoid a race
+ condition on shutdown which only called the end functions from
+ the console thread. However, when pressing Ctrl-C the quit
+ handler is called from the signal handler thread. (closes issue
+ #17698) Reported by: jmls ........ ................
+
+2010-10-07 22:38 +0000 [r290828-290829] David Vossel <dvossel@digium.com>
+
+ * channels/chan_gtalk.c: Add Philippe Sultan to chan_gtalk author
+ list. Philippe has made some notable contributions to the gtalk
+ channel driver. His name deserves to be listed amoung the authors
+ of that file. Thanks Philippe!
+
+ * channels/chan_gtalk.c: Outbound gtalk calls now work correctly.
+ There was a problem with how the candidates were being built on
+ an outbound call. This patch fixes that.
+
+2010-10-07 20:58 +0000 [r290752] Jason Parker <jparker@digium.com>
+
+ * autoconf/ast_ext_lib.m4, /, configure,
+ include/asterisk/autoconfig.h.in: Merged revisions 290751 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290751 | qwell | 2010-10-07 15:57:14 -0500
+ (Thu, 07 Oct 2010) | 16 lines Merged revisions 290750 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r290750 | qwell | 2010-10-07 15:56:04 -0500 (Thu, 07 Oct 2010) |
+ 9 lines Allow PRI to build properly when using --with-pri. Use
+ the directories found for the parent when using lib dependencies.
+ (closes issue #17314) Reported by: tzafrir Patches:
+ 17314-withdeps.diff uploaded by qwell (license 4) ........
+ ................
+
+2010-10-07 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.0-rc3 Released.
+
+2010-10-07 11:00 +0000 [r290713] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c, /: Merged revisions 290712 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r290712 | russell | 2010-10-07 12:53:56 +0200 (Thu, 07 Oct 2010)
+ | 4 lines Don't crash when Set() is called without a value.
+ Review: https://reviewboard.asterisk.org/r/949/ ........
+
+2010-10-06 21:22 +0000 [r290648-290674] David Vossel <dvossel@digium.com>
+
+ * channels/chan_gtalk.c: Fixes commented out code to use #if 0
+ instead. Thanks to rmudgett for catching this!
+
+ * channels/chan_gtalk.c: Fixes gtalk outbound DTMF to work
+ properly. Outbound DTMF with gtalk needs to be done within the
+ RTP stream. I discovered this after investigating a packet
+ capture from the gmail client. Instead of performing jingle
+ signaling DTMF, the gtalk servers expect all DTMF to arrive on
+ the RTP stream using RFC2833 way of doing things. Chan_gtalk also
+ had an issue with negotiating RTP payload type 106 for the
+ telephony-event and then sending DTMF as payload 101. This has
+ been resolved by always negotiating 101 as the payload type like
+ we do everywhere else. With this patch, incoming google voice
+ calls forwarded to Asterisk via gtalk work.
+
+2010-10-06 18:50 +0000 [r290614] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c: Merged revision 290613 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r290613 | rmudgett | 2010-10-06 13:42:41 -0500 (Wed,
+ 06 Oct 2010) | 5 lines Eliminate a redundant test for
+ AST_CONTROL_REDIRECTING. Eliminate redundant test for
+ AST_CONTROL_REDIRECTING that prevents running the redirecting
+ interception macro if it is defined. ..........
+
+2010-10-06 13:49 +0000 [r290576] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/file.c: Merged revisions 290575 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r290575 | tilghman | 2010-10-06 08:48:27 -0500 (Wed, 06 Oct 2010)
+ | 8 lines Allow streaming audio from a pipe. (closes issue
+ #18001) Reported by: jamicque Patches:
+ 20100926__issue18001.diff.txt uploaded by tilghman (license 14)
+ Tested by: jamicque ........
+
+2010-10-06 04:35 +0000 [r290542] Terry Wilson <twilson@digium.com>
+
+ * res/res_rtp_asterisk.c: Don't try to send RTP when remote_address
+ is null It is possible for ast_rtp_stop() to be called which will
+ clear the remote address and cause the sendto to fail and spam
+ warnings. Don't send in this case.
+
+2010-10-05 22:23 +0000 [r290479-290506] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: Fixes uninitialized memory problem in 'iax2
+ set debug peer' option.
+
+ * include/asterisk/jingle.h, channels/chan_gtalk.c,
+ res/res_jabber.c, include/asterisk/jabber.h: Fixes chan_gtalk to
+ work with gmail client This patch was written by Philippe Sultan
+ (phsultan). Thanks for keeping this up to date!
+
+2010-10-05 20:23 +0000 [r290408] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_jabber.c, /: Merged revisions 290396 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290396 | tilghman | 2010-10-05 15:21:02 -0500
+ (Tue, 05 Oct 2010) | 15 lines Merged revisions 290392 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r290392 | tilghman | 2010-10-05 15:20:07 -0500 (Tue, 05 Oct 2010)
+ | 8 lines Fix a crash by ensuring that we don't alter memory
+ after it's freed. (closes issue #17387) Reported by: jmls
+ Patches: 20100726__issue17387.diff.txt uploaded by tilghman
+ (license 14) Tested by: jmls ........ ................
+
+2010-10-05 20:09 +0000 [r290376-290378] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: Resolves dnsmgr memory corruption in
+ chan_iax2. (closes issue #17902) Reported by: afried Patches:
+ issue_17902.rev1.txt uploaded by russell (license 2) Tested by:
+ afried, russell, dvossel Review:
+ https://reviewboard.asterisk.org/r/965/
+
+ * /, apps/app_directed_pickup.c: Merged revisions 290375 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r290375 | dvossel | 2010-10-05 14:54:50 -0500 (Tue, 05 Oct 2010)
+ | 10 lines Fixes PickupChan() not working with full channel name.
+ (closes issue #18011) Reported by: schern Patches:
+ app_directed_pickup.c.2.patch uploaded by schern (license 995)
+ app_directed_pickup.c.trunk.patch uploaded by schern (license
+ 995) Tested by: schern, dvossel ........
+
+2010-10-05 14:15 +0000 [r290066-290289] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, configure.ac: Restore run directory for OS X, as well
+ as standardizing some other paths to Mac OS X.
+
+ * pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5,
+ pbx/ael/ael-test/ref.ael-test19,
+ pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c, main/pbx.c,
+ pbx/ael/ael-test/ref.ael-vtest17, /,
+ pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
+ pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3:
+ Merged revisions 290254 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r290254 | tilghman | 2010-10-04 18:14:59 -0500 (Mon, 04 Oct 2010)
+ | 11 lines Change new pattern matcher to regard dashes the same
+ as the old pattern matcher -- as visual candy to be ignored. Also
+ change the AEL parser to not generate dashes within extensions,
+ as those dashes would be ignored. Update the AEL tests to match
+ this behavior. (closes issue #17366) Reported by: murf Patches:
+ 20100727__issue17366.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman ........
+
+ * /, configure, configure.ac: Merged revisions 290201 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290201 | tilghman | 2010-10-04 15:22:03 -0500
+ (Mon, 04 Oct 2010) | 9 lines Merged revisions 290177 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r290177 | tilghman | 2010-10-04 15:15:26 -0500 (Mon, 04
+ Oct 2010) | 2 lines Fixing Mac OS X auto-builder. ........
+ ................
+
+ * /, configure, configure.ac: Merged revisions 290101 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290101 | tilghman | 2010-10-03 16:06:58 -0500
+ (Sun, 03 Oct 2010) | 9 lines Merged revisions 290100 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r290100 | tilghman | 2010-10-03 16:04:29 -0500 (Sun, 03
+ Oct 2010) | 2 lines Automatically re-run configure test for
+ menuselect, when the relevant makeopts settings change. ........
+ ................
+
+ * pbx/pbx_spool.c: Get notification only when file is closed, not
+ when created. (closes issue #17924) Reported by: mkeuter Patches:
+ asterisk-1.8-bugid17924.patch uploaded by abelbeck (license 946)
+ Tested by: abelbeck
+
+2010-10-02 17:57 +0000 [r290026] Kevin P. Fleming <kpfleming@digium.com>
+
+ * contrib/scripts/get_mp3_source.sh: Allow users to pass additional
+ arguments to the Subversion command that obtains the MP-3 source
+ code. (reported on IRC by jmls)
+
+2010-10-02 08:56 +0000 [r289951] Olle Johansson <oej@edvina.net>
+
+ * main/manager.c, /: Merged revisions 289950 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289950 | oej | 2010-10-02 10:52:03 +0200 (Lör,
+ 02 Okt 2010) | 9 lines Merged revisions 289949 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289949 | oej | 2010-10-02 10:50:05 +0200 (Lör, 02 Okt 2010) | 2
+ lines Add documentation for undocumented option to AMI action
+ originate ........ ................
+
+2010-10-02 04:46 +0000 [r289875] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 289874 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289874 | tilghman | 2010-10-01 23:45:49 -0500
+ (Fri, 01 Oct 2010) | 15 lines Merged revisions 289873 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01 Oct 2010)
+ | 8 lines When forwarding a message, a prepend means that the
+ filesystem will always have a better copy. (closes issue #17803)
+ Reported by: dpetersen Patches: 20100923__issue17803.diff.txt
+ uploaded by tilghman (license 14) Tested by: dpetersen ........
+ ................
+
+2010-10-02 02:43 +0000 [r289840] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c,
+ main/rtp_engine.c, /, channels/chan_sip.c: Merged revisions
+ 289798 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289798 | jpeeler | 2010-10-01 18:01:31 -0500
+ (Fri, 01 Oct 2010) | 22 lines Merged revisions 289797 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010)
+ | 15 lines Change RFC2833 DTMF event duration on end to report
+ actual elapsed time. The scenario here is with a non P2P early
+ media session. The reported time length of DTMF presses are
+ coming up short when sending to the remote side. Currently the
+ event duration is a running total that is incremented when
+ sending continuation packets. These continuation packets are only
+ triggered upon incoming media from the remote side, which means
+ that the running total probably is not going to end up matching
+ the actual length of time Asterisk received DTMF. This patch
+ changes the end event duration to be lengthened if it is detected
+ that the end event is going to come up short. Review:
+ https://reviewboard.asterisk.org/r/957/ ABE-2476 ........
+ ................
+
+2010-10-01 17:19 +0000 [r289718] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * res/res_jabber.c, /, configs/jabber.conf.sample: Merged revisions
+ 289704 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289704 | pabelanger | 2010-10-01 13:09:03 -0400
+ (Fri, 01 Oct 2010) | 13 lines Merged revisions 289703 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289703 | pabelanger | 2010-10-01 13:03:11 -0400 (Fri, 01 Oct
+ 2010) | 6 lines Disable debugging by default and reformat .config
+ file. Review: https://reviewboard.asterisk.org/r/929/ ........
+ ................
+
+2010-10-01 16:22 +0000 [r289701] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 289700 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289700 | jpeeler | 2010-10-01 11:21:04 -0500
+ (Fri, 01 Oct 2010) | 21 lines Merged revisions 289699 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010)
+ | 14 lines Ensure user portion of SIP URI matches dialplan when
+ using encoded characters. This commit takes a simliar approach to
+ 288112 and checks the dialplan to determine the proper action for
+ an incoming contact header as to whether or not it should be
+ decoded or not. sip_new was blindly always decoding the
+ extension, which also caused the outgoing contact header to be
+ incorrect as well as failing to match the encoded extension in
+ the dialplan. (closes issue #17892) Reported by: wdoekes Patches:
+ bug17892-1.patch uploaded by jpeeler (license 325) Tested by:
+ wdoekes ........ ................
+
+2010-10-01 09:42 +0000 [r289622] Stefan Schmidt <sst@sil.at>
+
+ * channels/chan_sip.c: don't iterate through all dialogs to find
+ and delete old subscribes On every incoming subscribe there is a
+ iteration through all dialogs to find old subscribes and delete
+ them. This is slow and not RFC conform. This was only needed in
+ 1.2 cause a subscribe was not deleted when a dialog was
+ destroyed, after 1.4 a subscribe get removed when its dialog is
+ destroyed. (closes issue #17950) Reported by: schmidts Tested by:
+ schmidts Review: https://reviewboard.asterisk.org/r/901/
+
+2010-09-30 20:23 +0000 [r289581] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_env.c: Solaris fixes.
+
+2010-09-30 19:53 +0000 [r289554] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 289553 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r289553 | mnicholson | 2010-09-30 14:51:27 -0500 (Thu, 30 Sep
+ 2010) | 4 lines Properly handle channel allocation failures duing
+ invites with replaces. ABE-2588 ........
+
+2010-09-30 19:28 +0000 [r289549] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: Merged revision 289547 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r289547 | rmudgett | 2010-09-30 14:16:36 -0500 (Thu,
+ 30 Sep 2010) | 10 lines In chan_misdn, the
+ DivertingLegInformation2 DivertingNr is garbage when the number
+ is restricted. The same thing happens with
+ DivertingLegInformation1 DivertedTo number. The
+ misdn_PresentedNumberUnscreened_extract() extracted the
+ Unscreened PartyNumber field unconditionally. It now checks the
+ presented number unscreened type to see if the PartyNumber was
+ even present. JIRA ABE-2595 ..........
+
+2010-09-30 17:50 +0000 [r289543] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/localtime.h, main/stdtime/localtime.c,
+ tests/test_time.c, tests/test_utils.c, res/res_agi.c: More
+ Solaris compatibility fixes
+
+2010-09-30 15:39 +0000 [r289426] Russell Bryant <russell@digium.com>
+
+ * apps/app_sms.c, /: Merged revisions 289425 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289425 | russell | 2010-09-30 10:37:29 -0500
+ (Thu, 30 Sep 2010) | 15 lines Merged revisions 289424 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289424 | russell | 2010-09-30 10:34:29 -0500 (Thu, 30 Sep 2010)
+ | 8 lines Fix a crash in app_sms. Since the data being passed to
+ the generator callback is on the stack of the SMS() application,
+ we must ensure that the generator is stopped before the
+ application exits. ABE-2587 ........ ................
+
+2010-09-29 21:12 +0000 [r289340] Jason Parker <jparker@digium.com>
+
+ * main/channel.c, /, main/features.c: Merged revisions 289339 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289339 | qwell | 2010-09-29 16:03:47 -0500
+ (Wed, 29 Sep 2010) | 15 lines Merged revisions 289338 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289338 | qwell | 2010-09-29 15:56:26 -0500 (Wed, 29 Sep 2010) |
+ 8 lines Allow a manager originate to succeed on forwarded
+ devices. The timeout to wait for an answer was being set to 0
+ when a device forwarded to another extension. We don't always
+ need the timeout set like this, so make it an optional parameter,
+ and don't use it in this case. ABE-2544 ........ ................
+
+2010-09-29 20:27 +0000 [r289336] Leif Madsen <lmadsen@digium.com>
+
+ * configs/res_ldap.conf.sample, /: Merged revisions 289334 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r289334 | lmadsen | 2010-09-29 15:24:47 -0500 (Wed, 29 Sep 2010)
+ | 1 line Update sample documentation to note md5secret
+ requirements. ........
+
+2010-09-29 20:20 +0000 [r289333] Russell Bryant <russell@digium.com>
+
+ * res/res_config_ldap.c, /: Merged revisions 289332 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r289332 | russell | 2010-09-29 15:15:57 -0500 (Wed, 29
+ Sep 2010) | 4 lines Don't completely ignore md5secret from LDAP
+ if the value does not begin with {md5}. This fixes a problem that
+ lmadsen ran in to where md5secret was not working for him.
+ ........
+
+2010-09-29 17:53 +0000 [r289268-289300] Matthew Nicholson <mnicholson@digium.com>
+
+ * configs/res_fax.conf.sample: Add 'ecm' to the sample fax config
+ file
+
+ * main/channel.c: Update the CDR record when
+ ast_channel_set_caller_event() is called (related to issue
+ #17569) Reported by: tbelder
+
+2010-09-29 16:16 +0000 [r289253] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Make development error message indicate which
+ channel.
+
+2010-09-29 15:04 +0000 [r289179] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, /: Merged revisions 289178 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289178 | mnicholson | 2010-09-29 10:04:11 -0500
+ (Wed, 29 Sep 2010) | 15 lines Merged revisions 289177 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289177 | mnicholson | 2010-09-29 10:03:27 -0500 (Wed, 29 Sep
+ 2010) | 8 lines Set the caller id on CDRs when it is set on the
+ parent channel. (closes issue #17569) Reported by: tbelder
+ Patches: 17569.diff uploaded by tbelder (license 618) Tested by:
+ tbelder ........ ................
+
+2010-09-28 18:18 +0000 [r289104] Tilghman Lesher <tlesher@digium.com>
+
+ * makeopts.in, apps/app_voicemail.c, Makefile, tests/test_time.c,
+ configure, include/asterisk/autoconfig.h.in,
+ include/asterisk/compat.h, main/strcompat.c, tests/test_utils.c,
+ configure.ac: Solaris compatibility fixes Review:
+ https://reviewboard.asterisk.org/r/942/
+
+2010-09-28 18:18 +0000 [r289099] Brett Bryant <bbryant@digium.com>
+
+ * main/channel.c, /: Merged revisions 289095 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289095 | bbryant | 2010-09-28 14:14:19 -0400
+ (Tue, 28 Sep 2010) | 21 lines Merged revisions 289094 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289094 | bbryant | 2010-09-28 14:10:19 -0400 (Tue, 28 Sep 2010)
+ | 14 lines Fixes an issue with the Newchannel AMI event during
+ the Masquerading process. Fixes an issue with the Newchannel AMI
+ event during the Masquerading process, where no Newchannel AMI
+ event was generated for the psuedo channel used during the
+ masquerading process. (closes issue #17987) Reported by:
+ RadicAlish Patches: newchannel.patch.txt uploaded by RadicAlish
+ (license 1122) Tested by: RadicAlish Review:
+ https://reviewboard.asterisk.org/r/937/ ........ ................
+
+2010-09-28 01:04 +0000 [r289054-289057] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Avoid deadlock processing incoming AOC-E
+ messages. Deadlock avoidance for the owner channel was not done
+ when processing incoming AOC-E messages.
+
+ * channels/sig_pri.c: Revert stuff not ready for commit in
+ -r289054.
+
+ * channels/sig_pri.c, channels/chan_sip.c: Break up long
+ ast_manager_event_multichan() event lines.
+
+2010-09-27 18:37 +0000 [r288961] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Still build SIP, even if res_crypto cannot
+ be built (use, not depend). (closes issue #18062) Reported by: a
+ user on the mailing list
+
+2010-09-27 13:03 +0000 [r288925-288927] Russell Bryant <russell@digium.com>
+
+ * res/res_agi.c: Fix some documentation typos and spelling errors.
+
+ * res/res_agi.c: Fix a documentation spelling error.
+
+2010-09-24 17:58 +0000 [r288821-288852] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Append Retry-After header on 500 error
+ response to Re-INVITE according to RFC3261 section 14.2. ABE-2301
+
+ * channels/chan_sip.c: Inspect Require header on BYE transaction
+ according to RFC3261 section 8.2.2.3. ABE-2293
+
+2010-09-24 16:02 +0000 [r288748] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 288747 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288747 | twilson | 2010-09-24 08:37:39 -0700
+ (Fri, 24 Sep 2010) | 12 lines Merged revisions 288746 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288746 | twilson | 2010-09-24 08:26:09 -0700 (Fri, 24 Sep 2010)
+ | 5 lines Don't fail a masquerade if it is already being hung up
+ This avoids noise on some Local channel situations where we don't
+ use /n. Thanks to Alec Davis for the suggestion. ........
+ ................
+
+2010-09-24 13:54 +0000 [r288606-288713] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_strings.c: Merged revisions 288712 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r288712 | tilghman | 2010-09-24 08:53:30 -0500 (Fri, 24
+ Sep 2010) | 5 lines Solaris won't printf a NULL. (closes issue
+ #18041) Reported by: asgaroth ........
+
+ * main/asterisk.exports.in: Export timersub for platforms which do
+ not have it
+
+ * include/asterisk/channel.h, cdr/cdr_pgsql.c, /, configure,
+ include/asterisk/autoconfig.h.in, include/asterisk/compat.h,
+ main/strcompat.c, configure.ac: Merged revisions 288637 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288637 | tilghman | 2010-09-23 22:36:01 -0500
+ (Thu, 23 Sep 2010) | 9 lines Merged revisions 288636 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r288636 | tilghman | 2010-09-23 22:20:24 -0500 (Thu, 23
+ Sep 2010) | 2 lines Solaris compatibility fixes ........
+ ................
+
+ * CHANGES: Add note about the checkhangup option of ${CHANNEL()}
+
+2010-09-23 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.0-rc2 Released.
+
+2010-09-23 18:05 +0000 [r288507-288572] Terry Wilson <twilson@digium.com>
+
+ * main/manager.c: Make AMI honor enabled=no (closes issue #18040)
+ Reported by: twilson Review:
+ https://reviewboard.asterisk.org/r/938/
+
+ * channels/chan_local.c, /: Merged revisions 288500 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288500 | twilson | 2010-09-22 16:10:09 -0700
+ (Wed, 22 Sep 2010) | 15 lines Merged revisions 288499 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288499 | twilson | 2010-09-22 16:00:30 -0700 (Wed, 22 Sep 2010)
+ | 8 lines Don't let a Local channel get bridged to itself If a
+ local channel gets bridged to itself, it becomes orphaned with no
+ devices left to actually tell it to hang up. This patch modifies
+ local_fixup() to detect this case and deny it. Review:
+ https://reviewboard.asterisk.org/r/934 ........ ................
+
+2010-09-22 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.0-rc1 Released.
+
+2010-09-22 17:49 +0000 [r288345-288418] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 288417 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288417 | dvossel | 2010-09-22 12:49:05 -0500
+ (Wed, 22 Sep 2010) | 11 lines Merged revisions 288416 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010)
+ | 5 lines RFC3261 section 12.2 explicitly says out of order
+ requests are responded with a 500 Server Internal Error response.
+ ABE-2458 ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 288344 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288344 | dvossel | 2010-09-22 11:53:28 -0500
+ (Wed, 22 Sep 2010) | 9 lines Merged revisions 288343 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22
+ Sep 2010) | 2 lines During check_pendings, if the dialog is
+ terminated with a CANCEL, change the invitestate to INV_CANCEL
+ like in sip_hangup. ........ ................
+
+2010-09-22 16:45 +0000 [r288341] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 288340 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288340 | russell | 2010-09-22 11:44:13 -0500
+ (Wed, 22 Sep 2010) | 18 lines Merged revisions 288339 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288339 | russell | 2010-09-22 11:39:16 -0500 (Wed, 22 Sep 2010)
+ | 11 lines Fix a 100% CPU consumption problem when setting
+ console=yes in asterisk.conf. The handling of -c and console=yes
+ should be the same, but they were not. When you specify -c, it
+ sets both a flag for console module and for asterisk not to
+ fork() off into the background. The handling of console=yes only
+ set console mode, so you would end up with a background process()
+ trying to run the Asterisk console and freaking out since it
+ didn't have anything to read input from. Thanks to beagles for
+ reporting and helping debug the problem! ........
+ ................
+
+2010-09-22 15:14 +0000 [r288268] Tilghman Lesher <tlesher@digium.com>
+
+ * UPGRADE.txt, cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample, /:
+ Merged revisions 288267 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288267 | tilghman | 2010-09-22 10:11:09 -0500
+ (Wed, 22 Sep 2010) | 23 lines Merged revisions 288265-288266 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288265 | tilghman | 2010-09-22 09:48:04 -0500 (Wed, 22 Sep 2010)
+ | 9 lines Allow the encoding to be set, in case local charset
+ does not agree with database. (closes issue #16940) Reported by:
+ jamicque Patches: 20100827__issue16940.diff.txt uploaded by
+ tilghman (license 14) 20100921__issue16940__1.6.2.diff.txt
+ uploaded by tilghman (license 14) Tested by: jamicque ........
+ r288266 | tilghman | 2010-09-22 10:04:52 -0500 (Wed, 22 Sep 2010)
+ | 5 lines Document addition of encoding parameter. (issue #16940)
+ Reported by: jamicque ........ ................
+
+2010-09-22 00:06 +0000 [r288194] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 288193 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288193 | rmudgett | 2010-09-21 19:03:37 -0500
+ (Tue, 21 Sep 2010) | 33 lines Merged revisions 288192 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288192 | rmudgett | 2010-09-21 18:55:58 -0500 (Tue, 21 Sep 2010)
+ | 26 lines In chan_iax2.c:schedule_delivery() calls
+ ast_bridged_channel() on an unlocked channel. Near the beginning
+ of schedule_delivery(), ast_bridged_channel() is called on
+ iaxs[fr->callno]->owner. However, the channel is not locked,
+ which can result in ast_bridged_channel() crashing should
+ owner->tech change to a technology that doesn't implement
+ bridged_channel. I also fixed the other calls to
+ ast_bridged_channel() in chan_iax2.c since the owner lock was not
+ held there either. Converted the existing channel deadlock
+ avoidance to use iax2_lock_owner(). Using the new function
+ simplified some awkward code. In the process of fixing the
+ locking on ast_bridged_channel(), I also found a memory leak in
+ socket_process() for v1.6.2 and v1.8. The local struct variable
+ ies.vars is not freed on early/abnormal function exits. (closes
+ issue #17919) Reported by: rain Patches: issue17919_v1.4.patch
+ uploaded by rmudgett (license 664) issue17919_w_leak_v1.6.2.patch
+ uploaded by rmudgett (license 664) issue17919_w_leak_v1.8.patch
+ uploaded by rmudgett (license 664) Review:
+ https://reviewboard.asterisk.org/r/926/ ........ ................
+
+2010-09-21 22:57 +0000 [r288159] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 288113 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288113 | tilghman | 2010-09-21 16:59:46 -0500
+ (Tue, 21 Sep 2010) | 22 lines Merged revisions 288112 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010)
+ | 15 lines Try both the encoded and unencoded subscription URI
+ for a match in hints. When a phone sends an encoded URI for a
+ subscription, the URI is not matched with the actual hint that is
+ in decoded format. For example, if we have an extension with a
+ hint that is named: "#5601" or "*5601", the subscription will
+ work fine if the phone subscribes with an already decoded URI,
+ but when it's decoded like "%255601" or "%2A5601", Asterisk is
+ unable to match it with the correct hint. (closes issue #17785)
+ Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt
+ uploaded by tilghman (license 14) Tested by: ramonpeek ........
+ ................
+
+2010-09-21 22:26 +0000 [r288157] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 288147 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r288147 | pabelanger | 2010-09-21 18:22:43 -0400 (Tue,
+ 21 Sep 2010) | 9 lines Setup timer before set_config(). (closes
+ issue #18019) Reported by: Netview Patches: issue_0018019.patch
+ uploaded by pabelanger (license 224) Tested by: Netview ........
+
+2010-09-21 21:03 +0000 [r288079-288082] Richard Mudgett <rmudgett@digium.com>
+
+ * doc/tex/partymanip.tex: Add note in party manipulation chapter on
+ interception macros.
+
+ * apps/app_queue.c, apps/app_dial.c: Simplify locking code for
+ REDIRECTING interception macro when forwarding a call. Simplified
+ the locking code by using a local copy of the redirecting party
+ information in app_dial.c:do_forward() and
+ app_queue.c:wait_for_answer() for launching the REDIRECTING
+ interception macro when a call is forwarded. Reduced the lock
+ time of the 'o->chan' and 'in' channels.
+
+ * main/channel.c: Protect channel access in CONNECTED_LINE and
+ REDIRECTING interception macro launch code.
+
+2010-09-21 19:48 +0000 [r288007] Brett Bryant <bbryant@digium.com>
+
+ * main/channel.c, /: Merged revisions 288006 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288006 | bbryant | 2010-09-21 15:46:20 -0400
+ (Tue, 21 Sep 2010) | 14 lines Merged revisions 288005 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288005 | bbryant | 2010-09-21 15:43:46 -0400 (Tue, 21 Sep 2010)
+ | 8 lines Add a check to fix a rare segmentation fault you'd get
+ if ast_frdup couldn't allocate memory on the first frame being
+ queued in ast_queue_frame. (closes issue #17882) Reported by:
+ seanbright Tested by: seanbright ........ ................
+
+2010-09-21 19:08 +0000 [r287935] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 287934 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287934 | tilghman | 2010-09-21 14:07:53 -0500
+ (Tue, 21 Sep 2010) | 9 lines Merged revisions 287933 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r287933 | tilghman | 2010-09-21 14:07:07 -0500 (Tue, 21
+ Sep 2010) | 2 lines Less than zero is an error, not any non-zero
+ value. ........ ................
+
+2010-09-21 19:02 +0000 [r287931] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c: Revert change in favor of a more targeted fix
+
+2010-09-21 18:32 +0000 [r287929] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Send a "415 Unsupported Media Type" after
+ failure to process sdp due to unknown Content-Encoding header.
+ ABE-2258
+
+2010-09-21 15:53 +0000 [r287897] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: Cut-n-paste error in builtin_blindtransfer().
+
+2010-09-21 15:43 +0000 [r287895] Russell Bryant <russell@digium.com>
+
+ * res/res_rtp_asterisk.c, main/dnsmgr.c, channels/chan_sip.c,
+ main/acl.c: Don't use ast_strdupa() from within the arguments to
+ a function. (closes issue #17902) Reported by: afried Patches:
+ issue_17902.rev1.txt uploaded by russell (license 2) Tested by:
+ russell Review: https://reviewboard.asterisk.org/r/927/
+
+2010-09-21 15:24 +0000 [r287893] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Anonymous callerid needs a "sip:" uri
+ prefix. (closes issue #17981) Reported by: avalentin Patches:
+ sip-anonymous-aastra.patch uploaded by avalentin (license 1107)
+ (plus an additional fix by me) Tested by: avalentin
+
+2010-09-21 13:41 +0000 [r287863] Russell Bryant <russell@digium.com>
+
+ * main/logger.c: Fix a regression in verbose logger processing.
+
+2010-09-21 04:37 +0000 [r287833] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c: Don't generate connected line buffer twice for
+ comparison
+
+2010-09-21 00:00 +0000 [r287760] Brett Bryant <bbryant@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 287759 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287759 | bbryant | 2010-09-20 19:58:26 -0400
+ (Mon, 20 Sep 2010) | 23 lines Merged revisions 287758 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010)
+ | 16 lines Fix misvalidation of meetme pins in conjunction with
+ the 'a' MeetMe flag. When using the 'a' MeetMe flag and having a
+ user and admin pin setup for your conference, using the user pin
+ would gain you admin priviledges. Also, when no user pin was set,
+ an admin pin was, the 'a' MeetMe flag wasn't used, and the user
+ tried to enter a conference then they were still prompted for a
+ pin and forced to hit #. (closes issue #17908) Reported by: kuj
+ Patches: pins_2.patch uploaded by kuj (license 1111) Tested by:
+ kuj Review: [full review board URL with trailing slash] ........
+ ................
+
+2010-09-20 23:51 +0000 [r287757] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c: Avoid infinite loop with certain local channel
+ connected line updates Compare connected line data before sending
+ a connected line indication to avoid possible loops. Review:
+ https://reviewboard.asterisk.org/r/932/
+
+2010-09-20 23:20 +0000 [r287701] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/channel.c, /: Merged revisions 287685 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r287685 | alecdavis | 2010-09-21 11:16:45 +1200 (Tue, 21 Sep
+ 2010) | 18 lines ast_channel_masquerade: Avoid recursive
+ masquerades. Check all 4 combinations of (original/clonechan) *
+ (masq/masqr). Initially original->masq and clonechan->masqr were
+ only checked. It's possible with multiple masq's planned - and
+ not yet executed, that the 'original' chan could already have
+ another masq'd into it - thus original->masqr would be set, that
+ masqr would lost. Likewise for the clonechan->masq. (closes issue
+ #16057;#17363) Reported by: amorsen;davidw,alecdavis Patches:
+ based on bug16057.diff4.txt uploaded by alecdavis (license 585)
+ Tested by: ramonpeek, davidw, alecdavis ........
+
+2010-09-20 23:14 +0000 [r287683] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: The inalarm flag was not set in sig_analog
+ struct if the port is initially in alarm. Fixed initial inalarm
+ value for sig_analog ports. Along with -r261007, this gets the
+ inalarm flag in sync with chan_dahdi for sig_analog ports.
+ (closes issue #16983)
+
+2010-09-20 22:21 +0000 [r287661] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/channel.c: ast_do_masquerade. Keep channels ao2_container
+ locked while unlink and linking channels. Previously, Masquerade
+ would unlock 'original' and 'clonechan' and allow another masq
+ thread to run. End result would be corrupted memory, and the
+ frequent report 'Bad Magic Number'. (closes issue #17801,#17710)
+ Reported by: notthematrix Patches: Based on bug17801.diff1.txt
+ uploaded by alecdavis (license 585) Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/928
+
+2010-09-20 22:09 +0000 [r287645-287647] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/channel.h, CHANGES, include/asterisk/framehook.h
+ (added), main/channel.c, main/framehook.c (added),
+ funcs/func_frame_trace.c (added): Addition of the FrameHook API
+ (AKA AwesomeHooks) So far all our tools for viewing and
+ manipulating media streams within Asterisk have been entirely
+ focused on audio. That made sense then, but is not scalable now.
+ The FrameHook API lets us tap into and manipulate _ANY_ type of
+ media or signaling passed on a channel present today or in the
+ future. This tool is a step in the direction of expanding
+ Asterisk's boundaries and will help generate some rather
+ interesting applications in the future. In addition to the
+ FrameHook API, a simple dialplan function exercising the api has
+ been included as well. This function is called FRAME_TRACE().
+ FRAME_TRACE() allows for the internal ast_frames read and written
+ to a channel to be output. Filters can be placed on this function
+ to debug only certain types of frames. This function could be
+ thought of as an internal way of doing ast_frame packet captures.
+ Review: https://reviewboard.asterisk.org/r/925/
+
+ * channels/chan_sip.c: Fixes issue with registrations not working
+ properly with pedantic=yes. (closes issue #18017) Reported by:
+ schmidts Patches: issues_18017_v1.diff uploaded by dvossel
+ (license 671) Tested by: schmidts
+
+2010-09-20 21:29 +0000 [r287643] Jason Parker <jparker@digium.com>
+
+ * /, channels/chan_skinny.c: Merged revisions 287642 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r287642 | qwell | 2010-09-20 16:28:32 -0500 (Mon, 20 Sep
+ 2010) | 8 lines Don't crash when parking a non-bridged call.
+ (closes issue #17680) Reported by: jmhunter Patches:
+ chan_skinny-park-v1.txt uploaded by DEA (license 3) Tested by:
+ jmhunter, DEA ........
+
+2010-09-20 21:19 +0000 [r287639] Brett Bryant <bbryant@digium.com>
+
+ * main/logger.c: Fixes an error with the logger that caused verbose
+ messages to be spammed to the screen if syslog was configured in
+ logger.conf (closes issue #17974) Reported by: lmadsen Review:
+ https://reviewboard.asterisk.org/r/915/
+
+2010-09-20 15:57 +0000 [r287559] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 287558 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287558 | mnicholson | 2010-09-20 10:56:21 -0500
+ (Mon, 20 Sep 2010) | 14 lines Use ast_str when processing hint
+ state changes Merged revisions 287555 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep
+ 2010) | 5 lines Use ast_dynamic_str when processing hint state
+ changes (related to issue #17928) Reported by: mdu113 ........
+ ................
+
+2010-09-19 16:09 +0000 [r287471] Olle Johansson <oej@edvina.net>
+
+ * main/manager.c, /: Merged revisions 287470 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287470 | oej | 2010-09-19 18:06:10 +0200 (Sön,
+ 19 Sep 2010) | 14 lines Merged revisions 287469 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287469 | oej | 2010-09-19 17:56:50 +0200 (Sön, 19 Sep 2010) | 7
+ lines Make sure we always free variables properly in manager
+ originate. (closes issue #17891) reported, solved and tested by
+ oej Review: https://reviewboard.asterisk.org/r/869/ ........
+ ................
+
+2010-09-17 21:08 +0000 [r287388] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 287387 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287387 | tilghman | 2010-09-17 16:08:00 -0500
+ (Fri, 17 Sep 2010) | 14 lines Merged revisions 287386 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010)
+ | 7 lines Blank columns should get set on reload, not ignored.
+ (closes issue #16893) Reported by: haakon Patches:
+ 20100818__issue16893.diff.txt uploaded by tilghman (license 14)
+ ........ ................
+
+2010-09-17 13:37 +0000 [r287309] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 287308 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287308 | mnicholson | 2010-09-17 08:36:07 -0500
+ (Fri, 17 Sep 2010) | 12 lines Merged revisions 287307 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287307 | mnicholson | 2010-09-17 08:34:34 -0500 (Fri, 17 Sep
+ 2010) | 5 lines Use ast_strdup() instead of ast_strdupa() while
+ processing in ast_hint_state_changed(). (related to issue #17928)
+ Reported by: mdu113 ........ ................
+
+2010-09-17 08:44 +0000 [r287269-287271] Jan Kalab <pitlicek@gmail.com>
+
+ * res/res_calendar_ews.c: Events are visible after they were
+ removed from EWS calendar Because we must merge calendar even
+ when it's empty. (closes issue #17786)
+
+ * res/res_calendar_ews.c: Asterisk crashing because of double free
+ when EWS request fails The free is done later in code. I think
+ ast_free() should have built in checks for double free. (closes
+ issue #17782)
+
+ * res/res_calendar_caldav.c, res/res_calendar_ews.c,
+ res/res_calendar_exchange.c, res/res_calendar_icalendar.c:
+ Support for HTTP redirects in calendar's URL libneon does not
+ support HTTP redirects (3xx responses) by default. You must tell
+ it to follow them. Also, another little unsigned int fix. (closes
+ issue #17776) Review: https://reviewboard.asterisk.org/r/921/
+
+2010-09-16 22:04 +0000 [r287195] Jason Parker <jparker@digium.com>
+
+ * contrib/init.d/rc.debian.asterisk: Don't fail when running the
+ Debian init script directly (as one would normally do). readlink
+ apparently returns 1 when the arg isn't a symlink, which caused
+ the script to exit. (closes issue #17910) Reported by: wurstsalat
+
+2010-09-16 21:57 +0000 [r287193] Russell Bryant <russell@digium.com>
+
+ * UPGRADE.txt, apps/app_queue.c, configs/queues.conf.sample: Set
+ the default for "autofill" and "shared_lastcall" to "yes" in
+ queues.conf. Review: https://reviewboard.asterisk.org/r/922/
+
+2010-09-16 20:07 +0000 [r287116-287120] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 287119 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287119 | mnicholson | 2010-09-16 15:06:16 -0500
+ (Thu, 16 Sep 2010) | 15 lines Merged revisions 287118 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287118 | mnicholson | 2010-09-16 15:04:46 -0500 (Thu, 16 Sep
+ 2010) | 8 lines Don't limit hint processing in
+ ast_hint_state_changed() to AST_MAX_EXTENSION length strings.
+ (closes issue #17928) Reported by: mdu113 Patches:
+ 20100831__issue17928.diff.txt uploaded by tilghman (license 14)
+ Tested by: mdu113 ........ ................
+
+ * main/cdr.c, /: Merged revisions 287115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287115 | mnicholson | 2010-09-16 14:53:41 -0500
+ (Thu, 16 Sep 2010) | 15 lines Merged revisions 287114 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287114 | mnicholson | 2010-09-16 14:52:39 -0500 (Thu, 16 Sep
+ 2010) | 8 lines Don't stop printing cdr variables if we encounter
+ one with a blank name or value. (closes issue #17900) Reported
+ by: under Patches: core-show-channel-cdr-fix1.diff uploaded by
+ mnicholson (license 96) Tested by: mnicholson ........
+ ................
+
+2010-09-15 22:17 +0000 [r287056] Terry Wilson <twilson@digium.com>
+
+ * res/res_srtp.c: Don't hang up a call on an SRTP unprotect failure
+ Also make it more obvious when there is an issue en/decrypting.
+ (closes issue #17563) Reported by: Alexcr Patches:
+ res_srtp.c.patch uploaded by sfritsch (license 1089) Tested by:
+ twilson
+
+2010-09-15 20:58 +0000 [r287020] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c: fix uninintialized variable
+
+2010-09-15 20:53 +0000 [r287017] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_msg_parser.c, channels/chan_misdn.c: Merged
+ revision 287014 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed,
+ 15 Sep 2010) | 58 lines The handling of call transfer signaling
+ for mISDN PTMP is not fully implemented. The handling of call
+ transfer signaling for mISDN PTMP is not fully implemented. The
+ signaling of number updates with ISDN/DSS1 ECT supplementary
+ services (ETS 300 369-1) comes along with a notification
+ indicator IE and redirection number IE for PTMP. The
+ implementation in the current Asterisk mISDN channel
+ unfortunately can handle these information elements only in a
+ NOTIFY message. These information elements are also signaled in a
+ FACILTY message with a RequestSubaddress facility, when the
+ subscriber is already in the active state (see 9.2.4 and 9.2.5 of
+ ETS 300 369-1). ********** abe_2526_ast.patch * Added support to
+ handle the notification indicator IE and redirection number IE
+ with the RequestSubaddress facility. * Made
+ misdn_update_connected_line() send a NOTIFY message if Asterisk
+ originated the call and it is not connected yet. * Made
+ misdn_update_connected_line() send a FACILITY message if the call
+ is already connected. This patch requires the presence of the
+ associated mISDN patches to compile. I had to enhance mISDN to
+ allow the notification indicator IE and the redirection number IE
+ to be used with a FACILITY message. Earlier versions of the
+ Digium enhanced mISDN are no longer going to work. **********
+ abe_2526_misdn.patch * Made an incoming FACILITY message allow
+ the presence of the notification indicator IE and the redirection
+ number IE. ********** abe_2526_misdnuser_v3.patch * Added support
+ to send and receive a FACILITY message with the notification
+ indicator IE and the redirection number IE. * Added the ability
+ to send a NOTIFY message in PTMP/NT mode to all responding
+ subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches:
+ abe_2526_ast.patch uploaded by rmudgett (license 664)
+ abe_2526_misdn.patch uploaded by rmudgett (license 664)
+ abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664)
+ Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526
+ ..........
+
+2010-09-15 20:32 +0000 [r286931-287015] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 286998 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286998 | jpeeler | 2010-09-15 15:28:02 -0500
+ (Wed, 15 Sep 2010) | 14 lines Merged revisions 286941 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15 Sep 2010)
+ | 7 lines Ensure mailbox is not filled to capacity before doing
+ message forwarding. Specifically, before prompting to record a
+ prepended message the capacity is checked first. If the mailbox
+ is full the extension will be reprompted. ABE-2517 ........
+ ................
+
+ * CHANGES, channels/chan_iax2.c, channels/sip/include/sip.h,
+ configs/features.conf.sample, channels/chan_mgcp.c,
+ include/asterisk/features.h, channels/chan_dahdi.c,
+ channels/sig_analog.c, channels/chan_sip.c, main/features.c: Add
+ parking extension for non-default parking lots. This is a new
+ feature that allows for parking to custom parking lots to be
+ accessed directly, rather than with channel variables or by
+ changing the default parking lot. The extension is set with the
+ parkext option just as the default parking lot is done. Also, the
+ manager action has been updated to optionally allow a specified
+ parking lot. (closes issue #14882) Reported by: vmikhnevych
+ Patches: patch_14882.txt uploaded by mnick (license 874) modified
+ by me Review: https://reviewboard.asterisk.org/r/884/
+
+2010-09-15 18:29 +0000 [r286904-286905] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_analog.c: Simplify some code in sig_analog.
+
+ * channels/sig_analog.c: Unable to originate calls using E&M over
+ T1. When originating a call from Unit Under Test to Reference
+ Unit using E&M RBS signaling mode, I get the following warning
+ message: "Ring/Off-hook in strange state 3 on channel 1". Fixed
+ the sig_analog outgoing flag. It was never set when sig_analog
+ was extracted from chan_dahdi. JIRA SWP-2191 JIRA AST-408
+
+2010-09-15 13:05 +0000 [r286868] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Set tohost to the domain specified in the
+ configuration file instead of the IP address of the host we are
+ calling. This fixes a regression introduced in r274783. (closes
+ issue #17960) Reported by: adriavidal Patches:
+ sip-tohost-fix1.diff uploaded by mnicholson (license 96) Tested
+ by: mich, mnicholson, adriavidal (closes issue #17676) Reported
+ by: outcast Patches: sip-tohost-fix1.diff uploaded by mnicholson
+ (license 96) Tested by: mnicholson
+
+2010-09-14 21:57 +0000 [r286834] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Sets subscribed type for outgoing MWI
+ subscriptions so correct Event header is used.
+
+2010-09-14 19:28 +0000 [r286682-286758] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 286757 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286757 | mnicholson | 2010-09-14 14:27:28 -0500
+ (Tue, 14 Sep 2010) | 20 lines Merged revisions 286756 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep
+ 2010) | 13 lines Don't clear the username from a realtime
+ database when a registration expires. Non-realtime chan_sip does
+ not clear the username from memory when a registration expiries
+ so realtime probably shouldn't either. (closes issue #17551)
+ Reported by: ricardolandim Patches:
+ reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license
+ 96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson
+ (license 96) reg-expiry-username-1.8-fix1.diff uploaded by
+ mnicholson (license 96) reg-expiry-username-trunk-fix1.diff
+ uploaded by mnicholson (license 96) Tested by: ricardolandim,
+ mnicholson ........ ................
+
+ * main/channel.c, /: Merged revisions 286681 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286681 | mnicholson | 2010-09-14 13:02:24 -0500
+ (Tue, 14 Sep 2010) | 14 lines Merged revisions 286679 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep
+ 2010) | 7 lines Only drop duplicate answer frames if the channel
+ is bridged. Back in r3710 ast_read() was modified to drop answer
+ frames on channels that were in the UP state. This modification
+ prevented bridges that were up before the answer from being
+ broken and reestablished by an ANSWER control frame. That change
+ also prevents pickup of channels called from the ast_dial
+ framework from working properly. The ast_dial framework expects
+ to see an ANSWER frame after dialing and the pickup code queues
+ one but ast_read() drops it. This new change only drops ANSWER
+ frames when the channel is bridged, allowing the answer queued by
+ the pickup code to properly pass through ast_read() on to the
+ ast_dial framework. ABE-2473 (related to issue #2342) ........
+ ................
+
+2010-09-14 15:30 +0000 [r286647] Richard Mudgett <rmudgett@digium.com>
+
+ * doc/tex/channelvariables.tex, doc/tex/partymanip.tex: Corrected
+ documented CONNECTED_LINE and REDIRECTING party manipulation
+ macro names.
+
+2010-09-14 06:55 +0000 [r286617] Jan Kalab <pitlicek@gmail.com>
+
+ * res/res_calendar_ews.c: Merging events for Exchange web service
+ doesn't work as expected, resulting in only one event in calendar
+ The solution is to use "global" counter of events, since we do
+ new requests for every event and calendar sync after every
+ request. So now we do sync only after last request. (closes issue
+ #17877) Review: https://reviewboard.asterisk.org/r/916/
+
+2010-09-14 05:07 +0000 [r286528-286588] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/realtime/mysql/voicemail_data.sql (added), /,
+ contrib/realtime/mysql/voicemail_messages.sql (added): Merged
+ revisions 286587 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r286587 | tilghman | 2010-09-14 00:06:05 -0500 (Tue, 14 Sep 2010)
+ | 2 lines Add documentation on missing backend tables for
+ Voicemail ........
+
+ * /, main/features.c: Merged revisions 286557 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r286557 | tilghman | 2010-09-13 18:48:51 -0500 (Mon, 13 Sep 2010)
+ | 2 lines C precedence got me ........
+
+ * /, main/features.c: Merged revisions 286527 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r286527 | tilghman | 2010-09-13 18:03:26 -0500 (Mon, 13 Sep 2010)
+ | 2 lines Refactor conversion to ast_poll() to fix callparking
+ regression. ........
+
+2010-09-13 19:40 +0000 [r286457] Jason Parker <jparker@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 286456 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r286456 | qwell | 2010-09-13 14:38:35 -0500 (Mon, 13 Sep 2010) |
+ 5 lines Remove "Internal IP" from sip show settings, as it's not
+ at all useful to display. (closes issue #17840) Reported by: oej
+ ........
+
+2010-09-13 15:52 +0000 [r286426] Richard Mudgett <rmudgett@digium.com>
+
+ * configs/chan_dahdi.conf.sample: Update chan_dahdi.conf.sample to
+ reflect new libpri T309 default value.
+
+2010-09-11 17:09 +0000 [r286270] Olle Johansson <oej@edvina.net>
+
+ * /, main/file.c: Merged revisions 286268 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286268 | oej | 2010-09-11 19:05:16 +0200 (Lör,
+ 11 Sep 2010) | 11 lines Merged revisions 286267 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4
+ lines Handle error response when we can't make file compatible
+ Review: https://reviewboard.asterisk.org/r/911/ ........
+ ................
+
+2010-09-10 22:04 +0000 [r286189] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/channel.h, include/asterisk/pbx.h,
+ include/asterisk/frame.h, channels/chan_local.c,
+ funcs/func_channel.c: Merged revisions 286115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286115 | twilson | 2010-09-10 15:35:25 -0500
+ (Fri, 10 Sep 2010) | 23 lines Merged revisions 286059 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010)
+ | 16 lines Inherit CHANNEL() writes to both sides of a Local
+ channel Having Local (/n) channels as queue members and setting
+ the language in the extension with Set(CHANNEL(language)=fr) sets
+ the language on the Local/...,2 channel. Hold time report
+ playbacks happen on the Local/...,1 channel and therefor do not
+ play in the specified language. This patch modifies
+ func_channel_write to call the setoption callback and pass the
+ CHANNEL() write info to the callback. chan_local uses this
+ information to look up the other side of the channel and apply
+ the same changes to it. (closes issue #17673) Reported by:
+ Guggemand Review: https://reviewboard.asterisk.org/r/903/
+ ........ ................
+
+2010-09-10 21:11 +0000 [r286120] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 286117 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286117 | pabelanger | 2010-09-10 16:55:06 -0400
+ (Fri, 10 Sep 2010) | 11 lines Merged revisions 286114 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286114 | pabelanger | 2010-09-10 16:35:08 -0400 (Fri, 10 Sep
+ 2010) | 4 lines Load iax.conf before registering any
+ functions/applications/actions. Review:
+ https://reviewboard.asterisk.org/r/914/ ........ ................
+
+2010-09-10 20:55 +0000 [r286118] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c, /: Merged revisions 286116 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286116 | rmudgett | 2010-09-10 15:42:44 -0500
+ (Fri, 10 Sep 2010) | 18 lines Merged revisions 286113 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010)
+ | 11 lines An outgoing call may not get hung up if a pre-connect
+ incoming ISDN call is disconnected. If the ISDN link a
+ pre-connect incoming call is using fails or is reset, the
+ outgoing leg may not hang up or be delayed in hanging up.
+ (Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER,
+ PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and
+ PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the
+ incoming call leg hangs up before connecting for any reason. It
+ makes no sense to send a BUSY or CONGESTION control frame to the
+ outgoing call leg under these circumstances. ........
+ ................
+
+2010-09-10 20:31 +0000 [r286112] Russell Bryant <russell@digium.com>
+
+ * main/db.c: Rate limit calls to fsync() to 1 per second after
+ astdb updates. Astdb was determined to be one of the most
+ significant bottlenecks in SIP registration processing. This
+ patch improved the speed of an astdb load test by 50000% (yes,
+ Fifty-Thousand Percent). On this particular load test setup, this
+ doubled the number of SIP registrations the server could handle.
+ Review: https://reviewboard.asterisk.org/r/825/
+
+2010-09-10 18:31 +0000 [r286025] Tilghman Lesher <tlesher@digium.com>
+
+ * /: Merged revisions 286024 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286024 | tilghman | 2010-09-10 13:30:21 -0500
+ (Fri, 10 Sep 2010) | 9 lines Merged revisions 286023 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r286023 | tilghman | 2010-09-10 13:22:04 -0500 (Fri, 10
+ Sep 2010) | 2 lines Missing newline ........ ................
+
+2010-09-10 13:13 +0000 [r285992] David Ruggles <thedavidfactor@gmail.com>
+
+ * doc/externalivr.txt, CHANGES: Added missing documentation for
+ ExternalIVR feature added in January 2010
+
+2010-09-10 05:32 +0000 [r285931-285962] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/select.h, /: Merged revisions 285961 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r285961 | tilghman | 2010-09-10 00:31:31 -0500 (Fri, 10 Sep 2010)
+ | 6 lines Another fix for Mac OS X. While trying to fix this the
+ "right" way, I wandered into dependency hell. Two hours later, I
+ backed out, and just removed the offending code. ast_inline_api
+ only goes one level deep and then it breaks. Ouch. ........
+
+ * tests/test_poll.c, include/asterisk/select.h, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 285930 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285930 | tilghman | 2010-09-09 20:16:32 -0500
+ (Thu, 09 Sep 2010) | 14 lines Merged revisions 285889 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010)
+ | 7 lines Fix Mac OS X build. This also fixes a rather grievous
+ calculation error for the offset of ast_fdset, which was masked
+ on Linux and FreeBSD, because these platforms check the first 256
+ FDs regardless of the bitmask setting (due to backwards
+ compatibility). ........ ................
+
+2010-09-09 22:52 +0000 [r285819] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, codecs/gsm/Makefile: Merged revisions 285818 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285818 | pabelanger | 2010-09-09 18:49:19 -0400
+ (Thu, 09 Sep 2010) | 15 lines Merged revisions 285817 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285817 | pabelanger | 2010-09-09 18:34:35 -0400 (Thu, 09 Sep
+ 2010) | 8 lines GCC 4.2.x optimizations result in improper
+ behavior of GSM codec (closes issue #17688) Reported by:
+ pprindeville Patches: asterisk-trunk-bugid11243.patch uploaded by
+ pprindeville (license 347) Tested by: mkeuter, pprindeville
+ ........ ................
+
+2010-09-09 20:11 +0000 [r285745] Jason Parker <jparker@digium.com>
+
+ * main/channel.c, /: Merged revisions 285744 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285744 | qwell | 2010-09-09 15:09:23 -0500
+ (Thu, 09 Sep 2010) | 16 lines Merged revisions 285742 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) |
+ 9 lines Transmit silence when reading DTMF in ast_readstring.
+ Otherwise, you could get issues with DTMF timeouts causing
+ hangups. (closes issue #17370) Reported by: makoto Patches:
+ channel-readstring-silence-generator.patch uploaded by makoto
+ (license 38) ........ ................
+
+2010-09-09 18:51 +0000 [r285640-285711] Brett Bryant <bbryant@digium.com>
+
+ * main/pbx.c, /: Merged revisions 285710 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010)
+ | 8 lines Fixes an issue with dialplan pattern matching where the
+ specificity for pattern ranges and pattern special characters was
+ inconsistent. (closes issue #16903) Reported by: Nick_Lewis
+ Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license
+ 657) Tested by: Nick_Lewis ........
+
+ * res/res_musiconhold.c, /: Merged revisions 285639 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285639 | bbryant | 2010-09-09 13:22:25 -0400
+ (Thu, 09 Sep 2010) | 14 lines Merged revisions 285638 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285638 | bbryant | 2010-09-09 13:20:17 -0400 (Thu, 09 Sep 2010)
+ | 7 lines Fixes an issue with MOH where it doesn't recover
+ cleanly when it can't play a file and would just stop, instead of
+ continuing to find the next playable file in the MOH class.
+ (closes issue #17807) Reported by: kshumard Review:
+ https://reviewboard.asterisk.org/r/910/ ........ ................
+
+2010-09-08 22:14 +0000 [r285564-285568] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 285567 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285567 | dvossel | 2010-09-08 17:11:28 -0500
+ (Wed, 08 Sep 2010) | 9 lines Merged revisions 285566 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r285566 | dvossel | 2010-09-08 17:07:31 -0500 (Wed, 08
+ Sep 2010) | 2 lines In retrans_pkt, do not unlock pvt until the
+ end of the function on a transmit failure. ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 285563 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r285563 | dvossel | 2010-09-08 16:47:29 -0500 (Wed, 08 Sep 2010)
+ | 54 lines Fixes interoperability problems with session timer
+ behavior in Asterisk. CHANGES: 1. Never put "timer" in "Require"
+ header. This is not to our benefit and RFC 4028 section 7.1 even
+ warns against it. It is possible for one endpoint to perform
+ session-timer refreshes while the other endpoint does not support
+ them. If in this case the end point performing the refreshing
+ puts "timer" in the Require field during a refresh, the dialog
+ will likely get terminated by the other end. 2. Change the
+ behavior of 'session-timer=accept' in sip.conf (which is the
+ default behavior of Asterisk with no session timer configuration
+ specified) to only run session-timers as result of an incoming
+ INVITE request if the INVITE contains an "Session-Expires"
+ header... Asterisk is currently treating having the "timer"
+ option in the "Supported" header as a request for session timers
+ by the UAC. I do not agree with this. Session timers should only
+ be negotiated in "accept" mode when the incoming INVITE supplies
+ a "Session-Expires" header, otherwise RFC 4028 says we should
+ treat a request containing no "Session-Expires" header as a
+ session with no expiration. Below I have outlined some situations
+ and what Asterisk's behavior is. The table reflects the behavior
+ changes implemented by this patch. SITUATIONS: -Asterisk as UAS
+ 1. Incoming INVITE: NO "Session-Expires" 2. Incoming INVITE: HAS
+ "Session-Expires" -Asterisk as UAC 3. Outgoing INVITE: NO
+ "Session-Expires". 200 Ok Response HAS "Session-Expires" header
+ 4. Outgoing INVITE: NO "Session-Expires". 200 Ok Response NO
+ "Session-Expires" header 5. Outgoing INVITE: HAS
+ "Session-Expires". Active - Asterisk will have an active refresh
+ timer regardless if the other endpoint does. Inactive - Asterisk
+ does not have an active refresh timer regardless if the other
+ endpoint does. XXXXXXX - Not possible for mode.
+ ______________________________________ |SITUATIONS |
+ 'session-timer' MODES | |___________|________________________| |
+ | originate | accept | |-----------|------------|-----------| |1.
+ | Active | Inactive | |2. | Active | Active | |3. | XXXXXXXX |
+ Active | |4. | XXXXXXXX | Inactive | |5. | Active | XXXXXXXX |
+ -------------------------------------- (closes issue #17005)
+ Reported by: alexrecarey ........
+
+2010-09-08 20:58 +0000 [r285533] Brett Bryant <bbryant@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 285532 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r285532 | bbryant | 2010-09-08 16:56:12 -0400 (Wed, 08 Sep 2010)
+ | 8 lines Fixes a bug with MeetMe where after announcing the
+ amount of time left in a conference, if music on hold was
+ playing, it doesn't restart. (closes issue #17408) Reported by:
+ sysreq Patches: asterisk-issue-17408_fixed.patch uploaded by
+ sysreq (license 1009) Tested by: sysreq ........
+
+2010-09-08 20:43 +0000 [r285527-285530] Jason Parker <jparker@digium.com>
+
+ * res/res_musiconhold.c, /, include/asterisk/astobj2.h: Merged
+ revisions 285529 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r285529 | qwell | 2010-09-08 15:42:44 -0500 (Wed, 08 Sep 2010) |
+ 1 line Follow coding guidelines in moh rescan fix. Also fix the
+ documentation that got me in trouble. ........
+
+ * res/res_musiconhold.c, /: Merged revisions 285526 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r285526 | qwell | 2010-09-08 15:31:43 -0500 (Wed, 08 Sep
+ 2010) | 8 lines Fixes issue where moh files were no longer
+ rescanned during a reload. (closes issue #16744) Reported by: pj
+ Patches: 16744-reload.diff uploaded by qwell (license 4) Tested
+ by: qwell ........
+
+2010-09-08 07:14 +0000 [r285484] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_channel.c: Documentation only
+
+2010-09-07 22:22 +0000 [r285455] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c: Don't automatically add domains for wildcard
+ bindaddrs. (closes issue #17832) Reported by: oej Patches:
+ 17832-wildcard.diff uploaded by qwell (license 4) Tested by:
+ qwell
+
+2010-09-07 21:20 +0000 [r285373-285386] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_spool.c: Don't notify on attribute changes, and change
+ how the queuing mechanism works. Fixes call spools in 1.8.
+ (closes issue #17337) Reported by: loloski Patches:
+ 20100827__issue17337.diff.txt uploaded by tilghman (license 14)
+ (closes issue #17924) Reported by: mkeuter Tested by: mkeuter
+
+ * funcs/func_channel.c: Add CHANNEL(checkhangup) to check whether a
+ channel is in the process of being hanged up. (closes issue
+ #17652) Reported by: kobaz Patches: func_channel.patch uploaded
+ by kobaz (license 834)
+
+2010-09-07 21:08 +0000 [r285371] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: Fix cut-n-paste error.
+
+2010-09-07 20:58 +0000 [r285369] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c: Add note to 'sip show settings' regarding
+ dual-stack support, and a :: bindaddress. (closes issue #17831)
+ Reported by: oej Patches: 17831-v6wildcardbind.diff uploaded by
+ qwell (license 4)
+
+2010-09-07 20:56 +0000 [r285268-285367] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 285366 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285366 | tilghman | 2010-09-07 15:31:41 -0500
+ (Tue, 07 Sep 2010) | 16 lines Merged revisions 285365 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285365 | tilghman | 2010-09-07 15:30:22 -0500 (Tue, 07 Sep 2010)
+ | 9 lines Catch invalid extensions at the parser, instead of
+ making the core deal with them. (closes issue #17794) Reported
+ by: PavelL Patches: 20100820__issue17794__1.6.2.diff.txt uploaded
+ by tilghman (license 14) 20100820__issue17794__1.4.diff.txt
+ uploaded by tilghman (license 14) Tested by: PavelL ........
+ ................
+
+ * include/asterisk/compiler.h, addons/ooh323c/src/ooSocket.h: Fix
+ build on FreeBSD 8.0, take 2.
+
+ * main/poll.c, /: Merged revisions 285267 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285267 | tilghman | 2010-09-07 14:07:17 -0500
+ (Tue, 07 Sep 2010) | 11 lines Merged revisions 285266 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285266 | tilghman | 2010-09-07 14:04:50 -0500 (Tue, 07 Sep 2010)
+ | 4 lines Use poll, if indicated to do so, in the ast_poll2
+ implementation. This fixes the unit tests on FreeBSD 8.0.
+ ........ ................
+
+2010-09-07 17:54 +0000 [r285197] Brett Bryant <bbryant@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 285196 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285196 | bbryant | 2010-09-07 13:49:07 -0400
+ (Tue, 07 Sep 2010) | 17 lines Merged revisions 285194 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285194 | bbryant | 2010-09-07 13:45:41 -0400 (Tue, 07 Sep 2010)
+ | 10 lines Fixes voicemail.conf issues where mailboxes with
+ passwords that don't precede a comma would throw unnecessary
+ error messages. (closes issue #15726) Reported by: 298 Patches:
+ M15726.diff uploaded by junky (license 177) Tested by: junky
+ Review: [full review board URL with trailing slash] ........
+ ................
+
+2010-09-07 17:47 +0000 [r285195] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: Merged revisions 285193 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 285192 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3 ........
+ r285192 | rmudgett | 2010-09-07 11:58:57 -0500 (Tue, 07 Sep 2010)
+ | 8 lines COLP/CONP and chan_misdn missing update chan_misdn does
+ not update the caller id of the channel if a new connected number
+ or ECT-INFORM (w/ new peer number on call transfer) is received.
+ JIRA ABE-2502 JIRA SWP-2058 ........ ........
+
+2010-09-06 20:10 +0000 [r285161-285162] Russell Bryant <russell@digium.com>
+
+ * configure: regenerate configure script.
+
+ * include/asterisk/autoconfig.h.in, configure.ac: Fix libsrtp -fPIC
+ check for when non-standard prefix is used. Thanks to loompek in
+ #asterisk for reporting the issue and testing this patch.
+
+2010-09-06 06:56 +0000 [r285090] Tilghman Lesher <tlesher@digium.com>
+
+ * BSDmakefile (added), makeopts.in, /: Merged revisions 285089 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r285089 | tilghman | 2010-09-06 01:55:17 -0500
+ (Mon, 06 Sep 2010) | 9 lines Merged revisions 285088 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r285088 | tilghman | 2010-09-06 01:54:18 -0500 (Mon, 06
+ Sep 2010) | 2 lines Silly convenience script for BSD platforms.
+ ........ ................
+
+2010-09-04 18:08 +0000 [r285057] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/cli.h: Add a C++ compatible version of
+ AST_CLI_DEFINE().
+
+2010-09-03 23:19 +0000 [r285017] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Call correct lock function as transferer is
+ a sip_pvt not a channel Both functions are #defined to ao2_lock,
+ but still...
+
+2010-09-03 22:21 +0000 [r285006] David Vossel <dvossel@digium.com>
+
+ * configs/sip.conf.sample, channels/sip/include/sip.h,
+ channels/chan_sip.c: Disables auth_options_request option by
+ default. The auth_options_request option was created to do
+ authentication on OPTIONS request just like INVITES are done.
+ Since it has been noted that some endpoints use OPTIONS requests
+ as a way of qualifying a peer and that a 401 authentication
+ response could result in interoperability issues, this option has
+ been disabled by default.
+
+2010-09-03 18:19 +0000 [r284967] Brett Bryant <bbryant@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 284958 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r284958 | bbryant | 2010-09-03 14:15:49 -0400 (Fri, 03
+ Sep 2010) | 8 lines This is a patch provided for issue #17935 to
+ add the ActionID to the IAXregistry AMI response. (closes issue
+ #17935) Reported by: alexkuklin Patches: iaxshowreg uploaded by
+ alexkuklin (license 1115) Tested by: alexkuklin ........
+
+2010-09-03 18:03 +0000 [r284950-284952] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: During OPTIONS authentication, the authpeer
+ does not need to be returned for any reason.
+
+ * configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h,
+ channels/chan_sip.c: authenticate OPTIONS requests just like we
+ would an INVITE OPTIONS requests should be treated the same as an
+ INVITE This includes authentication. This patch adds the ability
+ for incoming out of dialog OPTION requests to be authenticated
+ before providing a response indicating whether an extension is
+ available or not. The authentication routine works the exact same
+ way as it does for incoming INVITEs. This means that if a peer
+ has 'insecure=invite' in their peer definition, the same will be
+ true for the processing of the OPTIONS request. Review:
+ https://reviewboard.asterisk.org/r/881/
+
+2010-09-03 16:28 +0000 [r284921] Terry Wilson <twilson@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 284897 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284897 | twilson | 2010-09-03 11:20:45 -0500
+ (Fri, 03 Sep 2010) | 12 lines Merged revisions 284881 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010)
+ | 5 lines Properly detect when a sound file doesn't exist
+ ast_fileexists returns -1 for error and 0 for a non-existant
+ file. The existing code treated missing files as though they
+ existed. ........ ................
+
+2010-09-03 13:07 +0000 [r284849-284852] Jan Kalab <pitlicek@gmail.com>
+
+ * res/res_calendar_ews.c: Calendar categories and priorities:
+ strdupa() fix
+
+ * res/res_calendar_ews.c: Fix for calendar categories and
+ priorities according to ISO C90
+
+ * res/res_calendar_caldav.c, include/asterisk/calendar.h,
+ res/res_calendar_ews.c, res/res_calendar.c,
+ res/res_calendar_icalendar.c: Support for calendar events
+ priorities and categories Review 880
+
+2010-09-02 21:04 +0000 [r284781] Brett Bryant <bbryant@digium.com>
+
+ * main/manager.c, /: Merged revisions 284778 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284778 | bbryant | 2010-09-02 16:54:33 -0400
+ (Thu, 02 Sep 2010) | 14 lines Merged revisions 284777 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284777 | bbryant | 2010-09-02 16:25:03 -0400 (Thu, 02 Sep 2010)
+ | 7 lines Fixes a bug in manager.c where the default
+ configuration values weren't reset when the manager configuration
+ was reloaded. (closes issue #17917) Reported by: lmadsen Review:
+ https://reviewboard.asterisk.org/r/883/ ........ ................
+
+2010-09-02 21:02 +0000 [r284779-284780] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Simplified pri_dchannel() poll timeout
+ duration code.
+
+ * channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c:
+ Made output libpri event names if pri debugging is enabled when
+ sig_pri processes them. * Simplified CLI "pri debug xx span xx"
+ command code and removed redundant debugging enabled messages. *
+ Made CLI "pri debug xx span xx" command only close the debugging
+ log file if it was opened.
+
+2010-09-02 16:56 +0000 [r284705] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 284704 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284704 | dvossel | 2010-09-02 11:48:51 -0500
+ (Thu, 02 Sep 2010) | 13 lines Merged revisions 284703 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284703 | dvossel | 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010)
+ | 7 lines Removed relatedpeer code from sip_autodestruct Handling
+ of the relatedpeer structure associated with a sip_pvt should be
+ done during the final sip_destruction function, not in
+ sip_autodestruct. ........ ................
+
+2010-09-02 16:43 +0000 [r284701] Jason Parker <jparker@digium.com>
+
+ * formats/format_wav.c: Add slin16 support for format_wav (new
+ wav16 file extension) (closes issue #15029) Reported by: andrew
+ Patches: wav16.patch uploaded by andrew (license 240) Tested by:
+ qwell, andrew
+
+2010-09-02 16:34 +0000 [r284698] Richard Mudgett <rmudgett@digium.com>
+
+ * doc/tex/channelvariables.tex, doc/tex/partymanip.tex (added),
+ doc/tex/asterisk.tex: Added documentation for CONNECTEDLINE and
+ REDIRECTING functions. (closes issue #17808) Reported by: jtodd
+ Review: https://reviewboard.asterisk.org/r/875/
+
+2010-09-02 16:27 +0000 [r284597-284696] Tilghman Lesher <tlesher@digium.com>
+
+ * addons/ooh323c/src/oochannels.c: Fixing build
+
+ * channels/chan_usbradio.c, /: Merged revisions 284665 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r284665 | tilghman | 2010-09-02 11:07:19 -0500 (Thu, 02
+ Sep 2010) | 2 lines Fixing build. ........
+
+ * apps/app_queue.c, /: Merged revisions 284631 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r284631 | tilghman | 2010-09-02 00:30:16 -0500 (Thu, 02 Sep 2010)
+ | 7 lines Don't reset queue stats on a module reload. (closes
+ issue #17535) Reported by: raarts Patches:
+ 20100819__issue17535.diff.txt uploaded by tilghman (license 14)
+ ........
+
+ * channels/chan_iax2.c, apps/app_queue.c, apps/app_getcpeid.c,
+ apps/app_followme.c, main/loader.c, apps/app_speech_utils.c,
+ pbx/pbx_loopback.c, channels/chan_dahdi.c, funcs/func_aes.c,
+ include/asterisk/module.h, pbx/pbx_realtime.c, pbx/pbx_dundi.c,
+ apps/app_stack.c, channels/chan_mgcp.c, apps/app_voicemail.c,
+ apps/app_adsiprog.c, channels/chan_sip.c, channels/chan_agent.c:
+ When optional_api is non-optional, force dependent modules to be
+ loaded. (closes issue #17707) Reported by: ira Patches:
+ 20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman
+ (license 14) Tested by: tilghman Review:
+ https://reviewboard.asterisk.org/r/876/
+
+ * include/asterisk/channel.h, res/res_jabber.c, res/res_pktccops.c,
+ main/poll.c, channels/chan_usbradio.c, include/asterisk/select.h
+ (added), channels/chan_phone.c, channels/chan_misdn.c, configure,
+ main/features.c, include/asterisk/poll-compat.h,
+ tests/test_poll.c (added), addons/ooh323c/src/oochannels.c,
+ main/asterisk.c, addons/ooh323c/src/ooSocket.h, main/stun.c,
+ res/res_ais.c, /, include/asterisk/autoconfig.h.in, configure.ac,
+ channels/console_video.c: Merged revisions 284593,284595 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284593 | tilghman | 2010-09-01 17:59:50 -0500
+ (Wed, 01 Sep 2010) | 18 lines Merged revisions 284478 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010)
+ | 11 lines Ensure that all areas that previously used select(2)
+ now use poll(2), with implementations that need poll(2)
+ implemented with select(2) safe against 1024-bit overflows. This
+ is a followup to the fix for the pthread timer in 1.6.2 and
+ beyond, fixing a potential crash bug in all supported releases.
+ (closes issue #17678) Reported by: russell Branch:
+ https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select
+ Review: https://reviewboard.asterisk.org/r/824/ ........
+ ................ r284595 | tilghman | 2010-09-01 22:57:43 -0500
+ (Wed, 01 Sep 2010) | 2 lines Failed to rerun bootstrap.sh after
+ last commit ................
+
+2010-09-01 21:47 +0000 [r284561] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: During request to dialog matching, verify
+ init_ruri is present before comparing. During request to dialog
+ matching, we attempt a best effort routine for fork detection
+ which requires several elements to be in place. The dialog's
+ initial request uri is one of those elements. Since it is best
+ effort, if the init_ruri is not present for some reason we can
+ not proceed with that routine.
+
+2010-09-01 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.0-beta5 released.
+
+2010-09-01 18:44 +0000 [r284477] Terry Wilson <twilson@digium.com>
+
+ * res/res_srtp.c, res/res_rtp_asterisk.c,
+ include/asterisk/res_srtp.h, main/rtp_engine.c,
+ channels/chan_sip.c: Fix SRTP for changing SSRC and multiple
+ a=crypto SDP lines Adding code to Asterisk that changed the SSRC
+ during bridges and masquerades broke SRTP functionality. Also
+ broken was handling the situation where an incoming INVITE had
+ more than one crypto offer. This patch caches the SRTP policies
+ the we use so that we can change the ssrc and inform libsrtp of
+ the new streams. It also uses the first acceptable a=crypto line
+ from the incoming INVITE. (closes issue #17563) Reported by:
+ Alexcr Patches: srtp.diff uploaded by twilson (license 396)
+ Tested by: twilson Review:
+ https://reviewboard.asterisk.org/r/878/
+
+2010-09-01 18:16 +0000 [r284415-284473] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_pgsql.c, /: Merged revisions 284472 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r284472 | tilghman | 2010-09-01 13:13:35 -0500 (Wed, 01
+ Sep 2010) | 5 lines Don't warn on floats and timestamps (closes
+ issue #17082) Reported by: coolmig ........
+
+ * /, channels/chan_sip.c: Merged revisions 284399 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284399 | tilghman | 2010-08-31 15:18:32 -0500
+ (Tue, 31 Aug 2010) | 14 lines Merged revisions 284393 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010)
+ | 7 lines Don't send a devstate change on poke_noanswer if the
+ state did not change. (closes issue #17741) Reported by: schmidts
+ Patches: chan_sip.c.patch uploaded by schmidts (license 1077)
+ ........ ................
+
+2010-08-31 19:00 +0000 [r284318] Leif Madsen <lmadsen@digium.com>
+
+ * configs/say.conf.sample, /: Merged revisions 284317 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284317 | lmadsen | 2010-08-31 13:59:31 -0500
+ (Tue, 31 Aug 2010) | 15 lines Merged revisions 284316 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31 Aug 2010)
+ | 7 lines Update say.conf.sample to match the rules in say.c
+ (closes issue #17835) Reported by: RoadKill Patches:
+ say.conf.sample.patch.rules uploaded by RoadKill (license 933)
+ Tested by: RoadKill ........ ................
+
+2010-08-30 22:28 +0000 [r284281] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_festival.c: Merged revisions 284280 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r284280 | tilghman | 2010-08-30 17:27:06 -0500 (Mon, 30 Aug 2010)
+ | 11 lines Fix 3 coding errors: 1) After we close FD, we should
+ not be trying to write to it. 2) Call _exit(0), not exit(0), to
+ avoid running shutdown routines in a child. 3) Use endian, not
+ processor, detection to ensure bytes are written in the correct
+ order. (closes issue #15706) Reported by: modelnine Patches:
+ asterisk-1.6.1.1-festival-debug.patch uploaded by modelnine
+ (license 865) Tested by: gmartinez ........
+
+2010-08-29 07:05 +0000 [r284096-284158] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/res_curl.conf.sample (added): Missed adding this file
+
+ * sounds: Also ignore the checksums
+
+ * configs/cel_odbc.conf.sample (added), cel/cel_adaptive_odbc.c
+ (removed), cel/cel_odbc.c (added),
+ configs/cel_adaptive_odbc.conf.sample (removed): Rename CEL
+ adaptive driver to plain driver, since there isn't another ODBC
+ driver (and the other CEL drivers have adaptive capabilities,
+ anyway).
+
+2010-08-28 21:29 +0000 [r284065] Russell Bryant <russell@digium.com>
+
+ * main/manager.c: Be more flexible with whitespace on AMI action
+ headers. Previously, this code required exactly one space to be
+ after the ':' in headers for an AMI action. This now makes
+ whitespace optional, and allows whitespace that is there to vary
+ in amount. (closes issue #17862) Reported by: cmoye Patches:
+ manager.c.patch_trunk uploaded by cmoye (license 858)
+ manager.c.patch_1.8 uploaded by cmoye (license 858) Tested by:
+ cmoye
+
+2010-08-27 22:37 +0000 [r284032] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 284002 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284002 | dvossel | 2010-08-27 17:27:50 -0500
+ (Fri, 27 Aug 2010) | 14 lines Merged revisions 283960 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010)
+ | 8 lines Parse all "Accept" headers for SIP SUBSCRIBE requests.
+ (closes issue #17758) Reported by: ibc Patches:
+ multiple_accept_headers_1.4.diff uploaded by dvossel (license
+ 671) ........ ................
+
+2010-08-27 21:33 +0000 [r283951] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_realtime.c: Print exten@context:priority in verbose
+ messages from pbx_realtime.
+
+2010-08-27 20:31 +0000 [r283882] Jason Parker <jparker@digium.com>
+
+ * main/config.c, addons/res_config_mysql.c, res/res_config_odbc.c,
+ /: Merged revisions 283881 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r283881 | qwell | 2010-08-27 15:30:27 -0500
+ (Fri, 27 Aug 2010) | 15 lines Merged revisions 283880 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283880 | qwell | 2010-08-27 15:29:11 -0500 (Fri, 27 Aug 2010) |
+ 8 lines Fix issue with decoding ^-escaped characters in realtime.
+ (closes issue #17790) Reported by: denzs Patches:
+ 17790-chunky.diff uploaded by qwell (license 4) Tested by: qwell,
+ denzs ........ ................
+
+2010-08-26 23:47 +0000 [r283770] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c: Convert MOH to use generic timers. (closes
+ issue #17726) Reported by: lmadsen Patches:
+ 20100825__issue17726__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: tilghman
+
+2010-08-26 15:26 +0000 [r283692] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 283691 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r283691 | dvossel | 2010-08-26 10:24:40 -0500
+ (Thu, 26 Aug 2010) | 25 lines Merged revisions 283690 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010)
+ | 19 lines Fixed how Asterisk destroys a dialog on channel hangup
+ before invite receives a response. If an ast_channel with a SIP
+ tech pvt hangs up before the sip dialog gets a response to its
+ outgoing INVITE, Asterisk used to pretend_ack the INVITE. This is
+ not rfc compliant and results in confusion at the other endpoint.
+ sip_pretend_ack will ack and remove all the packets in the
+ retransmit queue. This means that the INVITE will stop
+ retransmitting, and that any response to that INVITE that comes
+ after the pretend_ack occurs will be ignored. Instead of faking
+ any sort of acknowledgement for an outgoing INVITE during an
+ internal hangup, we should let the protocol stack process the
+ INVITE transaction and terminate the dialog properly. This is
+ achieved by setting the PENDING_BYE flag. When this flag is used,
+ once the dialog proceeds to an escapable state the transaction
+ will either be canceled with a SIP_CANCEL or completed followed
+ immediately by a BYE. Attempting to do this any other way is
+ incorrect. If the endpoint is not responding to the INVITE
+ request, the INVITE must continue to be retransmitted until it
+ times out which will result in the dialog being destroyed.
+ ........ ................
+
+2010-08-26 13:26 +0000 [r283627-283659] Russell Bryant <russell@digium.com>
+
+ * res/res_odbc.c: Slight improvement to a debug message.
+
+ * keys/iaxtel.pub (removed), keys/freeworlddialup.pub (removed),
+ Makefile: Remove public keys that are no longer useful.
+
+ * configs/manager.conf.sample: Move httptimeout out from in between
+ port and bindaddr.
+
+2010-08-25 22:57 +0000 [r283595] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 283594 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r283594 | dvossel | 2010-08-25 17:56:42 -0500 (Wed, 25 Aug 2010)
+ | 7 lines Add to and from tags to NOTIFY dialog-info xml body so
+ pickup can occur. When pedantic mode is used, the dialog-info xml
+ generated during a ringing event must contain the to and from tag
+ values. Otherwise if a pickup occurs using INVITE with replaces,
+ Astrisk will not be able to locate the subscription. ........
+
+2010-08-25 16:12 +0000 [r283561] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_odbc.c: Initialize connect timeout on each time through
+ the loop. (closes issue #17911) Reported by: wurstsalat
+
+2010-08-25 15:54 +0000 [r283559] David Vossel <dvossel@digium.com>
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c: Merged
+ revisions 283558 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r283558 | dvossel | 2010-08-25 10:52:54 -0500 (Wed, 25 Aug 2010)
+ | 10 lines Asterisk will not advertise session timers are
+ supported when 'session-timers=refuse' is used. Asterisk now
+ dynamically builds the "Supported" header depending on what is
+ enabled/disabled in sip.conf. Session timers used to always be
+ advertised as being supported even when they were disabled in the
+ configuration. This caused problems with some end points. (issue
+ #17005) ........
+
+2010-08-25 14:55 +0000 [r283527] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Convert ast_log(LOG_DEBUG, ...) to
+ ast_debug(...)
+
+2010-08-24 20:34 +0000 [r283493] David Vossel <dvossel@digium.com>
+
+ * UPGRADE.txt, configs/sip.conf.sample, channels/sip/include/sip.h:
+ Changes the default behavior for sip.conf's pedantic option from
+ "no" to "yes".
+
+2010-08-24 18:56 +0000 [r283457] Leif Madsen <lmadsen@digium.com>
+
+ * res/res_rtp_asterisk.c, channels/chan_sip.c: Fix issue where TOS
+ is no longer set on RTP packets. Fix issue where the tos is no
+ longer being set on RTP packets through res_rtp_asterisk. (closes
+ issue #17890) Reported by: elguero Patches: qos_18.diff uploaded
+ by elguero (license 37) Review:
+ https://reviewboard.asterisk.org/r/868
+
+2010-08-24 16:11 +0000 [r283382] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 283381 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r283381 | dvossel | 2010-08-24 11:07:37 -0500
+ (Tue, 24 Aug 2010) | 18 lines Merged revisions 283380 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010)
+ | 11 lines This fix makes sure the ast_channel hangs up correctly
+ when the dialog's PENDING_BYE flag is set. When the pending bye
+ flag is used, it is possible that the dialog will terminate and
+ leave the sip_pvt->owner channel up. This is because we never
+ hangup the ast_channel after sending the SIP_BYE request. When we
+ receive the response for the SIP_BYE we set need_destroy which we
+ would expect to destroy the dialog on the next do_monitor loop,
+ but this is not the case. The dialog will only be destroyed once
+ the owner is hungup even with the need_destroy flag set. This
+ patch sets the softhangup flag on the ast_channel when a SIP_BYE
+ request is sent as a result of the pending bye flag. ........
+ ................
+
+2010-08-24 12:49 +0000 [r283350] Russell Bryant <russell@digium.com>
+
+ * funcs/func_odbc.c: Don't attempt to release a NULL ODBC handle.
+
+2010-08-23 21:33 +0000 [r283319] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_adaptive_odbc.c, cdr/cdr_odbc.c, cel/cel_adaptive_odbc.c,
+ /: Merged revisions 283318 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r283318 | tilghman | 2010-08-23 16:32:14 -0500 (Mon, 23 Aug 2010)
+ | 2 lines CDR drivers depend upon res_odbc, not directly on the
+ ODBC libraries ........
+
+2010-08-23 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.0-beta4 Released.
+
+2010-08-23 13:35 +0000 [r283177-283241] Russell Bryant <russell@digium.com>
+
+ * configs/cel.conf.sample: Add sample configuration for cel_radius.
+
+ * main/cel.c, include/asterisk/cel.h: Make the AST_CEL_AMA enum
+ match up with the AST_CDR_ ama flag values. Really, having 2
+ enums for this is silly and error prone, demonstrated by the
+ crash that I hit because there was an assumption in the code that
+ the values in each matched up. However, this is a quick fix to
+ get them to match up so it will work.
+
+ * main/cel.c: Don't blow up on an invalid AMA flag.
+
+ * configs/cel_custom.conf.sample: Tack on ${eventextra} to the
+ sample cel_custom.conf.
+
+ * configs/cel_custom.conf.sample: Cut down on excessive quotation.
+
+2010-08-23 12:06 +0000 [r283175] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_stun_monitor.c: Don't fail to start if the config file is
+ missing.
+
+2010-08-23 11:58 +0000 [r283173] Russell Bryant <russell@digium.com>
+
+ * configs/cel_custom.conf.sample: Expand cel_custom.conf.sample.
+ Include the usage of CSV_QUOTE() to ensure data has valid CSV
+ formatting. Also list the special CEL variables that are
+ available for use in the mapping.
+
+2010-08-20 16:51 +0000 [r283050-283125] Richard Mudgett <rmudgett@digium.com>
+
+ * /: Recorded merge of revisions 283124 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r283124 | rmudgett | 2010-08-20 11:48:10 -0500
+ (Fri, 20 Aug 2010) | 16 lines Merged revisions 283123 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r283123 | rmudgett | 2010-08-20 11:46:22 -0500
+ (Fri, 20 Aug 2010) | 9 lines Merged revision 278274 from
+ https://origsvn.digium.com/svn/asterisk/trunk .......... r278274
+ | rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1
+ line Reference correct struct member for unlikely event
+ PRI_EVENT_CONFIG_ERR. .......... ................
+ ................
+
+ * channels/sig_pri.c, /: Merged revisions 283049 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r283049 | rmudgett | 2010-08-20 10:31:03 -0500
+ (Fri, 20 Aug 2010) | 29 lines Merged revisions 283048 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20 Aug 2010)
+ | 22 lines Q931 - Sending PROGRESS after sending ALERTING is a
+ protocol error The PRI layer in chan_dadhi will check if a
+ PROGRESS message has already been sent, and not allow sending
+ another (although that is technically allowed by the Q931 spec),
+ however it does not protect against sending an ALERTING and then
+ sending a PROGRESS message, which is a violation of the
+ specification. Most switches don't seem to care too deeply about
+ this, but some do, and will disconnect the call when receiving
+ this invalid sequence. Protocol specification reference:
+ T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview
+ protocol control (network side) point-point (sheet 3 of 8)"
+ (closes issue #17874) Reported by: nic_bellamy Patches:
+ asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by
+ nic bellamy (license 299)
+ asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded
+ by nic bellamy (license 299)
+ asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded
+ by nic bellamy (license 299) ........ ................
+
+2010-08-20 12:45 +0000 [r282979-283013] Russell Bryant <russell@digium.com>
+
+ * configs/cel_adaptive_odbc.conf.sample: Fix a typo in a column
+ name.
+
+ * apps/app_celgenuserevent.c: Add an argument missing from the
+ CELGenUserEvent documentation.
+
+2010-08-19 21:07 +0000 [r282891-282895] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 282894 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r282894 | dvossel | 2010-08-19 16:05:54 -0500
+ (Thu, 19 Aug 2010) | 18 lines Merged revisions 282893 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010)
+ | 11 lines tos_sip option was not being set correctly When
+ tos_sip is used, the tos of the sip socket is only set correctly
+ if the socket binding changes on a reload. If the binding stays
+ the same but the TOS changes, the new tos value would not take
+ into effect. This patch fixes that. (closes issue #17712)
+ Reported by: nickb ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 282890 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r282890 | dvossel | 2010-08-19 15:31:22 -0500 (Thu, 19 Aug 2010)
+ | 5 lines fixes sip peer memory leaks in the peer_by_ip table
+ (issue #17798) ........
+
+2010-08-19 20:01 +0000 [r282860] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 282859 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r282859 | mnicholson | 2010-08-19 14:44:00 -0500
+ (Thu, 19 Aug 2010) | 23 lines Merged revisions 277944 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul
+ 2010) | 16 lines Regression with T.38 negotiation Prior to
+ 1.4.26.3 T.38 negotiation worked properly, in the case of the
+ reporter. (issue #16852) Reported by: cfc (closes issue #16705)
+ Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded
+ by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa,
+ samdell3 Review: https://reviewboard.asterisk.org/r/754/ ........
+ ................
+
+2010-08-19 14:44 +0000 [r282826] Tilghman Lesher <tlesher@digium.com>
+
+ * main/netsock2.c: Only output debugging if the debug level is on.
+
+2010-08-19 02:18 +0000 [r282740] Terry Wilson <twilson@digium.com>
+
+ * configs/sip.conf.sample, /: Merged revisions 282730 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r282730 | twilson | 2010-08-18 21:14:28 -0500
+ (Wed, 18 Aug 2010) | 9 lines Merged revisions 282729 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18
+ Aug 2010) | 2 lines Add some documentation about codec
+ negotiation to sip.conf ........ ................
+
+2010-08-18 15:28 +0000 [r282671-282672] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h: Use the correct type for aoce_delayhangup bit
+ field.
+
+ * channels/chan_dahdi.c: Use the correct operator when calculating
+ the PRI span devstate.
+
+2010-08-18 13:10 +0000 [r282639] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Properly handle 200 and unknown responses
+ conatined in NOTIFY requests received in response to REFER
+ requests. This patch fixes the way asterisk handles NOTIFY
+ requests received in response to REFER requests. These changes to
+ NOTIFY handler were first introduced in r217482. This new change
+ properly handles the 200 response by queueing an
+ AST_TRANSFER_SUCCESS control frame and also prevents that control
+ frame from being queued when provisional and unknown responses
+ are received. (issue #17486) Reported by: davidw Tested by:
+ mnicholson (issue #12713) Reported by: davidw Review:
+ https://reviewboard.asterisk.org/r/860/
+
+2010-08-18 12:30 +0000 [r282638] Russell Bryant <russell@digium.com>
+
+ * channels/chan_multicast_rtp.c: Split _all_ arguments before
+ parsing them. This fixes multicast RTP paging using linksys mode.
+
+2010-08-18 07:49 +0000 [r282608] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/sig_pri.c, /: Merged revisions 282607 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r282607 | tilghman | 2010-08-18 02:43:14 -0500 (Wed, 18 Aug 2010)
+ | 9 lines Don't warn on callerid when completely text, instead of
+ numeric with localdialplan prefixes. (closes issue #16770)
+ Reported by: jamicque Patches: 20100413__issue16770.diff.txt
+ uploaded by tilghman (license 14) 20100811__issue16770.diff.txt
+ uploaded by tilghman (license 14) Tested by: jamicque ........
+
+2010-08-17 21:36 +0000 [r282543-282577] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 282576 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r282576 | dvossel | 2010-08-17 16:35:17 -0500 (Tue, 17 Aug 2010)
+ | 9 lines fixes no default transport for temp peer creation in
+ chan_sip (closes issue #17829) Reported by: falves11 Patches:
+ issue_17829.rev1.txt uploaded by russell (license 2)
+ issue_17829.diff uploaded by dvossel (license 671) Tested by:
+ falves11 ........
+
+ * channels/chan_iax2.c: ACCEPT message should respond with the new
+ FORMAT2 ie (closes issue #17804) Reported by: tpanton
+
+ * include/asterisk/unaligned.h: fixes truncated uint64_t value in
+ put_unaligned_uint64_t() function (issue #17804)
+
+2010-08-16 18:01 +0000 [r282470] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/asterisk.tex, doc/tex/sounds.tex (added), /: Merged
+ revisions 282469 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r282469 | lmadsen | 2010-08-16 13:00:09 -0500 (Mon, 16 Aug 2010)
+ | 7 lines Add information about creating sounds files using the
+ sounds tools publically available so that others can create their
+ own sounds prompts using the same tools we use to generate sounds
+ releases. This allows people creating their own prompts to sound
+ consistent with the prompts available from the open source
+ project. SWP-595 ........
+
+2010-08-16 17:53 +0000 [r282468] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, /: Merged revisions 282467 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r282467 | twilson | 2010-08-16 12:32:01 -0500
+ (Mon, 16 Aug 2010) | 23 lines Merged revisions 282430 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010)
+ | 16 lines Send a SRCCHANGE indication when we masquerade
+ Masquerading a channel means that the src of the audio is
+ potentially changing, so send a SRCCHANGE so that RTP-based media
+ streams can get a new SSRC generated to reflect the change.
+ Original patch by addix (along with lots of testing--thanks!).
+ (closes issue #17007) Reported by: addix Patches:
+ 1001-reset-SSRC-original-channel.diff uploaded by addix (license
+ 1006) srcchange.diff uploaded by twilson (license 396) Tested by:
+ addix, twilson Review: https://reviewboard.asterisk.org/r/862/
+ ........ ................
+
+2010-08-14 04:53 +0000 [r282366] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c, include/asterisk/sched.h: Fix our FRACKing
+ issue with chan_iax2 a different way. Review:
+ https://reviewboard.asterisk.org/r/861/
+
+2010-08-13 23:53 +0000 [r282334] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: PRI CCSS may use a stale dial string for
+ the recall dial string. If an outgoing call negotiates a
+ different B channel than initially requested, the saved original
+ dial string was not transferred to the new B channel. CCSS uses
+ that dial string to generate the recall dial string.
+
+2010-08-13 22:23 +0000 [r282236-282302] David Vossel <dvossel@digium.com>
+
+ * UPGRADE.txt, configs/sip.conf.sample, CHANGES,
+ channels/chan_sip.c: remove current STUN support from chan_sip.c
+ This patch removes the current broken/useless stun support from
+ chan_sip. (closes issue #17622) Reported by: philipp2 Review:
+ https://reviewboard.asterisk.org/r/855/
+
+ * CHANGES: res_stun_monitor and corresponding options CHANGES
+ documentation
+
+ * configs/res_stun_monitor.conf.sample (added),
+ configs/sip.conf.sample, channels/chan_iax2.c,
+ configs/iax.conf.sample, channels/chan_sip.c,
+ include/asterisk/event_defs.h, res/res_stun_monitor.c (added):
+ res_stun_monitor for monitoring network changes behind a NAT
+ device Review: https://reviewboard.asterisk.org/r/854
+
+ * /, channels/chan_sip.c: Merged revisions 282235 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r282235 | dvossel | 2010-08-13 13:54:53 -0500 (Fri, 13 Aug 2010)
+ | 16 lines only do magic pickup when notifycid is enabled A new
+ way of doing BLF pickup was introduced into 1.6.2. This feature
+ adds a call-id value into the XML of a SIP_NOTIFY message sent to
+ alert a subscriber that a device is ringing. This option should
+ only be enabled when the new 'notifycid' option is set... but
+ this was not the case. Instead the call-id value was included for
+ every RINGING Notify message, which caused a regression for
+ people who used other methods for call pickup. (closes issue
+ #17633) Reported by: urosh Patches: chan_sip.txt uploaded by
+ urosh (license ) blf_cid_issue.diff uploaded by dvossel (license
+ 671) Tested by: dvossel, urosh, okrief, alecdavis ........
+
+2010-08-13 16:02 +0000 [r282200-282201] Terry Wilson <twilson@digium.com>
+
+ * configure.ac: Whitespace fix :-/
+
+ * configure, configure.ac: Detect when libsrtp cannot be linked in
+ a shared library The libsrtp build system currently does not
+ produce a shared library or a static library compiled with -fPIC,
+ so on 64-bit systems it is possible that we will get a compile
+ error if libsrtp is installed and res_srtp is selected in
+ menuselect. This patch attempts to detect this situation and
+ provide the user with instructions to work around the problem.
+
+2010-08-12 22:51 +0000 [r282131] Jason Parker <jparker@digium.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 282130 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r282130 | qwell | 2010-08-12 17:50:54 -0500
+ (Thu, 12 Aug 2010) | 9 lines Merged revisions 282129 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r282129 | qwell | 2010-08-12 17:49:28 -0500 (Thu, 12 Aug
+ 2010) | 1 line Register CLI commands before parsing config, in
+ case there is a config error. ........ ................
+
+2010-08-12 22:06 +0000 [r282098] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/ccss.h, main/ccss.c: Separate call completion
+ config parameter allocation and default initialization. If you
+ ever have a need to reset the call completion config parameters
+ to defaults, now you can. And no Virginia, C++ idioms do not
+ always work in C.
+
+2010-08-12 20:41 +0000 [r282066] Russell Bryant <russell@digium.com>
+
+ * CHANGES, main/cli.c: Add a "core reload" CLI command. Review:
+ https://reviewboard.asterisk.org/r/859/
+
+2010-08-12 20:15 +0000 [r282047] David Vossel <dvossel@digium.com>
+
+ * CHANGES, include/asterisk/translate.h, main/cli.c,
+ main/translate.c: improved translation paths for wideband codecs
+ The problem I'm addressing is that Asterisk's current method of
+ building the least cost translation paths between codecs does not
+ take into account sample rate. For instance, it was possible for
+ siren14 (a 32khz codec), to contain the a translation path to
+ siren7 (a 16khz audio codec) that goes through slin at 8khz. In
+ this case Asterisk takes a 32khz codec, down samples it to 8khz
+ and then up samples it to 16khz which is terrible regardless if
+ it is computationally less expensive. This patch now builds
+ translation paths that give priority to maintaining the best
+ possible sample rate before taking into consideration
+ computational cost. This patch also adds cli commands to expose
+ what translation paths are actually being used. Changes: 1.
+ Translation paths will never contain a step that changes the
+ sample rate unless absolutely necessary. 2. When choosing the
+ best codec to make two channels compatible. Shared codecs with
+ the highest sample rate are given priority. 3. A new cli command
+ to show all translation paths available for a specific codec
+ 'core show translation paths [codec name]' has been added. 4.
+ 'core show translation' which displays the translation matrix now
+ includes the new higher bit audio codecs in the table. 5. 'core
+ show channel [channel name]' now displays the translation paths
+ if translation is used. (closes issue #16841) Reported by:
+ dvossel Review: https://reviewboard.asterisk.org/r/842/
+
+2010-08-12 18:03 +0000 [r281982-282015] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c: Put back pointer value output for ast_debug(), such
+ that it is only removed for verbose output.
+
+ * main/pbx.c: Remove debugging output from verbose messages.
+ Pointer values to internal objects is not terribly useful to
+ users in the verbose messages about adding extensions and
+ contexts.
+
+2010-08-12 03:03 +0000 [r281913] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 281912 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r281912 | jpeeler | 2010-08-11 22:01:38 -0500
+ (Wed, 11 Aug 2010) | 27 lines Merged revisions 281911 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010)
+ | 20 lines Ensure SSRC is changed when media source is changed to
+ resolve audio delay. This change causes the SSRC to change right
+ before the channels are bridged, which is what used to happen. It
+ seems that fixes were made to attempt limiting SSRC changes,
+ targeted mainly at sending DTMF. DTMF is not affecting the SSRC
+ with this change. There are two other control frames sent in
+ ast_channel_bridge that probably should also be changed to
+ AST_CONTROL_SRCCHANGE as well, but I'm going to leave this change
+ up to the discretion of resolving issue #17007. For reference -
+ old review implementing new control frame SRCCHANGE:
+ https://reviewboard.asterisk.org/r/540 (closes issue #17404)
+ Reported by: sdolloff Patches: bug17404.patch uploaded by jpeeler
+ (license 325) Tested by: sdolloff ........ ................
+
+2010-08-11 21:12 +0000 [r281875] Leif Madsen <lmadsen@digium.com>
+
+ * configs/say.conf.sample, /: Merged revisions 281873 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r281873 | lmadsen | 2010-08-11 16:09:47 -0500
+ (Wed, 11 Aug 2010) | 14 lines Merged revisions 281819 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r281819 | lmadsen | 2010-08-11 13:28:10 -0500 (Wed, 11 Aug 2010)
+ | 6 lines Add Danish support to say.conf.sample (closes issue
+ #17836) Reported by: RoadKill Patches: say.conf.sample.patch.dk
+ uploaded by RoadKill (license 933) ........ ................
+
+2010-08-11 21:11 +0000 [r281874] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: handle all possible responses to REFER
+ requests (closes issue #17486) Reported by: davidw Patches:
+ Issue17486-counterbid.diff.txt uploaded by davidw (license 780)
+ Tested by: davidw Review: https://reviewboard.asterisk.org/r/837/
+
+2010-08-11 20:30 +0000 [r281870] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_analog.c, channels/sig_analog.h: Fix a call to
+ analog_set_pulsedial() not setting 0 or 1 only. * Also a couple
+ minor tweaks.
+
+2010-08-11 17:54 +0000 [r281764] Leif Madsen <lmadsen@digium.com>
+
+ * configs/say.conf.sample, /: Merged revisions 281763 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r281763 | lmadsen | 2010-08-11 12:54:09 -0500
+ (Wed, 11 Aug 2010) | 14 lines Merged revisions 281762 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r281762 | lmadsen | 2010-08-11 12:51:40 -0500 (Wed, 11 Aug 2010)
+ | 6 lines Allow say.conf to handle large numbers ending with
+ multiple zeros. (closes issue #17833) Reported by: RoadKill
+ Patches: say.conf.sample.patch.largenumbers uploaded by RoadKill
+ (license 933) ........ ................
+
+2010-08-11 17:27 +0000 [r281760] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Avoid a deadlock in
+ add_header_max_forwards(). Related to r276951
+
+2010-08-11 15:18 +0000 [r281723] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_readexten.c: Merged revisions 281722 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r281722 | tilghman | 2010-08-11 10:17:20 -0500 (Wed, 11
+ Aug 2010) | 7 lines Only set status TIMEOUT, if we have no
+ digits. (closes issue #15188) Reported by: jcovert Patches:
+ app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license
+ 551) ........
+
+2010-08-11 13:30 +0000 [r281687] <simon.perreault@viagenie.ca>
+
+ * include/asterisk/netsock2.h, configs/sip.conf.sample,
+ channels/sip/config_parser.c, main/netsock2.c: Fix parsing of
+ IPv6 address literals in outboundproxy (closes issue #17757)
+ Reported by: oej Patches: 17757.diff uploaded by sperreault
+ (license 252) sip.conf.diff uploaded by sperreault (license 252)
+ Tested by: oej
+
+2010-08-10 21:47 +0000 [r281568-281650] Russell Bryant <russell@digium.com>
+
+ * UPGRADE.txt, configs/sip.conf.sample, channels/sip/include/sip.h:
+ Change the default value for alwaysauthreject in sip.conf to
+ "yes". (closes issue #17756) Reported by: oej
+
+ * main/sched.c, /: Merged revisions 281574 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r281574 | russell | 2010-08-10 13:04:32 -0500 (Tue, 10 Aug 2010)
+ | 9 lines Don't move the time threshold for running scheduled
+ events on every iteration. Instead, only calculate the time
+ threshold each time ast_sched_runq() is called. (closes issue
+ #17742) Reported by: schmidts Patches: sched.c.patch uploaded by
+ schmidts (license 1077) ........
+
+ * apps/app_dial.c, /: Merged revisions 281567 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r281567 | russell | 2010-08-10 12:47:13 -0500
+ (Tue, 10 Aug 2010) | 15 lines Merged revisions 281566 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010)
+ | 8 lines Reset visible indication after answer. (closes issue
+ #17641) Reported by: klaus3000 Patches:
+ ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by
+ klaus3000 (license 65) Tested by: schmidts ........
+ ................
+
+2010-08-10 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.0-beta3 Released.
+
+2010-08-10 17:48 +0000 [r281529-281568] Russell Bryant <russell@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 281567 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r281567 | russell | 2010-08-10 12:47:13 -0500
+ (Tue, 10 Aug 2010) | 15 lines Merged revisions 281566 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010)
+ | 8 lines Reset visible indication after answer. (closes issue
+ #17641) Reported by: klaus3000 Patches:
+ ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by
+ klaus3000 (license 65) Tested by: schmidts ........
+ ................
+
+ * channels/chan_sip.c: Ensure that the proper external address is
+ used for the RTP destination. (closes issue #17044) Reported by:
+ ebroad Tested by: ebroad Review:
+ https://reviewboard.asterisk.org/r/566/
+
+ * main/cli.c: Resolve a problem with channel name tab completion.
+ Hitting tab without typing any part of a channel name resulted in
+ no results. This now results in getting a full list of active
+ channels, just as it did in previous versions of Asterisk.
+ Review: https://reviewboard.asterisk.org/r/818/
+
+2010-08-10 07:26 +0000 [r281497] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c: Fixed the issue caused by EXTEN including
+ user parameters.
+
+2010-08-09 23:04 +0000 [r281466] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_local.c: Add some more stuff to copy from 281429.
+
+2010-08-09 20:47 +0000 [r281432] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 281430 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r281430 | dvossel | 2010-08-09 15:46:50 -0500 (Mon, 09 Aug 2010)
+ | 13 lines fixes SIP peers memory leak We zeroed out the peer's
+ addr before it was removed from the peers_by_ip container. This
+ made it impossible to be removed from the container as the addr
+ is the key used by the container to find the peer. (closes issue
+ #17774) Reported by: kkm Patches:
+ 017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888)
+ 017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888)
+ ........
+
+2010-08-09 20:43 +0000 [r281429] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 281391 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r281391 | jpeeler | 2010-08-09 15:07:29 -0500
+ (Mon, 09 Aug 2010) | 20 lines Merged revisions 281390 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09 Aug 2010)
+ | 13 lines Prevent loss of Caller ID information set on local
+ channel after masquerade. Caller ID set on the channel before a
+ masquerade occurs when using a local channel would cause the
+ information to be lost. The problem was that the information was
+ set on a channel destined to be hung up. The somewhat confusing
+ fix is to detect if any Caller ID has been set on the channel and
+ if so preswap the Caller ID data so that basically the masquerade
+ puts the data back. (closes issue #17138) Reported by: kobaz
+ Review: https://reviewboard.asterisk.org/r/847/ ........
+ ................
+
+2010-08-09 14:49 +0000 [r281358] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: Validate minrate, maxrate, and modem settings
+ before attempting a fax session. FAX-224
+
+2010-08-09 14:31 +0000 [r281356] <simon.perreault@viagenie.ca>
+
+ * configs/sip.conf.sample: Added comment about IPv4-mapped IPv6
+ addresses and the output of netstat.
+
+2010-08-09 12:51 +0000 [r281294-281325] Russell Bryant <russell@digium.com>
+
+ * configs/cdr.conf.sample: Add a couple of default values to the
+ documentation of cdr.conf.
+
+ * configs/cdr.conf.sample: Reorder some options in cdr.conf.sample.
+ Put all of the options that affect the contents of CDRs together,
+ instead of having the batch mode options in the middle of them.
+
+2010-08-06 18:57 +0000 [r281085] Tilghman Lesher <tlesher@digium.com>
+
+ * main/utils.c: Fix alignment of stringfields on the SPARC
+ architecture (closes issue #17789) Reported by: Ian Mason
+ Patches: 20100806__issue17789__2.diff.txt uploaded by tilghman
+ (license 14) Tested by: Ian_Mason
+
+2010-08-05 13:16 +0000 [r281052] Russell Bryant <russell@digium.com>
+
+ * main/cdr.c, /: Merged revisions 281051 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r281051 | russell | 2010-08-05 08:11:32 -0500 (Thu, 05 Aug 2010)
+ | 9 lines Cleanup default option value handling for cdr.conf
+ [general]. The default values would differ depending on whether
+ or not cdr.conf exists. That is no longer the case. Apply a
+ default value to the unanswered option. Define all default values
+ as named constants. ........
+
+2010-08-05 07:46 +0000 [r280984] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/pbx.h, main/pbx.c, /: Merged revisions 280983
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r280983 | tilghman | 2010-08-05 02:40:47 -0500
+ (Thu, 05 Aug 2010) | 15 lines Merged revisions 280982 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010)
+ | 8 lines Change context lock back to a mutex, because
+ functionality depends upon the lock being recursive. (closes
+ issue #17643) Reported by: zerohalo Patches:
+ 20100726__issue17643.diff.txt uploaded by tilghman (license 14)
+ Tested by: zerohalo ........ ................
+
+2010-08-04 15:11 +0000 [r280909] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: Initialize FAXOPT() status variables in sendfax
+ and receivefax instead of when the details structure is created.
+
+2010-08-04 14:04 +0000 [r280809-280879] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_mgcp.c: Check cur value before attempting a deref.
+ (closes issue #17775) Reported by: svinson Patches:
+ 20100804__issue17775.diff.txt uploaded by tilghman (license 14)
+ Tested by: svinson (closes issue #17743) Reported by: tgruenberg
+ Patches: 20100804__issue17775.diff.txt uploaded by tilghman
+ (license 14) Tested by: tgruenberg
+
+ * CHANGES, funcs/func_strings.c: Sneak FIELDNUM() into 1.8. Returns
+ a 1-based index into a list of a specified item. Matches up with
+ FIELDQTY() and CUT(). (closes issue #17713) Reported by: gareth
+ Patches: svn-279754.diff uploaded by gareth (license 208) Tested
+ by: gareth, tilghman Review:
+ https://reviewboard.asterisk.org/r/810/
+
+2010-08-03 19:54 +0000 [r280777-280778] <simon.perreault@viagenie.ca>
+
+ * channels/chan_sip.c: Fixed IPv6-related SIP parsing bugs. (closes
+ issue #17663) Reported by: oej Patches: diff uploaded by
+ sperreault (license 252) diff2 uploaded by sperreault (license
+ 252) get_domain.diff uploaded by sperreault (license 252)
+
+ * configs/sip.conf.sample: Better documentation related to IPv6.
+ (closes issue #17737) Reported by: oej Patches: doc.diff uploaded
+ by sperreault (license 252) Tested by: mmichelson
+
+2010-08-03 18:48 +0000 [r280742] Russell Bryant <russell@digium.com>
+
+ * addons/Makefile, addons/mp3 (removed),
+ contrib/scripts/get_mp3_source.sh (added): Remove the MP3 decoder
+ source code and replace it with a small shell script. Review:
+ https://reviewboard.asterisk.org/r/836/
+
+2010-08-03 18:42 +0000 [r280624-280740] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/asterisk.sgml, /, doc/asterisk.8, doc/Makefile (added):
+ Merged revisions 280739 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r280739 | tilghman | 2010-08-03 13:39:28 -0500 (Tue, 03 Aug 2010)
+ | 2 lines Document -B and -W flags and regenerate manpage from
+ sgml ........
+
+ * apps/app_voicemail.c, /: Merged revisions 280671 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r280671 | tilghman | 2010-08-02 16:26:11 -0500 (Mon, 02
+ Aug 2010) | 2 lines Allow the pipe, but also allow the comma
+ ........
+
+ * main/Makefile: Make this a little more deterministic... we want
+ the latest value, not just a 1 somewhere.
+
+ * main/Makefile: Apparently, the values in makeopts are sometimes
+ 1:1 and sometimes 1. Compensate for this.
+
+2010-07-29 21:07 +0000 [r280557] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: Fix regression introduced in r1664. Give the fax
+ stack time to shutdown and populate the FAXOPT output variables.
+ FAX-222
+
+2010-07-29 20:43 +0000 [r280552] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 280551 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r280551 | dvossel | 2010-07-29 15:42:29 -0500 (Thu, 29 Jul 2010)
+ | 11 lines fixes wrong SRV query for TLS connection (closes issue
+ #17612) Reported by: marcelloceschia Patches:
+ chan-sip_srvQuery.patch uploaded by marcelloceschia (license
+ 1079) chan-sip_Trunk_srvQuery.patch uploaded by st (license 907)
+ chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia
+ (license 1079) Tested by: marcelloceschia, st, pabelanger
+ ........
+
+2010-07-29 20:35 +0000 [r280549] Russell Bryant <russell@digium.com>
+
+ * configs/ccss.conf.sample: Add header to ccss.conf to appease oej.
+ (closes issue #17755) Reported by: oej
+
+2010-07-29 19:47 +0000 [r280519] Sean Bright <sean@malleable.com>
+
+ * channels/sig_pri.c: Fix compilation error in chan_dahdi (strdupa
+ -> ast_strdupa). (closes issue #17751) Reported by: b11d Patches:
+ strdupa_oops.diff uploaded by malcolmd (license 924)
+
+2010-07-29 19:13 +0000 [r280450] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /: Merged revisions 280449 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r280449 | dvossel | 2010-07-29 14:05:25 -0500
+ (Thu, 29 Jul 2010) | 18 lines Merged revisions 280448 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010)
+ | 12 lines fixes issue with translator frame not getting freed A
+ translator frame even if it local storage so the translation path
+ can be freed. This issue prevented g729 licenses from being freed
+ up. (closes issue #17630) Reported by: manvirr Patches:
+ encoder_fix.diff uploaded by dvossel (license 671) Tested by:
+ manvirr, dvossel ........ ................
+
+2010-07-29 18:37 +0000 [r280414-280446] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * tests/test_utils.c: Remove res_crypto dependency.
+
+ * tests/test_utils.c: crypto_loaded_test depends on res_crypto,
+ else test will fail.
+
+2010-07-29 16:25 +0000 [r280391] Russell Bryant <russell@digium.com>
+
+ * main/rtp_engine.c: Don't blow up if get_codec() was not provided
+ in the RTP glue.
+
+2010-07-29 16:07 +0000 [r280346] Jean Galarneau <jgalarneau@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 280345 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r280345 | jeang | 2010-07-29 11:01:35 -0500
+ (Thu, 29 Jul 2010) | 10 lines Merged revisions 280341 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) |
+ 2 lines Fix a dsp structure leak occuring when a local channel is
+ put into a meetme conference, then masquaraded away. ABE-2422
+ ........ ................
+
+2010-07-29 15:57 +0000 [r280307-280343] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_usbradio.c: Use PRIx64 instead of PRId64 in format
+ string. related to r280302
+
+ * main/channel.c, channels/chan_local.c, /: Merged revisions 280306
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r280306 | mnicholson | 2010-07-29 08:45:11 -0500 (Thu, 29 Jul
+ 2010) | 2 lines Implement support for ast_channel_queryoption on
+ local channels. Currently only AST_OPTION_T38_STATE is supported.
+ ABE-2229 Review: https://reviewboard.asterisk.org/r/813/ ........
+ Additionally, pass AST_CONTROL_T38_PARAMETERS control frames
+ through generic bridges. This change appears to have been
+ unintentionally left out of rev 203699.
+
+2010-07-29 00:45 +0000 [r280302] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * channels/chan_usbradio.c: Use PRId64 with format_t
+
+2010-07-28 20:49 +0000 [r280269] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/sip/reqresp_parser.c: Give test category missing leading
+ slash
+
+2010-07-28 20:12 +0000 [r280235] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 280229 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r280229 | rmudgett | 2010-07-28 14:57:49 -0500 (Wed, 28
+ Jul 2010) | 2 lines Add missing enum value "unknown" to the SS7
+ called_nai and calling_nai config options. ........
+
+2010-07-28 20:03 +0000 [r280233] Jason Parker <jparker@digium.com>
+
+ * sounds/Makefile, /: Merged revisions 280231 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r280231 | qwell | 2010-07-28 15:02:27 -0500 (Wed, 28 Jul 2010) |
+ 6 lines Work around some silly behavior on BSD. A non-zero exit
+ from a subshell should make the build fail. (closes issue #17621)
+ ........
+
+2010-07-28 19:34 +0000 [r280225] Terry Wilson <twilson@digium.com>
+
+ * res/res_rtp_asterisk.c: Do rtp/rtcp debugging when it is turned
+ on w/o filtering
+
+2010-07-28 18:24 +0000 [r280195] Jason Parker <jparker@digium.com>
+
+ * sounds/Makefile, /: Merged revisions 280193 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r280193 | qwell | 2010-07-28 13:05:54 -0500 (Wed, 28 Jul 2010) |
+ 9 lines Remove unnecessary subshells. Attempt to make
+ checksumming work. Also improves readability. (issue #17621)
+ Reported by: bjm Review: https://reviewboard.asterisk.org/r/808/
+ ........
+
+2010-07-28 16:52 +0000 [r280161] Sean Bright <sean@malleable.com>
+
+ * apps/app_queue.c, /: Merged revisions 280160 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r280160 | seanbright | 2010-07-28 12:51:11 -0400 (Wed, 28 Jul
+ 2010) | 8 lines Plug a reference leak in app_queue when adding
+ members dynamically. (closes issue #17738) Reported by:
+ bobwienholt Patches: issue17738.patch uploaded by bobwienholt
+ (license 950) Tested by: bobwienholt, seanbright ........
+
+2010-07-28 13:52 +0000 [r280090] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/live_ast, /: Merged revisions 280089 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r280089 | lmadsen | 2010-07-28 08:51:16 -0500
+ (Wed, 28 Jul 2010) | 9 lines Merged revisions 280088 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r280088 | lmadsen | 2010-07-28 08:50:38 -0500 (Wed, 28
+ Jul 2010) | 1 line Update help text to be less confusing.
+ ........ ................
+
+2010-07-28 13:01 +0000 [r280058] Russell Bryant <russell@digium.com>
+
+ * res/res_crypto.c: s/init keys/keys init/
+
+2010-07-28 01:37 +0000 [r280023] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * channels/chan_usbradio.c: Resolve compiler warning about
+ formatting (closes issue #17732) Reported by: pabelanger
+
+2010-07-27 22:30 +0000 [r280019-280020] Sean Bright <sean@malleable.com>
+
+ * main/editline/el.h, main/term.c, main/cli.c,
+ main/editline/parse.c, main/editline/tokenizer.c,
+ main/editline/config.sub, main/editline/parse.h,
+ main/editline/tokenizer.h, configure, main/editline/histedit.h,
+ main/editline/sig.c, main/editline/PLATFORMS,
+ main/editline/sig.h, main/editline/key.c, main/editline/editrc.5,
+ main/editline/np/fgetln.c, main/editline/key.h,
+ main/editline/TEST/test.c, main/Makefile,
+ main/editline/configure, main/editline/Makefile.in, configure.ac,
+ main/editline/configure.in, main/editline/readline/readline.h,
+ main/editline/README, main/editline/editline.3,
+ main/editline/vi.c, main/editline/sys.h, main/editline/emacs.c,
+ main/asterisk.c, main/editline/install-sh, main/editline/term.c,
+ main/editline/config.guess, main/editline/read.c,
+ main/editline/term.h, main/editline/map.c,
+ main/editline/np/strlcpy.c, main/editline (added),
+ main/editline/config.h.in, main/editline/read.h,
+ main/editline/tty.c, main/editline/np/unvis.c,
+ main/editline/prompt.c, main/editline/map.h, main/editline/tty.h,
+ main/editline/chared.c, main/editline/prompt.h,
+ main/editline/np/strlcat.c, main/editline/chared.h,
+ main/editline/np, main/editline/TEST, main/editline/refresh.c,
+ main/editline/history.c, main/editline/readline,
+ include/asterisk/term.h, main/editline/refresh.h,
+ main/editline/search.c, main/editline/hist.c,
+ main/editline/search.h, main/editline/hist.h,
+ main/editline/np/vis.c, build_tools/menuselect-deps.in, main,
+ main/editline/readline.c, main/editline/np/vis.h,
+ main/editline/INSTALL, makeopts.in, main/editline/CHANGES,
+ main/editline/common.c, main/xmldoc.c, main/editline/makelist.in,
+ include/asterisk/autoconfig.h.in, main/editline/el.c: Revert
+ r280019 for now - This was poorly executed.
+
+ * include/asterisk/term.h, makeopts.in, main/asterisk.c,
+ main/xmldoc.c, main/cli.c, main/term.c, main/editline (removed),
+ build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, main/Makefile, configure.ac,
+ main: Add ability to use system libedit and update bundled
+ libedit. The version of libedit that is bundled with asterisk is
+ old and has some bugs. This patch updates the bundled version of
+ libedit within asterisk, and also updates asterisk to use the
+ system libedit instead if one is available (and pkg-config is
+ available). This review integrates several patches from other
+ users specifically kkm and tzafrir. (closes issue #15929)
+ Reported by: kkm Patches: 015929-astcli-editrc-trunk.240324.diff
+ uploaded by kkm (license 888) (issue #16858) Reported by:
+ jw-asterisk (closes issue #17039) Reported by: tzafrir Patches:
+ 0001-allow-using-system-copy-of-libedit.patch uploaded by tzafrir
+ (license 46) Review: https://reviewboard.asterisk.org/r/807/
+
+2010-07-27 21:16 +0000 [r279953] Russell Bryant <russell@digium.com>
+
+ * res/ais, main/db1-ast/mpool, Makefile.rules, res/snmp, cdr,
+ formats, codecs/gsm/src, funcs, bridges, codecs/lpc10,
+ main/db1-ast/btree, configure, main/editline, codecs/g722, main,
+ main/db1-ast/recno, channels/sip, makeopts.in, pbx, res, res/ael,
+ channels, main/stdtime, main/editline/np, codecs, utils,
+ main/db1-ast/hash, cel, apps, configure.ac, main/db1-ast/db: Add
+ --enable-coverage option to configure script. This option enables
+ the proper compiler flags for tracking code coverage, which is
+ useful along side automated testing.
+
+2010-07-27 20:57 +0000 [r279949] David Vossel <dvossel@digium.com>
+
+ * main/audiohook.c, main/channel.c, /,
+ include/asterisk/audiohook.h: Merged revisions 279946 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r279946 | dvossel | 2010-07-27 15:54:32 -0500
+ (Tue, 27 Jul 2010) | 24 lines Merged revisions 279945 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010)
+ | 19 lines remove empty audiohook write list on channel If a
+ channel has an audiohook write list created on it, that list
+ stays on the channel until the channel is destroyed. There is no
+ reason to keep that list on the channel if it becomes empty. If
+ it is empty that just means we are doing needless translating for
+ every ast_read and ast_write. This patch removes the audiohook
+ list from the channel once it is detected to be empty on either a
+ read or write. If a audiohook is added back to the channel after
+ this list is destroyed, the list just gets recreated as if it
+ never existed to begin with. (closes issue #17630) Reported by:
+ manvirr Review: https://reviewboard.asterisk.org/r/799/ ........
+ ................
+
+2010-07-27 19:50 +0000 [r279916] Russell Bryant <russell@digium.com>
+
+ * channels/sig_pri.c, channels/chan_dahdi.c: Fix inband DTMF
+ detection on outgoing ISDN calls. This is a regression from the
+ sig_pri split from chan_dahdi. When a call is first initiated,
+ the inband DTMF detector is not enabled if it's an outgoing ISDN
+ call. However, it needs to be turned on once the media path
+ starts up. This handling was put back in the open_media()
+ callback of chan_dahdi. In sig_pri, open_media() calls were added
+ to a few places where it was needed, including handling of
+ PRI_EVENT_RINGING, PRI_EVENT_PROGRESS, and PRI_EVENT_PROCEEDING.
+ Thanks to rmudgett for helping me with the patch!
+
+2010-07-27 18:54 +0000 [r279887] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix parsing error in sip_sipredirect(). The
+ code was written in a way that did a bad job of parsing the port
+ out of a URI. Specifically, it would do badly when dealing with
+ an IPv6 address. In this particular scenario, there was no value
+ from parsing the port out, so I just removed that logic. And
+ while I was messing around in the function, I changed some
+ variable names to be more descriptive. (closes issue #17661)
+ Reported by: oej Patches: 17661.diff uploaded by mmichelson
+ (license 60)
+
+2010-07-27 16:40 +0000 [r279850] Jason Parker <jparker@digium.com>
+
+ * sounds/Makefile, /: Merged revisions 279849 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r279849 | qwell | 2010-07-27 11:39:16 -0500 (Tue, 27 Jul 2010) |
+ 1 line Simply sounds/Makefile some more. ........
+
+2010-07-27 16:09 +0000 [r279817] David Vossel <dvossel@digium.com>
+
+ * main/netsock2.c, channels/chan_sip.c: fix sip transaction match
+ with authentication, fix confusing log message when using
+ getaddrinfo
+
+2010-07-27 16:06 +0000 [r279815] Russell Bryant <russell@digium.com>
+
+ * channels/chan_dahdi.c: Support "channels" in addition to
+ "channel" in chan_dahdi.conf. Review:
+ https://reviewboard.asterisk.org/r/804
+
+2010-07-27 15:15 +0000 [r279785] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 279784 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r279784 | mmichelson | 2010-07-27 10:13:24 -0500 (Tue, 27 Jul
+ 2010) | 14 lines Fix bad behavior of dynamic_exclude_static
+ option in sip.conf. We were attempting to create a contactdeny
+ rule based on the peer's IP address before the peer's IP address
+ had been set. By moving the processing further down in the
+ function, we can ensure stuff works as we expect for it to.
+ (closes issue #17717) Reported by: mmichelson Patches:
+ 17717.patch uploaded by mmichelson (license 60) Tested by:
+ DennisD ........
+
+2010-07-27 02:57 +0000 [r279726-279755] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * channels/chan_dahdi.c: If dringXcontext is null, fallback to
+ default context value. (closes issue #17693) Reported by:
+ iasgoscouk Patches: issue17693.patch uploaded by pabelanger
+ (license 224) Tested by: iasgoscouk Review:
+ https://reviewboard.asterisk.org/r/803/
+
+ * main/http.c: Use ast_sockaddr_setnull() when http is not enabled.
+ Otherwise, ast_tcptls_server_start() will still start http.
+ (closes issue #17708) Reported by: pabelanger Patches: http.patch
+ uploaded by pabelanger (license 224)
+
+2010-07-26 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.0-beta2 Released.
+
+2010-07-26 23:29 +0000 [r279689] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * UPGRADE.txt, CHANGES: Updated documentation for FAX logger level.
+
+2010-07-26 23:03 +0000 [r279658] Jason Parker <jparker@digium.com>
+
+ * sounds/Makefile (added), /, sounds/Makefile.380 (removed),
+ configure, include/asterisk/autoconfig.h.in, sounds/Makefile.381
+ (removed), configure.ac: Merged revisions 279657 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r279657 | qwell | 2010-07-26 17:59:52 -0500 (Mon, 26 Jul
+ 2010) | 5 lines Really fix sounds Makefile (and make it
+ readableish). There was a rather large syntax error that should
+ have caused ALL versions of GNU make to fail. I don't know how it
+ worked. ........
+
+2010-07-26 21:53 +0000 [r279636] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Ignore a control subclass of -1 in
+ ast_waitfordigit_full().
+
+2010-07-26 21:20 +0000 [r279599-279619] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 279609 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r279609 | tilghman | 2010-07-26 16:18:17 -0500 (Mon, 26
+ Jul 2010) | 2 lines Dunno why this worked on my machine, but it
+ works better this way. ........
+
+ * res/res_config_ldap.c, /: Merged revisions 279597 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r279597 | ghenry | 2010-07-26 15:25:54 -0500 (Mon, 26
+ Jul 2010) | 13 lines Apply all patches in:
+ https://issues.asterisk.org/view.php?id=13573 (closes issue
+ #13573) Reported by: navkumar Patches:
+ res_config_ldap-category.diff uploaded by navkumar (license 580)
+ res_config_ldap.patch uploaded by bencer (license 961)
+ res_config_ldap uploaded by bencer (license 961) Tested by:
+ suretec ........
+
+ * /: Reverting property remove
+
+2010-07-26 20:58 +0000 [r279598] Gavin Henry <ghenry@suretecsystems.com>
+
+ * /: Merged revisions 279597 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/1.6.2
+ -----------------------------------------------------------------------
+ r279597 | ghenry | 2010-07-26 15:25:53 -0500 (Mon, 26 Jul 2010) |
+ 13 lines Apply all patches in:
+ https://issues.asterisk.org/view.php?id=13573 [^] (closes issue
+ 0013573) Reported by: navkumar Patches:
+ res_config_ldap-category.diff uploaded by navkumar (license 580)
+ res_config_ldap.patch uploaded by bencer (license 961)
+ res_config_ldap uploaded by bencer (license 961) Tested by:
+ suretec
+ ------------------------------------------------------------------------
+
+2010-07-26 19:59 +0000 [r279568] David Vossel <dvossel@digium.com>
+
+ * channels/sip/include/sip.h,
+ channels/sip/include/reqresp_parser.h, channels/chan_sip.c,
+ channels/sip/reqresp_parser.c: transaction matching using top
+ most Via header This patch modifies the way chan_sip.c does
+ transaction to dialog matching. Asterisk now stores information
+ in the top most Via header of the initial incoming request and
+ compares that against other Requests that have the same call-id.
+ This results in Asterisk being able to detect a forked call in
+ which it has received multiple legs of the fork. I completely
+ stripped out the previous matching code and made the comparisons
+ a little more explicit and easier to understand. My comments in
+ the code should offer all the details involving this patch. This
+ patch also fixes a bug with the usage of the OBJ-MULTIPLE flag to
+ find multiple dialogs with the same call-id. Since the callback
+ function was returning (CMP_MATCH | CMP_STOP) only the first item
+ found was being returned. I fixed this by making a new callback
+ function for finding multiple dialogs that only returns
+ (CMP_MATCH) on a match allowing for multiple items to be
+ returned. Review: https://reviewboard.asterisk.org/r/776/
+
+2010-07-26 19:51 +0000 [r279566] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * UPGRADE.txt, CHANGES, configs/logger.conf.sample: Add
+ documentation for FAX logger level. (closes issue #17715)
+ Reported by: vrban Patches: 17715.patch uploaded by pabelanger
+ (license 224) Tested by: vrban
+
+2010-07-26 19:18 +0000 [r279562] Tilghman Lesher <tlesher@digium.com>
+
+ * sounds/Makefile (removed), /, sounds/Makefile.380 (added),
+ configure, include/asterisk/autoconfig.h.in, sounds/Makefile.381
+ (added), configure.ac: Merged revisions 279561 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r279561 | tilghman | 2010-07-26 14:15:59 -0500 (Mon, 26 Jul 2010)
+ | 2 lines Use a special Makefile for noobs who still have GNU
+ Make 3.80. ........
+
+2010-07-26 16:04 +0000 [r279504] Mark Michelson <mmichelson@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ channels/sip/reqresp_parser.c: Allow for systems without locale
+ support to be usable. A recent change to SIP URI comparison code
+ added a locale-specific string comparison to the mix, and certain
+ systems do not support such functions. This fix allows for those
+ systems to still use Asterisk 1.8 (closes issue #17697) Reported
+ by: pprindeville Patches: asterisk-trunk-bugid17697.patch
+ uploaded by pprindeville (license 347) Tested by: mmichelson
+
+2010-07-26 15:43 +0000 [r279502] Sean Bright <sean@malleable.com>
+
+ * autoconf/ast_ext_lib.m4, /: Merged revisions 279501 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r279501 | seanbright | 2010-07-26 11:41:13 -0400 (Mon,
+ 26 Jul 2010) | 5 lines Expand the correct value within
+ AST_OPTION_ONLY. (closes issue #17703) Reported by: stuarth
+ ........
+
+2010-07-26 03:27 +0000 [r279472] Tilghman Lesher <tlesher@digium.com>
+
+ * formats/format_sln16.c, formats/format_wav_gsm.c,
+ formats/format_siren7.c, formats/format_ilbc.c,
+ formats/format_vox.c, formats/format_pcm.c,
+ formats/format_h263.c, formats/format_g723.c,
+ formats/format_h264.c, formats/format_g726.c,
+ formats/format_jpeg.c, formats/format_siren14.c,
+ formats/format_gsm.c, formats/format_g719.c,
+ formats/format_g729.c, formats/format_sln.c,
+ formats/format_wav.c, formats/format_ogg_vorbis.c: Formats need
+ to load before apps, because some apps call
+ ast_format_str_reduce() at load time.
+
+2010-07-25 21:26 +0000 [r279442] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * tests/test_func_file.c: Add trailing backslash to silence warning
+ message.
+
+2010-07-25 18:21 +0000 [r279390-279410] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_odbc.c: Don't re-register CDR module on reload. (closes
+ issue #17304) Reported by: jnemeth Patches:
+ 20100507__issue17304.diff.txt uploaded by tilghman (license 14)
+ Tested by: jnemeth
+
+ * main/logger.c: Don't assume qlog is open. (closes issue #17704)
+ Reported by: vrban Patches: issue17704.patch uploaded by
+ pabelanger (license 224) Tested by: vrban
+
+2010-07-24 23:58 +0000 [r279348] Bradley Latus <brad.latus@gmail.com>
+
+ * doc/asterisk.8: Minor update to man page
+
+2010-07-24 20:47 +0000 [r279273-279314] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * Makefile: Remove duplicate -c flag when using $(INSTALL) (closes
+ issue #17695) Reported by: pabelanger Patches: Makefile.diff
+ uploaded by pabelanger (license 224)
+
+ * include/asterisk/netsock2.h: Check if ast_sockaddr is NULL then
+ return. (closes issue #17677) Reported by: outcast Patches:
+ issue0017677.patch uploaded by pabelanger (license 224) Tested
+ by: elguero
+
+ * main/manager.c: Default sin_family to AF_INET for TCP / TLS
+ Bindaddress. Otherwise, 'manager show settings' will generate
+ errors if manager is not enabled.
+
+2010-07-23 22:20 +0000 [r279227] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_queue.c, apps/app_dial.c, /: Merged revisions 279207 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r279207 | rmudgett | 2010-07-23 17:11:23 -0500
+ (Fri, 23 Jul 2010) | 14 lines Merged revisions 279206 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010)
+ | 7 lines SIP promiscuous redirect could fail to dial the
+ redirect. The ast_channel was created with one variable to
+ ast_request() but the call to ast_call() that initiates the
+ outgoing call was using a different variable. The two variables
+ are not equivalent if the call_forward string included a channel
+ technology specifier. e.g., SIP/200 ........ ................
+
+2010-07-12 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.8.0-beta1 Released.
+
+2010-07-23 18:56 +0000 [r279113] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_odbc.c: Silly 64-bit compilers (who uses 64-bit anyway?)
+
+2010-07-23 18:23 +0000 [r279056-279094] Russell Bryant <russell@digium.com>
+
+ * /: fix up properties on 1.8 branch
+
+ * / (added): Create a branch for Asterisk 1.8.
+
+ ___ _ _ _ _ ___
+ / _ \ ___| |_ ___ _ __(_)___| | __ / | ( _ )
+ | |_| / __| __/ _ \ '__| / __| |/ / | | / _ \
+ | _ \__ \ || __/ | | \__ \ < | || (_) |
+ |_| |_|___/\__\___|_| |_|___/_|\_\ |_(_)___/
+
+2010-07-23 17:05 +0000 [r278982-278985] Tilghman Lesher <tlesher@digium.com>
+
+ * autoconf/ast_check_pwlib.m4, /, configure, configure.ac: Merged
+ revisions 278984 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r278984 | tilghman | 2010-07-23 12:04:15 -0500 (Fri, 23 Jul 2010)
+ | 5 lines Establish a maximum version for openh323 (i.e. not
+ opal), because chan_h323 will fail to load, even if it links.
+ (issue #17679) Reported by: am ........
+
+ * /, main/asterisk.c: Merged revisions 278981 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r278981 | tilghman | 2010-07-23 11:42:25 -0500 (Fri, 23 Jul 2010)
+ | 8 lines Avoid race with consolethread on shutdown (on parallel
+ processors). (closes issue #17080) Reported by: sybasesql
+ Patches: 20100721__issue17080.diff.txt uploaded by tilghman
+ (license 14) Tested by: sybasesql ........
+
+2010-07-23 16:33 +0000 [r278980] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, channels/sip/reqresp_parser.c,
+ channels/sip/include/reqresp_parser.h: SIP URI comparison fixes.
+ This initially was created to work around the issue of using a
+ string comparison instead of a binary comparison for IP
+ addresses. It evolved a bit when test cases were created and it
+ was discovered that comparison of URI parameters was not working
+ exactly as it should. sip_uri_cmp() and its helpers have been
+ moved to reqresp_parser.c and a new test has been added. (closes
+ issue #17662) Reported by: oej Review:
+ https://reviewboard.asterisk.org/r/792
+
+2010-07-23 16:19 +0000 [r278957] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/res_odbc.h, res/res_config_odbc.c,
+ configs/extconfig.conf.sample, CHANGES, main/config.c,
+ res/res_odbc.c, configs/res_odbc.conf.sample: Merge the realtime
+ failover branch
+
+2010-07-23 16:07 +0000 [r278947] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * doc/asterisk.8: Some left-over hyphen-minus fixes in the man page
+
+2010-07-23 15:57 +0000 [r278944-278945] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: ... just kidding. Enable SIP by default. :-)
+
+ * channels/chan_sip.c: Disable SIP support by default for Asterisk
+ 1.8.
+
+2010-07-23 15:52 +0000 [r278943] Mark Michelson <mmichelson@digium.com>
+
+ * addons/chan_ooh323.c: Well, who knew chan_ooh323 used udptl? I
+ sure didn't!
+
+2010-07-23 15:41 +0000 [r278942] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
+ Rename sig_pri_pri to sig_pri_span. More descriptive of concept.
+
+2010-07-23 15:16 +0000 [r278908] Mark Michelson <mmichelson@digium.com>
+
+ * main/udptl.c, channels/chan_sip.c, include/asterisk/udptl.h,
+ channels/sip/include/sip.h: Allow IPv6 addresses for UDPTL
+ streams. Review: https://reviewboard.asterisk.org/r/795
+
+2010-07-23 13:37 +0000 [r278875] Olle Johansson <oej@edvina.net>
+
+ * res/res_config_ldap.c: Minor corrections to the LDAP realtime
+ driver Review: https://reviewboard.asterisk.org/r/798/ Thanks
+ Mark for a quick review!
+
+2010-07-23 13:26 +0000 [r278873] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * Makefile, agi/Makefile, sounds/Makefile: Portability updates for
+ Makefiles. When possible, use $(INSTALL). This allows us to use
+ the functionality within install for setting directory / file
+ permissions, a requirement for unprivileged installation. Also
+ move any directory we plan to create within the installdirs
+ macro. Plus various other formatting issues. (issue #17436)
+ Reported by: pabelanger Patches: non-root.patch.v8 uploaded by
+ pabelanger (license 224) Tested by: pabelanger Review:
+ https://reviewboard.asterisk.org/r/654/
+
+2010-07-23 11:01 +0000 [r278809-278841] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c: missed FXS kewl
+ start polarityswitch when finally on hook. (issue #17318)
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
+ channels/sig_analog.c, channels/sig_analog.h: Support FXS module
+ Polarity Reversal on remote party Answer and Hangup FXS lines
+ normally connect to a telephone. However, when FXS lines are
+ routed to an external PBX or Key System to act as "external" or
+ "CO" lines, it is extremely difficult, if not impossible for the
+ external PBX to know when the call has been disconnected without
+ receiving a polarity reversal on the line. Now using
+ answeronpolarityswitch and hanguponpolarityswitch keywords that
+ previously were used only for FXO ports, now applies like
+ functionality for an FXS port, but from the connected equipment's
+ point of view. (closes issue #17318) Reported by: armeniki
+ Patches: fxs_linepolarity.diff5.txt uploaded by alecdavis
+ (license 585) Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/797/
+
+2010-07-22 21:16 +0000 [r278777] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: DNID not cleared when channel hang up
+ (Affects PRI and SS7) The "dahdi show channels" CLI command still
+ reports the DNID of the previous call even if the call is already
+ hang up. The "dahdi show channels" command of older releases
+ clear the DNID once the channel is hang up. Regression from the
+ sig_analog/sig_pri extraction from chan_dahdi. (closes issue
+ #17623) Reported by: klaus3000 Patches: issue17623.patch uploaded
+ by rmudgett (license 664) Tested by: rmudgett
+
+2010-07-22 19:45 +0000 [r278708] Jeff Peeler <jpeeler@digium.com>
+
+ * main/xmldoc.c: Add method for finding XML doc files for systems
+ that don't support GLOB_BRACE. In particular, Solaris and perhaps
+ others do not support the above mentioned GNU extension. In this
+ case the paths are simply expanded without the braces and the
+ calls to glob are made separately. Note: I could not explain
+ memory allocation failures that were being reported from within
+ libxml itself when making calls to glob without using
+ GLOB_NOCHECK. This is the only reason why that flag is being
+ used. (closes issue #15402) Reported by: snuffy Patches:
+ bug_xmlpatt-v3.diff uploaded by snuffy (license 35), modified by
+ me
+
+2010-07-22 14:58 +0000 [r278620] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 278618 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r278618 | mmichelson | 2010-07-22 09:55:04 -0500 (Thu, 22 Jul
+ 2010) | 13 lines Allow PLC to function properly when channels use
+ SLIN for audio. If a channel involved in a bridge was using SLIN
+ audio, then translation paths were not guaranteed to be set up
+ properly since in all likelihood the number of translation steps
+ was only 1. This patch enforces the transcode_via_slin behavior
+ if transcode_via_slin or generic_plc is enabled and one of the
+ formats to make compatible is SLIN. AST-352 ........
+
+2010-07-22 14:56 +0000 [r278619] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: update sip subscription debug message to a
+ warning message If the Expire header of a SUBSCRIBE is less that
+ our expiremin, a log warning will be displayed.
+
+2010-07-22 05:29 +0000 [r278579] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/doxyref.h: Add the full current set of CDR
+ drivers
+
+2010-07-21 19:16 +0000 [r278539] David Vossel <dvossel@digium.com>
+
+ * tests/test_func_file.c: make func_file unit test's category
+ consistent with other tests
+
+2010-07-21 19:11 +0000 [r278538] Terry Wilson <twilson@digium.com>
+
+ * channels/iax2-parser.h, include/asterisk/crypto.h,
+ main/aescrypt.c (removed), include/asterisk/aes_internal.h
+ (removed), funcs/func_aes.c, res/res_crypto.c, main/aestab.c
+ (removed), main/aesopt.h (removed), include/asterisk/aes.h
+ (removed), main/aeskey.c (removed), pbx/pbx_dundi.c,
+ channels/chan_iax2.c, res/res_crypto.exports.in,
+ pbx/dundi-parser.h: Remove built-in AES code and use optional_api
+ instead Review: https://reviewboard.asterisk.org/r/793/
+
+2010-07-21 18:52 +0000 [r278536] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: send "423 Interval too small" Response to
+ Subscribe with Expires less that min allowed [RFC3265]3.1.6.1....
+ The notifier MAY also check that the duration in the "Expires"
+ header is not too small. If and only if the expiration interval
+ is greater than zero AND smaller than one hour AND less than a
+ notifier- configured minimum, the notifier MAY return a "423
+ Interval too small" error which contains a "Min-Expires" header
+ field. The "Min- Expires" header field is described in SIP [1].
+
+2010-07-21 17:44 +0000 [r278501] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c: Fix invalid test
+ for rxisoffhook in FXO channels This fixes some cases of no
+ outgoing calls on FXO before an incoming call. Remove an
+ unnecessary testing of an "off-hook" bit from DAHDI for FXO
+ (KS/GS) channels.In some cases the bit would not be initialized
+ properly before the first inbound call and thus prevent an
+ outgoing call. If those tests are actually required by anybody,
+ they should define DAHDI_CHECK_HOOKSTATE in channels/sig_analog.c
+ . (closes issue #14577) Reported by: jkroon Patches:
+ asterisk_chan_dahdi_hookstate_fix_trunk_new.diff uploaded by
+ frawd (license 610) Tested by: frawd Review:
+ https://reviewboard.asterisk.org/r/699/
+
+2010-07-21 16:15 +0000 [r278465] Russell Bryant <russell@digium.com>
+
+ * res/res_timing_pthread.c: Use poll() instead of select() in
+ res_timing_pthread to avoid stack corruption. This code did not
+ properly check FD_SETSIZE to ensure that it did not try to
+ select() on fds that were too large. Switching to poll() removes
+ the limitation on the maximum fd value. (closes issue #15915)
+ Reported by: keiron (closes issue #17187) Reported by: Eddie
+ Edwards (closes issue #16494) Reported by: Hubguru (closes issue
+ #15731) Reported by: flop (closes issue #12917) Reported by:
+ falves11 (closes issue #14920) Reported by: vrban (closes issue
+ #17199) Reported by: aleksey2000 (closes issue #15406) Reported
+ by: kowalma (closes issue #17438) Reported by: dcabot (closes
+ issue #17325) Reported by: glwgoes (closes issue #17118) Reported
+ by: erikje possibly other issues, too ...
+
+2010-07-21 15:56 +0000 [r278463] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c: Ensure realtime conferences are treated the
+ same as static conferences when trying to find an empty one.
+ Also, parse the useropts properly, when retrieving from realtime,
+ and add them to the existing flags. (closes issue #17502)
+ Reported by: kenji Patches: 20100720__issue17502.diff.txt
+ uploaded by tilghman (license 14) Tested by: kenji
+
+2010-07-21 15:54 +0000 [r278426-278462] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax_spandsp.c: Properly show the current page being
+ transfered for 'fax show session'
+
+ * channels/chan_sip.c: Properly set the port number for UDPTL media
+ sessions.
+
+ * res/res_fax.c: Don't print failure status when the remote end
+ hangs up, it may not be an actual failure.
+
+2010-07-21 13:02 +0000 [r278425] Russell Bryant <russell@digium.com>
+
+ * main/features.c, UPGRADE.txt, configs/features.conf.sample:
+ Update documentation for 'comebacktoorigin' in featuers.conf. The
+ documentation for this option did not match the code. Fix that
+ along with some minor cleanups to the code along the way.
+ Document a slight change in behavior (to something that was
+ previously undocumented) in UPGRADE.txt.
+
+2010-07-21 06:45 +0000 [r278393] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Change order so that it more closely
+ matches the related SIP command. (closes issue #17648) Reported
+ by: GMLudo Review: https://reviewboard.asterisk.org/r/789/
+
+2010-07-21 03:53 +0000 [r278361] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: include stat.h for everybody, needed for
+ device2chan
+
+2010-07-20 23:23 +0000 [r278275-278307] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_pgsql.c, main/logger.c, CHANGES,
+ contrib/realtime/mysql/queue_log.sql (added),
+ configs/logger.conf.sample: Separate queue_log arguments into
+ separate fields, and allow the text file to be used, even when
+ realtime is used. (closes issue #17082) Reported by: coolmig
+ Patches: 20100720__issue17082.diff.txt uploaded by tilghman
+ (license 14) Tested by: coolmig
+
+ * /, apps/app_voicemail.c: Merged revisions 278261 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r278261 | tilghman | 2010-07-20 17:23:13 -0500 (Tue, 20
+ Jul 2010) | 7 lines Delete IMAP messages in reverse order, to
+ ensure reordering after each expunge does not cause deletion of
+ the wrong message. (closes issue #16350) Reported by: noahisaac
+ Patches: 20100623__issue16350.diff.txt uploaded by tilghman
+ (license 14) ........
+
+2010-07-20 22:38 +0000 [r278274] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Reference correct struct member for unlikely
+ event PRI_EVENT_CONFIG_ERR.
+
+2010-07-20 22:26 +0000 [r278272] Tilghman Lesher <tlesher@digium.com>
+
+ * main/autoservice.c, /, main/features.c,
+ include/asterisk/channel.h: Merged revisions 278167 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20
+ Jul 2010) | 4 lines Do not queue up DTMF frames while a call is
+ on hold. (Fixes ABE-2110) ........
+
+2010-07-20 21:41 +0000 [r278234] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes sip CANCEL race condition If Asterisk
+ sends a 4xx error and the other side sends a CANCEl before
+ receiving the 4xx and responding with the ACK, Asterisk will
+ process the CANCEL and send a 487 Request Terminated as a new
+ final response to the INVITE. Since we are issuing a new final
+ response to the INVITE, the old one must be pretend_acked else it
+ will keep retransmitting.
+
+2010-07-20 21:01 +0000 [r278168] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: This commit contains several changes to the way
+ output channel variables are handled. FAX output channel
+ variables will now match the values reported by FAXOPT() and
+ should be set in all failure and success cases. This commit also
+ contains a few modifications to the way FAXOPT() variables are
+ populated in a few spots and fixes for some reference count leaks
+ of the session details structure in some failure cases. Also
+ found and fixed more cases where FAXOPT(status) may not have
+ gotten set. FAX-214 FAX-203
+
+2010-07-20 19:35 +0000 [r278132] Tilghman Lesher <tlesher@digium.com>
+
+ * cel/cel_pgsql.c, cdr/cdr_sqlite3_custom.c, channels/chan_local.c,
+ res/res_timing_dahdi.c, cdr/cdr_adaptive_odbc.c,
+ res/res_calendar_caldav.c, formats/format_sln16.c,
+ formats/format_wav_gsm.c, channels/chan_iax2.c, main/config.c,
+ main/loader.c, res/res_rtp_multicast.c, channels/chan_dahdi.c,
+ res/res_smdi.c, channels/chan_skinny.c,
+ include/asterisk/module.h, formats/format_pcm.c,
+ channels/chan_alsa.c, formats/format_h263.c, res/res_curl.c,
+ cdr/cdr_odbc.c, formats/format_jpeg.c, res/res_speech.c,
+ formats/format_gsm.c, cdr/cdr_manager.c, formats/format_g719.c,
+ res/res_calendar_exchange.c, cel/cel_tds.c, formats/format_wav.c,
+ channels/chan_bridge.c, channels/chan_agent.c,
+ formats/format_ogg_vorbis.c, res/res_monitor.c,
+ res/res_calendar_ews.c, res/res_config_curl.c,
+ channels/chan_misdn.c, funcs/func_curl.c,
+ res/res_timing_kqueue.c, formats/format_g726.c, main/asterisk.c,
+ res/res_odbc.c, cel/cel_adaptive_odbc.c, res/res_calendar.c,
+ cel/cel_radius.c, channels/chan_multicast_rtp.c,
+ apps/app_meetme.c, formats/format_sln.c, res/res_musiconhold.c,
+ channels/chan_gtalk.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c,
+ res/res_jabber.c, res/res_config_sqlite.c,
+ formats/format_siren7.c, cdr/cdr_csv.c, formats/format_ilbc.c,
+ res/res_config_odbc.c, cel/cel_manager.c, cel/cel_custom.c,
+ cdr/cdr_sqlite.c, res/res_agi.c, res/res_timing_timerfd.c,
+ apps/app_confbridge.c, formats/format_h264.c,
+ res/res_config_ldap.c, addons/chan_mobile.c,
+ formats/format_siren14.c, cdr/cdr_custom.c, channels/chan_mgcp.c,
+ res/res_rtp_asterisk.c, res/res_config_pgsql.c,
+ res/res_calendar_icalendar.c, channels/chan_sip.c,
+ cdr/cdr_syslog.c, res/res_fax.c, res/res_crypto.c,
+ res/res_adsi.c, include/asterisk/config.h, pbx/pbx_lua.c,
+ channels/chan_console.c, apps/app_queue.c, cdr/cdr_tds.c,
+ res/res_srtp.c, channels/chan_jingle.c, formats/format_vox.c,
+ res/res_timing_pthread.c, channels/chan_h323.c,
+ cel/cel_sqlite3_custom.c, formats/format_g723.c,
+ funcs/func_devstate.c, formats/format_g729.c,
+ addons/res_config_mysql.c: Add load priority order, such that
+ preload becomes unnecessary in most cases
+
+2010-07-20 18:11 +0000 [r278051-278096] Russell Bryant <russell@digium.com>
+
+ * contrib/scripts/install_prereq: Add a package to install_prereq.
+
+ * channels/chan_local.c: Only call ast_channel_cc_params_init() if
+ allocating a channel succeeds.
+
+2010-07-20 16:50 +0000 [r278024] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, /: Merged revisions 278023 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r278023 | tilghman | 2010-07-20 11:37:18 -0500 (Tue, 20 Jul 2010)
+ | 7 lines Off-by-one error (closes issue #16506) Reported by:
+ nik600 Patches: 20100629__issue16506.diff.txt uploaded by
+ tilghman (license 14) ........
+
+2010-07-19 21:07 +0000 [r277945] Jean Galarneau <jgalarneau@digium.com>
+
+ * /, main/features.c: Merged revisions 277906 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277906 | jeang | 2010-07-19 15:16:36 -0500 (Mon, 19 Jul 2010) |
+ 7 lines Avoid trying to pickup a parked extension before the park
+ operation is completed. A crash could occur if the extension is
+ picked up while the parking extension is being announced. Testing
+ pu->notquiteyet while searching for a parked extension resolves
+ this crash. (ABE-2418) ........
+
+2010-07-19 17:16 +0000 [r277872-277873] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample,
+ channels/sip/include/sip.h: Fix port setting of external address
+ in SIP. There are two changes here: 1. Since the externip setting
+ can now have a port attached to it, calling it "externip" is
+ misleading. The option is now documented and parsed as
+ "externaddr." This also extends to the "matchexterniplocally"
+ setting. It is now documented and parsed as
+ "matchexternaddrlocally." The old names for the options may still
+ be used, but they are no longer used in the sip.conf.sample file.
+ 2. If no port is set for the externaddr, and UDP is the transport
+ to be used, then we will set the port of the externaddr to that
+ of the udpbindaddr. This was how things worked prior to the IPv6
+ merge, so this is a regression fix. (closes issue #17665)
+ Reported by: mmichelson Patches: 17665.diff#2 uploaded by
+ pprindeville (license 347) Tested by: pprindeville
+
+ * tests/test_acl.c: Remove the fe80:1234::1234 test case from
+ test_acl.c The ACL test was failing on Mac OS X because it would
+ convert the above invalid link-local address into fe80::1234
+ while reporting no error from getaddrinfo(). Linux does not do
+ this.
+
+2010-07-19 14:39 +0000 [r277837] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h: Fix regression with distinctive ring
+ detection. The issue here is that passing an array to a function
+ prohibits the ARRAY_LEN macro from returning the real size. To
+ avoid this the size is now defined and use of ARRAY_LEN is
+ avoided. (closes issue #15718) Reported by: alecdavis Patches:
+ bug15718.patch uploaded by jpeeler (license 325)
+
+2010-07-19 14:17 +0000 [r277814] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/acl.h, main/netsock2.c, main/manager.c,
+ channels/chan_sip.c, channels/chan_skinny.c, tests/test_acl.c,
+ main/acl.c, include/asterisk/netsock2.h, configs/sip.conf.sample,
+ channels/chan_iax2.c: Make ACLs IPv6-capable. ACLs can now be
+ configured to match IPv6 networks. This is only relevant for ACLs
+ in chan_sip for now since other channel drivers do not support
+ IPv6 addressing. However, once those channel drivers are
+ outfitted to support IPv6 addressing, the ACLs will already be
+ ready for IPv6 support. https://reviewboard.asterisk.org/r/791
+
+2010-07-17 17:42 +0000 [r277773-277775] Tilghman Lesher <tlesher@digium.com>
+
+ * /, autoconf/ast_func_fork.m4, configure,
+ include/asterisk/autoconfig.h.in: Merged revisions 277738 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277738 | tilghman | 2010-07-17 11:59:11 -0500 (Sat, 17 Jul 2010)
+ | 5 lines Remove uclibc cross-compile triplet, as uclibc has a
+ working fork()... it's only uclinux that does not. (closes issue
+ #17616) Reported by: pprindeville ........
+
+ * res/res_config_pgsql.c, res/res_config_odbc.c, /,
+ include/asterisk/config.h, main/config.c,
+ addons/res_config_mysql.c: Merged revisions 277568 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16
+ Jul 2010) | 8 lines Since we split values at the semicolon, we
+ should store values with a semicolon as an encoded value. (closes
+ issue #17369) Reported by: gkservice Patches:
+ 20100625__issue17369.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman ........
+
+2010-07-17 13:10 +0000 [r277703] Russell Bryant <russell@digium.com>
+
+ * Makefile, configure, include/asterisk/autoconfig.h.in,
+ configure.ac, makeopts.in: Allow xmllint to be used for XML docs
+ validation. xmllint seems to be more commonly available since it
+ comes with libxml2.
+
+2010-07-17 00:03 +0000 [r277667] Bradley Latus <brad.latus@gmail.com>
+
+ * res/res_fax.c: Update res_fax.c to be a good xml citizen. (closes
+ issues #17667) Reported by: snuffy
+
+2010-07-16 23:23 +0000 [r277657] Tim Ringenbach <tim.ringenbach@gmail.com>
+
+ * main/features.c: Merged revisions 277625 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277625 | tringenbach | 2010-07-16 17:43:39 -0500 (Fri, 16 Jul
+ 2010) | 9 lines Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on
+ attended transfer. ast_bridge_call() clears
+ AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended transfer,
+ ast_bridge_call() is called for a second bridge on the same
+ channel, and it clears that flag, which still needs to get set
+ for when the original ast_bridge_call() gets control back and
+ checks it. Review: https://reviewboard.asterisk.org/r/741
+ ........
+
+2010-07-16 21:24 +0000 [r277530] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 277497 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277497 | mnicholson | 2010-07-16 16:18:38 -0500 (Fri, 16 Jul
+ 2010) | 4 lines Default to no udptl error correction so that
+ error correction will be disabled in the event that the remote
+ end indicates that they do not support the error correction mode
+ we requested. FAX-128 ........
+
+2010-07-16 21:16 +0000 [r277488] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_queue.c: Fix reporting estimated queue hold time. Just
+ say the number of seconds (after minutes) rather than doing some
+ incorrect calculation with respect to minutes. (closes issue
+ #17498) Reported by: corruptor Patches: holdesecs_bug.diff
+ uploaded by corruptor (license 253)
+
+2010-07-16 20:35 +0000 [r277484] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/sched.h, main/sched.c: Finally, a method that
+ really fixes the assertions in chan_iax2.c related to cancelling
+ lagid. No, replacing usleep(1) with sched_yield() did not have an
+ effect.
+
+2010-07-16 20:27 +0000 [r277467] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 277419 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r277419 | rmudgett | 2010-07-16 15:18:54 -0500 (Fri, 16
+ Jul 2010) | 15 lines priexclusive in chan_dahdi.conf ignored when
+ reloading dahdi module During a reload, the priexclusive and
+ outsignalling parameters are not read in from the config file as
+ intended. Unfortunately, they get set to defaults as a result.
+ This patch makes sure that they do not get set to defaults during
+ a reload. (closes issue #17441) Reported by: mtryfoss Patches:
+ issue17441_v1.4.patch uploaded by rmudgett (license 664)
+ issue17441_v1.6.2.patch uploaded by rmudgett (license 664)
+ issue17441_trunk.patch uploaded by rmudgett (license 664) Tested
+ by: rmudgett ........
+
+2010-07-16 20:25 +0000 [r277452] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c, contrib/realtime/mysql/musiconhold.sql
+ (added): Add documentation for MOH realtime fields
+
+2010-07-16 19:32 +0000 [r277409] Matthew Nicholson <mnicholson@digium.com>
+
+ * tests/test_devicestate.c: updated devicestate test for device
+ state changes
+
+2010-07-16 19:22 +0000 [r277366] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_queue.c: Add missing handling for ringing state for use
+ with queue empty options. (closes issue #17471) Reported by:
+ jazzy Patches: app_queue.c.diff uploaded by jazzy (license 1056)
+
+2010-07-16 18:31 +0000 [r277331] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 277327 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277327 | mnicholson | 2010-07-16 13:30:22 -0500 (Fri, 16 Jul
+ 2010) | 8 lines Interpret device state AST_DEVICE_UNKNOWN as
+ extension state AST_EXTENSION_NOT_INUSE. (closes issue #16035)
+ Reported by: francesco_r Patches: pbx.c.patch uploaded by
+ viniciusfontes (license 978) Tested by: francesco_r, agx, lawbar
+ ........
+
+2010-07-16 18:14 +0000 [r277263] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, /: Merged revisions 277261 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277261 | tilghman | 2010-07-16 13:04:11 -0500 (Fri, 16 Jul 2010)
+ | 5 lines If variable gotten is not set, will segfault on
+ Solaris. (closes issue #17636) Reported by: bklang ........
+
+2010-07-16 18:05 +0000 [r277250-277262] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c: Print f->subclass.integer instead of f->subclass.
+ (fix build breakage introduced in r277250)
+
+ * main/channel.c, /: Merged revisions 277247 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277247 | mnicholson | 2010-07-16 12:29:57 -0500 (Fri, 16 Jul
+ 2010) | 4 lines For pass through DTMF tones, measure the actual
+ duration between the begin and end packets on the wire. If it is
+ detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf
+ emulation. AST-362 ........
+
+2010-07-16 17:13 +0000 [r277183] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, apps/app_amd.c: Merged revisions 277182 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277182 | pabelanger | 2010-07-16 13:10:36 -0400 (Fri, 16 Jul
+ 2010) | 8 lines Total analysis time error with SIP and silence
+ suppression When using app_amd with SIP providers that have
+ silence suppression on, the iTotalTime count increases
+ exponentially. (closes issue #17656) Reported by: juls ........
+
+2010-07-16 16:25 +0000 [r277175] Mark Michelson <mmichelson@digium.com>
+
+ * channels/sip/reqresp_parser.c: Fix up some weird indentation
+ problems in reqresp_parser.c
+
+2010-07-16 15:20 +0000 [r277143] Sean Bright <sean@malleable.com>
+
+ * main/translate.c: Avoid crashing when installing a duplicate
+ translation path with a lower cost. (closes issue #17092)
+ Reported by: moy Patches: translate.rev254273.patch uploaded by
+ moy (license 222) Tested by: moy
+
+2010-07-16 13:40 +0000 [r277103] Eliel C. Sardanons <eliels@gmail.com>
+
+ * CREDITS: Add Despegar.com (my main sponsor) to the CREDITS file.
+
+2010-07-16 13:32 +0000 [r276950-277102] Olle Johansson <oej@edvina.net>
+
+ * main/dnsmgr.c, main/srv.c: Formatting changes
+
+ * channels/chan_sip.c: Formatting fixes
+
+ * configs/sip.conf.sample: Clarify syntax changes
+
+ * CREDITS: Adding a few more to the list of CREDITS
+
+ * channels/chan_sip.c: Formatting changes (guideline corrections)
+ Found a unused bag of curly brackets under my table. I always
+ wondered where they had gone. They where indeed needed in
+ chan_sip.c
+
+ * CREDITS: Adding a few more credits
+
+ * channels/chan_sip.c, doc/tex/channelvariables.tex,
+ configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h: Add
+ ability to configure the Max-Forwards header in the dialplan, as
+ well as in sip.conf configuration for the channel and for
+ devices. The Max-Forwards header is used to prevent loops in a
+ SIP network. Each intermediary, like SIP proxys and SBCs,
+ decrement this counter and detects when it reaches zero, at which
+ point the SIP request is nicely killed in a SIP-friendly way.
+ Review: https://reviewboard.asterisk.org/r/778/ Thanks to dvossel
+ for the review and good advice.
+
+ * CHANGES, apps/app_queue.c: Add a dialplan function to check if a
+ queue exists: QUEUE_EXISTS Review:
+ https://reviewboard.asterisk.org/r/777/
+
+2010-07-16 06:04 +0000 [r276910-276911] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_jabber.c: And yet one more
+
+ * res/res_jabber.c: "Item may be used uninitialized in this
+ function."
+
+2010-07-16 05:42 +0000 [r276909] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix reversed logic of if statement. Found
+ based on message from Philip Prindeville on the Asterisk
+ Developers mailing list.
+
+2010-07-16 05:38 +0000 [r276830-276908] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, configure.ac: Detect the --dynamic-list flag a bit
+ better
+
+ * configure, main/Makefile, configure.ac, makeopts.in: Fix build on
+ FreeBSD
+
+ * tests/test_utils.c: Fix trunk build for Mac OS X 10.6
+
+ * contrib/realtime/mysql/iaxfriends.sql,
+ contrib/realtime/mysql/meetme.sql,
+ contrib/realtime/postgresql/realtime.sql,
+ contrib/realtime/mysql/sipfriends.sql: Allow ipaddress to contain
+ the maximum IPv6 address. Also, update meetme to the full list of
+ supported fields.
+
+ * configure, autoconf/ast_gcc_attribute.m4: Quote AC_SUBST within
+ m4_ifval, so it does not get prematurely expanded. (closes issue
+ #17654) Reported by: pprindeville Patches: issue17654.diff
+ uploaded by qwell (license 4) Tested by: qwell, pprindeville
+
+2010-07-15 20:21 +0000 [r276788] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_sip.c: Correct not setting the bindport before
+ attempting to open the socket. Related to changes from 276571, I
+ was accidentally testing with a port set in my configuration
+ causing me to miss this. Also moved the TCP handling as well to
+ occur before build_peer is called.
+
+2010-07-15 19:46 +0000 [r276731-276769] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in,
+ include/asterisk/compat.h, configure.ac: Define LLONG_MAX on
+ systems that do not have it. (closes issue #17644) Reported by:
+ pprindeville
+
+ * configure, main/Makefile, autoconf/ast_gcc_attribute.m4,
+ configure.ac, makeopts.in: Fix linking asterisk on CentOS 5,
+ which is using gcc 4.1.1. Gcc 4.1.2 has the real fix. Review:
+ https://reviewboard.asterisk.org/r/790/
+
+2010-07-15 13:51 +0000 [r276653] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 276652 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r276652 | jpeeler | 2010-07-15 08:48:58 -0500 (Thu, 15 Jul 2010)
+ | 2 lines In a perfect world, the frame source would never be
+ NULL. In the meantime, don't crash when it is. ........
+
+2010-07-15 12:21 +0000 [r276616] Russell Bryant <russell@digium.com>
+
+ * contrib/scripts/install_prereq: Add lua5.1 to the handy dandy
+ list of packages.
+
+2010-07-14 22:58 +0000 [r276571] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_sip.c: Fix MWI notification transmission problems
+ over SIP. MWI updates were not being sent if no messages were
+ found in the event cache. This was corrected since a phone may
+ need to clear its MWI status configured previously from another
+ mailbox. Upon module or sip reload, MWI updates could not be sent
+ due to the sipsock socket not being set early enough in
+ reload_config. The code handling the descriptor assignment and
+ such has simply been moved before the call to build_peer. Issuing
+ a sip reload cleared the IP address of the peer, but skipped
+ checking the database for registration information. The database
+ is now checked both for sip reload and actually reloading the
+ module. If a transmission occurs before the do_monitor thread has
+ started, do not attempt to send a signal to it. (closes issue
+ #17398) Reported by: ip-rob
+
+2010-07-14 22:32 +0000 [r276570] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_rtp_asterisk.c, main/dnsmgr.c, channels/chan_sip.c,
+ main/acl.c: Fix errors where incorrect address information was
+ printed. ast_sockaddr_stringiy_fmt (which is call by all
+ ast_sockaddr_stringify* functions) uses thread-local storage for
+ storing the string that it creates. In cases where
+ ast_sockaddr_stringify_fmt was being called twice within the same
+ statement, the result of one call would be overwritten by the
+ result of the other call. This usually was happening in
+ printf-like statements and was resulting in the same stringified
+ addressed being printed twice instead of two separate addresses.
+ I have fixed this by using ast_strdupa on the result of stringify
+ functions if they are used twice within the same statement. As
+ far as I could tell, there were no instances where a pointer to
+ the result of such a call were saved anywhere, so this is the
+ only situation I could see where this error could occur.
+
+2010-07-14 21:29 +0000 [r276531] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_h323.c: Make compile again.
+
+2010-07-14 21:11 +0000 [r276490-276493] Tilghman Lesher <tlesher@digium.com>
+
+ * main/loader.c: Oops, merge reverted this fix.
+
+ * include/asterisk/adsi.h, include/asterisk/agi.h,
+ include/asterisk/crypto.h, main/asterisk.dynamics, main/Makefile,
+ tests/test_utils.c, main/adsistub.c (removed), main/cryptostub.c
+ (removed), res/res_adsi.c, res/res_crypto.c,
+ res/res_crypto.exports.in (added), res/res_adsi.exports.in,
+ main/loader.c, include/asterisk/optional_api.h: Remove the old
+ stub files, preferring the optional_api method. (closes issue
+ #17475) Reported by: tilghman Review:
+ https://reviewboard.asterisk.org/r/695/
+
+2010-07-14 20:15 +0000 [r276441] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/loader.c: Don't try to call an embedded module's
+ backup_globals() function until after confirming it exists.
+
+2010-07-14 19:51 +0000 [r276439] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: handle special case were "200 Ok" to pending
+ INVITE never receives ACK Unlike most responses, the 200 Ok to a
+ pending INVITE Request is acknowledged by an ACK Request. If the
+ ACK Request for this Response is not received the previous
+ behavior was to immediately destroy the dialog and hangup the
+ channel. Now in an effort to be more RFC compliant, instead of
+ immediately destroying the dialog during this special case,
+ termination is done with a BYE Request as the dialog is
+ technically confirmed when the 200 Ok is sent even if the ACK is
+ never received. The behavior of immediately hanging up the
+ channel remains. This only affects how dialog termination
+ proceeds for this one special case. RFC 3261 section 13.3.1.4 "If
+ the server retransmits the 2xx response for 64*T1 seconds without
+ receiving an ACK, the dialog is confirmed, but the session SHOULD
+ be terminated. This is accomplished with a BYE, as described in
+ Section 15."
+
+2010-07-14 16:58 +0000 [r276393] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_vpb.cc, channels/chan_sip.c,
+ include/asterisk/channel.h, channels/sig_pri.c,
+ channels/chan_iax2.c, main/cel.c, channels/chan_oss.c,
+ main/channel.c, main/cdr.c, channels/chan_jingle.c,
+ channels/chan_usbradio.c, channels/chan_dahdi.c,
+ channels/chan_phone.c, channels/sig_analog.c,
+ channels/chan_misdn.c, channels/chan_skinny.c,
+ channels/chan_h323.c, res/snmp/agent.c, apps/app_amd.c,
+ funcs/func_callerid.c, channels/sig_ss7.c, channels/chan_mgcp.c:
+ Expand the caller ANI field to an ast_party_id Expand the ani
+ field in ast_party_caller and ast_party_connected_line to an
+ ast_party_id. This is an extension to the ast_callerid
+ restructuring patch in review:
+ https://reviewboard.asterisk.org/r/702/ Review:
+ https://reviewboard.asterisk.org/r/744/
+
+2010-07-14 16:40 +0000 [r276392] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: collapse debug code in retrans_pkt into
+ separate lines I've been working in this function a bunch lately,
+ and these huge debug strings are getting annoying.
+
+2010-07-14 16:39 +0000 [r276391] Richard Mudgett <rmudgett@digium.com>
+
+ * res/snmp/agent.c: Make compile again.
+
+2010-07-14 16:36 +0000 [r276389] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_sip.c: Do not skip sending MWI for a peer if an
+ address is defined. Really just a merge mistake from IPv6
+
+2010-07-14 16:09 +0000 [r276349] Tim Ringenbach <tim.ringenbach@gmail.com>
+
+ * cel/cel_pgsql.c, doc/tex/celdriver.tex, doc/tex/cdrdriver.tex:
+ Fix documentation for pgsql cel and cdr, and slightly improve
+ pgsql_cel. Change the documented pgsql schema to use "timestamp"
+ instead of "time", as the latter is only a time without a date.
+ Added some missing columns for cel's pgsql schema, and corrected
+ spelling on some others. Updated cel's uniqueid size to be the
+ same as the cdr. Added id column to cel's pgsql schema and
+ updated code to allow unknown columns to get their default value
+ instead of forcing 0 or empty string. Added microseconds to the
+ timestamp cel logs to pgsql. Review:
+ https://reviewboard.asterisk.org/r/734
+
+2010-07-14 15:48 +0000 [r276347] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_local.c, addons/chan_ooh323.c,
+ apps/app_alarmreceiver.c, channels/chan_iax2.c, main/cli.c,
+ channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/chan_skinny.c, main/features.c, apps/app_dumpchan.c,
+ channels/sig_analog.h, apps/app_amd.c, channels/sig_ss7.c,
+ apps/app_dial.c, main/pbx.c, apps/app_privacy.c, apps/app_fax.c,
+ channels/chan_agent.c, apps/app_disa.c,
+ include/asterisk/channel.h, apps/app_talkdetect.c, main/cel.c,
+ funcs/func_redirecting.c (removed), channels/chan_misdn.c,
+ apps/app_macro.c, apps/app_zapateller.c, apps/app_voicemail.c,
+ channels/chan_unistim.c, tests/test_substitution.c,
+ channels/chan_vpb.cc, apps/app_meetme.c, main/ccss.c,
+ apps/app_readexten.c, channels/chan_gtalk.c, apps/app_followme.c,
+ include/asterisk/callerid.h, main/cdr.c, main/channel.c,
+ channels/chan_phone.c, main/dial.c, apps/app_setcallerid.c,
+ apps/app_osplookup.c, main/manager.c, apps/app_minivm.c,
+ res/res_agi.c, main/app.c, apps/app_rpt.c, channels/chan_mgcp.c,
+ apps/app_parkandannounce.c, apps/app_while.c,
+ funcs/func_dialplan.c, channels/chan_sip.c, UPGRADE.txt,
+ channels/chan_console.c, channels/sig_pri.c, apps/app_queue.c,
+ channels/chan_oss.c, channels/chan_usbradio.c,
+ channels/chan_jingle.c, funcs/func_blacklist.c,
+ apps/app_directed_pickup.c, main/file.c,
+ funcs/func_connectedline.c (removed), channels/chan_h323.c,
+ main/callerid.c, res/snmp/agent.c, apps/app_sms.c,
+ apps/app_stack.c, funcs/func_callerid.c: ast_callerid
+ restructuring The purpose of this patch is to eliminate struct
+ ast_callerid since it has turned into a miscellaneous collection
+ of various party information. Eliminate struct ast_callerid and
+ replace it with the following struct organization: struct
+ ast_party_name { char *str; int char_set; int presentation;
+ unsigned char valid; }; struct ast_party_number { char *str; int
+ plan; int presentation; unsigned char valid; }; struct
+ ast_party_subaddress { char *str; int type; unsigned char
+ odd_even_indicator; unsigned char valid; }; struct ast_party_id {
+ struct ast_party_name name; struct ast_party_number number;
+ struct ast_party_subaddress subaddress; char *tag; }; struct
+ ast_party_dialed { struct { char *str; int plan; } number; struct
+ ast_party_subaddress subaddress; int transit_network_select; };
+ struct ast_party_caller { struct ast_party_id id; char *ani; int
+ ani2; }; The new organization adds some new information as well.
+ * The party name and number now have their own presentation value
+ that can be manipulated independently. ISDN supplies the
+ presentation value for the name and number at different times
+ with the possibility that they could be different. * The party
+ name and number now have a valid flag. Before this change the
+ name or number string could be empty if the presentation were
+ restricted. Most channel drivers assume that the name or number
+ is then simply not available instead of indicating that the name
+ or number was restricted. * The party name now has a character
+ set value. SIP and Q.SIG have the ability to indicate what
+ character set a name string is using so it could be presented
+ properly. * The dialed party now has a numbering plan value that
+ could be useful to have available. The various channel drivers
+ will need to be updated to support the new core features as
+ needed. They have simply been converted to supply current
+ functionality at this time. The following items of note were
+ either corrected or enhanced: * The CONNECTEDLINE() and
+ REDIRECTING() dialplan functions were consolidated into
+ func_callerid.c to share party id handling code. * CALLERPRES()
+ is now deprecated because the name and number have their own
+ presentation values. * Fixed app_alarmreceiver.c
+ write_metadata(). The workstring[] could contain garbage. It also
+ can only contain the caller id number so using
+ ast_callerid_parse() on it is silly. There was also a typo in the
+ CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse()
+ on the channel's caller id number string. ast_callerid_parse()
+ alters the given buffer which in this case is the channel's
+ caller id number string. Then using ast_shrink_phone_number()
+ could alter it even more. * Fixed caller ID name and number
+ memory leak in chan_usbradio.c. * Fixed uninitialized char arrays
+ cid_num[] and cid_name[] in sig_analog.c. * Protected access to a
+ caller channel with lock in chan_sip.c. * Clarified intent of
+ code in app_meetme.c sla_ring_station() and dial_trunk(). Also
+ made save all caller ID data instead of just the name and number
+ strings. * Simplified cdr.c set_one_cid(). It hand coded the
+ ast_callerid_merge() function. * Corrected some weirdness with
+ app_privacy.c's use of caller presentation. Review:
+ https://reviewboard.asterisk.org/r/702/
+
+2010-07-14 11:51 +0000 [r276268] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/voicemail.conf.sample: Merged revisions 276267 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r276267 | lmadsen | 2010-07-14 06:49:01 -0500 (Wed, 14 Jul 2010)
+ | 1 line Update documentation for voicemail.conf externpass
+ option. ........
+
+2010-07-13 22:18 +0000 [r276219] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c, channels/sip/include/sip.h: chan_sip: RFC
+ compliant retransmission timeout Retransmission of packets should
+ not be based on how many packets were sent, but instead on a
+ timeout period. Depending on whether or not the packet is for a
+ INVITE or NON-INVITE transaction, the number of packets sent
+ during the retransmission timeout period will be different, so
+ timing out based on the number of packets sent is not accurate.
+ This patch fixes this by removing the retransmit limit and only
+ stopping retransmission after a timeout period is reached. By
+ default this timeout period is 64*(Timer T1) for both INVITE and
+ non-INVITE transactions. For more information on sip timer values
+ refer to RFC3261 Appendix A. Review:
+ https://reviewboard.asterisk.org/r/749/
+
+2010-07-13 21:42 +0000 [r276206] Terry Wilson <twilson@digium.com>
+
+ * channels/sip/include/dialog.h, channels/chan_sip.c: Revert early
+ destruction of RTP sessions Some code improperly assumes that the
+ sessions are still there, so revert the change until I can find
+ all of them and fix them.
+
+2010-07-13 19:15 +0000 [r276124-276127] Russell Bryant <russell@digium.com>
+
+ * /: Recorded merge of revisions 276126 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r276126 | russell | 2010-07-13 14:14:54 -0500 (Tue, 13 Jul 2010)
+ | 2 lines Only reset a CDR that exists. ........
+
+ * /, main/features.c: Merged revisions 276123 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r276123 | russell | 2010-07-13 14:06:53 -0500 (Tue, 13 Jul 2010)
+ | 2 lines Use chan->cdr instead of chan_cdr (just like peer->cdr
+ instead of peer_cdr in the last commit). ........
+
+2010-07-13 19:05 +0000 [r276114-276122] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_env.c: Oops, XML documentation fix.
+
+ * funcs/func_env.c: It really cannot fail in the places below, but
+ the stupid compiler doesn't know that.
+
+ * funcs/func_env.c: Weird compiler error on Bamboo.
+
+ * funcs/func_env.c, CHANGES, tests/test_func_file.c (added): FILE()
+ now supports line-mode and writing (altering) files. (closes
+ issue #16461) Reported by: skyman Patches:
+ 20100622__issue16461.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman Review:
+ https://reviewboard.asterisk.org/r/737/
+
+2010-07-13 17:37 +0000 [r276074] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 275773 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275773 | jpeeler | 2010-07-12 15:34:51 -0500 (Mon, 12 Jul 2010)
+ | 12 lines Make user removals and traversals thread safe in
+ meetme. Race conditions present in meetme involving the user list
+ where a lack of locking has the potential for a user to be
+ removed during a traversal or as in the case of the reporter
+ after checking if the list is empty could cause a crash. Fixing
+ this was done by convering the userlist to an ao2 container.
+ (closes issue #17390) Reported by: Vince Review:
+ https://reviewboard.asterisk.org/r/746/ ........
+
+2010-07-13 17:11 +0000 [r275998] Terry Wilson <twilson@digium.com>
+
+ * channels/sip/include/dialog.h, channels/chan_sip.c: Destroy RTP
+ fds when we schedule final dialog destruction Since we are only
+ keeping the dialog around for retransmissions at this point and
+ there is no possibility that we are still handling RTP, go ahead
+ and destroy the RTP sessions. Keeping them alive for 32 past when
+ they are used is unnecessary and can lead to problems with having
+ too many open file descriptors, etc.
+
+2010-07-13 16:53 +0000 [r275995] Russell Bryant <russell@digium.com>
+
+ * /, main/features.c: Merged revisions 275994 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275994 | russell | 2010-07-13 11:51:18 -0500 (Tue, 13 Jul 2010)
+ | 14 lines Access peer->cdr directly instead of through a saved
+ off reference. At this point in the code, it is possible that
+ peer_cdr may be invalid. Specifically, in the blind transfer
+ code, CDRs are swapped between channels. So, peer_cdr is no
+ longer == peer->cdr. The scenario that exposed a crash in this
+ code was a blind transfer that hit the system call limit, causing
+ the transferee channel to get destroyed after the transfer
+ attempt failed. Even if it succeeds and this code doesn't crash,
+ this code was still trying to reset a CDR on a channel that was
+ now owned by a different thread, which is a BadThing(tm).
+ (ABE-2417) ........
+
+2010-07-13 14:48 +0000 [r275910] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/realtime_pgsql.sql (removed),
+ contrib/scripts/iax-friends.sql (removed), /,
+ contrib/realtime/mysql/iaxfriends.sql, contrib/scripts/meetme.sql
+ (removed), contrib/realtime (added), contrib/realtime/postgresql,
+ contrib/realtime/postgresql/realtime.sql, contrib/realtime/mysql,
+ contrib/realtime/oracle, contrib/scripts/sip-friends.sql
+ (removed), contrib/realtime/mysql/sipfriends.sql,
+ contrib/realtime/mysql/voicemail.sql, contrib/scripts/vmdb.sql
+ (removed), contrib/realtime/mysql/meetme.sql,
+ contrib/realtime/sqlserver: Merged revisions 275909 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r275909 | tilghman | 2010-07-13 09:47:30 -0500 (Tue, 13
+ Jul 2010) | 2 lines Move SQL scripts into their own
+ database-specific directories. ........
+
+2010-07-13 11:41 +0000 [r275863] Russell Bryant <russell@digium.com>
+
+ * configs/voicemail.conf.sample,
+ contrib/scripts/voicemailpwcheck.py (added): Add example script
+ for use with the externpasscheck voicemail.conf option. (closes
+ issue #17628) Reported by: lmadsen Tested by: russell, lmadsen
+ Review: https://reviewboard.asterisk.org/r/774/
+
+2010-07-12 23:27 +0000 [r275816] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Don't try to ref authpeer when it isn't set
+
+2010-07-12 17:54 +0000 [r275725] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Add which ITU spec specifies the numbering plan.
+
+2010-07-12 17:21 +0000 [r275682] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 275665 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275665 | jpeeler | 2010-07-12 11:58:39 -0500 (Mon, 12 Jul 2010)
+ | 11 lines Change ast_write to not stop generator when called
+ from ast_prod. For SIP channels configured with the
+ progressinband option on, the ringback was being immediately
+ stopped. This problem was due to ast_prod being moved for a
+ deadlock fix in 259858. Prodding the channel after setting up the
+ generator triggered the check in ast_write to stop the generator.
+ The fix here should write the frame the same as was done before
+ the call to ast_prod was moved. (closes issue #17372) Reported
+ by: tech_admin ........
+
+2010-07-12 15:37 +0000 [r275626] Leif Madsen <lmadsen@digium.com>
+
+ * cdr/cdr_pgsql.c: cdr_pgsql does not detect when a table is found.
+ This change adds an ERROR message to let you know when a failure
+ exists to get the columns from the pgsql database, which
+ typically means that the table does not exist. (closes issue
+ #17478) Reported by: kobaz Patches: cdr_pgsql.patch uploaded by
+ kobaz (license 834) Tested by: kobaz, russell, lmadsen
+
+2010-07-12 14:55 +0000 [r275587] Mark Michelson <mmichelson@digium.com>
+
+ * main/netsock2.c: Allow netsock2.c to compile on systems that do
+ not define AI_NUMERICSERV. (closes issue #17617) Reported by:
+ pprindeville Patches: asterisk-trunk-bugid17617.patch uploaded by
+ pprindeville (license 347)
+
+2010-07-12 04:16 +0000 [r275551] TransNexus OSP Development <support@transnexus.com>
+
+ * configs/osp.conf.sample, apps/app_osplookup.c: Added support for
+ indirect work mode.
+
+2010-07-10 20:49 +0000 [r275509] Eliel C. Sardanons <eliels@gmail.com>
+
+ * apps/app_meetme.c: When creating a conference for a unit test, it
+ is not mandatory to open a dahdi pseudo channel, so if we fail
+ doing it, continue creating the conference.
+
+2010-07-10 14:48 +0000 [r275424-275467] Russell Bryant <russell@digium.com>
+
+ * CHANGES: Make indentation consistent, move some queue features to
+ the queue section.
+
+ * CREDITS, channels/chan_unistim.c, configs/unistim.conf.sample,
+ CHANGES: Add support for devices with less than 3 lines on the
+ LCD. (closes issue #17600) Reported by: minaguib Patches:
+ ast_unistim_height_v2.patch uploaded by minaguib (license 1078)
+ Tested by: minaguib
+
+ * main/features.c, configs/features.conf.sample: Fix some issues
+ related to dynamic feature groups in features.conf. The bridge
+ handling code did not properly consider feature groups when
+ setting parameters that would affect whether or not a native
+ bridge would be attempted. If DYNAMIC_FEATURES only include a
+ feature group, a native bridge would occur that may prevent
+ features from working. Fix a bug in verbose output that would
+ show the key mapping as empty if it was using the default mapping
+ and not a custom mapping in the feature group. Add feature groups
+ to the output of "features show". Adjust the feature execution
+ logic to match that of the logic when executing a feature that
+ was not configured through a feature group. Update
+ features.conf.sample to show that an '=' is still required if
+ using the default key mapping from [applicationmap]. Finally,
+ clean up a little bit of formatting to better coform to coding
+ guidelines while in the area. (closes issue #17589) Reported by:
+ lmadsen Patches: issue_17589.rev4.txt uploaded by russell
+ (license 2) Tested by: russell, lmadsen
+
+2010-07-09 20:58 +0000 [r275385] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix error in parsing SIP registry strings
+ from ASTdb. It was essentially an off-by-one error. The easiest
+ way to fix this was to use the handy-dandy
+ AST_NONSTANDARD_RAW_ARGS macro to parse the pieces of the
+ registration string out. Tested and it works wonderfully.
+
+2010-07-09 20:01 +0000 [r275312] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c, channels/chan_iax2.c: Get more information
+ about the Bamboo test failures
+
+2010-07-09 19:58 +0000 [r275309-275310] Russell Bryant <russell@digium.com>
+
+ * main/features.c: Add missing ao2_iterator_destroy().
+
+ * apps/app_voicemail.c: Fix compile error.
+
+2010-07-09 19:46 +0000 [r275308] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix port parsing in check_via. If a Via
+ header contained an IPv6 address, we would not properly parse the
+ port. We would instead get the information after the first colon
+ in the address. (closes issue #17614) Reported by: oej Patches:
+ diff uploaded by sperreault (license 252)
+
+2010-07-09 19:32 +0000 [r275307] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * CHANGES, apps/app_voicemail.c: Include rdnis in msgXXXX.txt file.
+ (closes issue #17566) Reported by: outcast Patches:
+ voicemail-rdnis.patch uploaded by outcast (license 1071) Tested
+ by: outcast
+
+2010-07-09 19:29 +0000 [r275294] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix an issue where the port for p->ourip was
+ being set to 0. This should fix all the CDR tests that were not
+ passing. When they would originate a call, all fields in the
+ INVITE that contained the source port would have the port set to
+ 0. Most troubling of these was the Contact header. Tests are
+ passing locally now and should also pass on the bamboo build
+ agents.
+
+2010-07-09 19:21 +0000 [r275249] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, channels/chan_sip.c: Merged revisions 275241 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275241 | pabelanger | 2010-07-09 15:20:00 -0400 (Fri, 09 Jul
+ 2010) | 8 lines Fix logging message for stale nonce. (closes
+ issue #17582) Reported by: kenner Patches: chan_sip.c.diff
+ uploaded by kenner (license 1040) Tested by: lmadsen ........
+
+2010-07-09 18:55 +0000 [r275227] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c, channels/chan_iax2.c: Weird, no output and
+ Bamboo still fails...
+
+2010-07-09 18:24 +0000 [r275186] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/loader.c: Merged revisions 275182 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275182 | mnicholson | 2010-07-09 13:23:23 -0500 (Fri, 09 Jul
+ 2010) | 2 lines give a better error message when attempting to
+ unload a module that is not loaded ........
+
+2010-07-09 18:21 +0000 [r275172] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c, channels/chan_iax2.c: Add some diagnostic
+ feedback to our data tests
+
+2010-07-09 18:11 +0000 [r275147] Russell Bryant <russell@digium.com>
+
+ * configs/features.conf.sample: Move parking lot sample config out
+ from the middle of dynamic features sample config.
+
+2010-07-09 17:50 +0000 [r275144] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/loader.c: Merged revisions 275143 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275143 | mnicholson | 2010-07-09 12:50:05 -0500 (Fri, 09 Jul
+ 2010) | 2 lines don't unload modules that returned
+ AST_MODULE_LOAD_DECLINE when they were loaded ........
+
+2010-07-09 17:00 +0000 [r275105] Tilghman Lesher <tlesher@digium.com>
+
+ * main/netsock2.c, tests/test_substitution.c, tests/test_heap.c,
+ apps/app_meetme.c, tests/test_gosub.c, funcs/func_strings.c,
+ tests/test_event.c, channels/sip/reqresp_parser.c,
+ channels/chan_iax2.c, tests/test_stringfields.c,
+ tests/test_time.c, tests/test_devicestate.c, tests/test_utils.c,
+ main/features.c, res/res_agi.c, include/asterisk/netsock2.h,
+ tests/test_astobj2.c, channels/chan_sip.c,
+ tests/test_ast_format_str_reduce.c, tests/test_app.c,
+ funcs/func_math.c, include/asterisk/channel.h,
+ tests/test_sched.c, tests/test_pbx.c, tests/test_strings.c,
+ main/data.c, tests/test_skel.c, tests/test_acl.c,
+ channels/sip/dialplan_functions.c, tests/test_aoc.c, main/test.c,
+ channels/sip/config_parser.c, res/res_timing_kqueue.c,
+ apps/app_voicemail.c: Kill some startup warnings and errors and
+ make some messages more helpful in tracking down the source.
+
+2010-07-09 16:39 +0000 [r275104] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Return logic of sip_debug_test_addr() to its
+ original functionality.
+
+2010-07-09 16:05 +0000 [r275028] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 275027 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275027 | mnicholson | 2010-07-09 11:04:21 -0500 (Fri, 09 Jul
+ 2010) | 8 lines Clear the AST_CDR_FLAG_DIALED flag for channels
+ going into the pbx via the G option in app_dial (closes issue
+ #17592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff
+ uploaded by mnicholson (license 96) Tested by: jamicque,
+ mnicholson ........
+
+2010-07-09 15:35 +0000 [r275022] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/test.h, /, main/test.c: Merged revisions 275021
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r275021 | russell | 2010-07-09 10:33:08 -0500 (Fri, 09 Jul 2010)
+ | 4 lines Document that a leading and trailing slash is expected
+ for test categories. Also, emit a warning if a test is registered
+ without one of these. ........
+
+2010-07-09 14:27 +0000 [r274984] Mark Michelson <mmichelson@digium.com>
+
+ * channels/sip/reqresp_parser.c: Fix sip_uri_parse test comparison.
+ Part of the change with the IPv6 changes is to treat a host:port
+ as a single 'domain' entity. This test was not updated to have
+ the correct expectation after calling parse_uri().
+
+2010-07-09 13:30 +0000 [r274909-274947] <simon.perreault@viagenie.ca>
+
+ * channels/chan_sip.c: Copy the address into the peer structure
+ after we set the default port
+
+ * main/netsock2.c: Sadly we can't dereference a pointer cast and
+ use it as an lvalue without getting this warning (at least with
+ gcc 4.4.4): netsock2.c:492: warning: dereferencing pointer
+ ‘({anonymous})’ does break strict-aliasing rules So we're back to
+ using memcpy()...
+
+2010-07-09 12:48 +0000 [r274907] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/indications.h: Extend length limit on country
+ name in indications.conf.
+
+2010-07-09 11:06 +0000 [r274866] Olle Johansson <oej@edvina.net>
+
+ * configs/cdr.conf.sample, cdr/cdr_csv.c: Make it possible to
+ disable individual cdr files per accountcode in cdr_csv Review:
+ https://reviewboard.asterisk.org/r/678/
+
+2010-07-08 23:46 +0000 [r274827-274828] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_jingle.c, channels/chan_h323.c,
+ channels/chan_gtalk.c: Fix calls of ast_sockaddr_from_sin() from
+ IPv6 integration.
+
+ * addons/chan_ooh323.c: Fix compile of chan_ooh323.c from IPv6
+ integration.
+
+2010-07-08 22:16 +0000 [r274783-274786] Mark Michelson <mmichelson@digium.com>
+
+ * /: And the automerge property.
+
+ * /: Delete properties I merged during v6-new merge.
+
+ * channels/chan_unistim.c, include/asterisk/acl.h, main/netsock2.c
+ (added), channels/sip/include/dialog.h,
+ channels/chan_multicast_rtp.c, addons/chan_ooh323.c,
+ main/rtp_engine.c, /, channels/sip/reqresp_parser.c,
+ include/asterisk/tcptls.h, channels/chan_gtalk.c,
+ channels/chan_iax2.c, main/config.c, res/res_rtp_multicast.c,
+ main/manager.c, channels/chan_skinny.c,
+ channels/sip/include/globals.h, main/http.c, main/app.c,
+ include/asterisk/netsock2.h (added), apps/app_externalivr.c,
+ configs/sip.conf.sample, include/asterisk/rtp_engine.h,
+ channels/sip/include/sip.h, channels/chan_mgcp.c,
+ channels/sip/include/reqresp_parser.h, res/res_rtp_asterisk.c,
+ main/dnsmgr.c, channels/chan_sip.c, include/asterisk/config.h,
+ main/acl.c, CHANGES, channels/chan_jingle.c, main/tcptls.c,
+ channels/sip/dialplan_functions.c, channels/chan_h323.c,
+ include/asterisk/dnsmgr.h: Add IPv6 to Asterisk. This adds a
+ generic API for accommodating IPv6 and IPv4 addresses within
+ Asterisk. While many files have been updated to make use of the
+ API, chan_sip and the RTP code are the files which actually
+ support IPv6 addresses at the time of this commit. The way has
+ been paved for easier upgrading for other files in the near
+ future, though. Big thanks go to Simon Perrault, Marc Blanchet,
+ and Jean-Philippe Dionne for their hard work on this. (closes
+ issue #17565) Reported by: russell Patches:
+ asteriskv6-test-report.pdf uploaded by russell (license 2)
+ Review: https://reviewboard.asterisk.org/r/743
+
+2010-07-08 22:05 +0000 [r274773-274782] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Generate a correct AstData string for
+ ast_callerid.cid_ton
+
+ * main/channel.c: Fix trunk compile.
+
+2010-07-08 14:48 +0000 [r274727] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/pbx.c, channels/chan_sip.c, apps/app_meetme.c,
+ include/asterisk/indications.h, channels/chan_agent.c,
+ include/asterisk/channel.h, include/asterisk/cdr.h,
+ include/asterisk/data.h, channels/chan_iax2.c, apps/app_queue.c,
+ main/indications.c, main/channel.c, main/cdr.c,
+ channels/chan_dahdi.c, main/data.c, res/res_odbc.c,
+ apps/app_voicemail.c: Implement AstData API data providers as
+ part of the GSOC 2010 project, midterm evaluation. Review:
+ https://reviewboard.asterisk.org/r/757/
+
+2010-07-07 20:09 +0000 [r274686] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Fixes some ref count issues introduced by
+ r274539
+
+2010-07-07 18:32 +0000 [r274595-274639] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Add missing conditional around chan_dahdi
+ mfcr2_skip_category config parameter.
+
+ * channels/chan_dahdi.c, /: Merged revisions 274579 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r274579 | rmudgett | 2010-07-07 13:12:41 -0500 (Wed, 07
+ Jul 2010) | 1 line Close the DAHDI FD on error when processing
+ chan_dahdi toneduration config parameter. ........
+
+2010-07-07 16:40 +0000 [r274540] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: Set proper FAXOPT(status), FAXOPT(statusstr), and
+ FAXOPT(error) values where possible. Previously some failure
+ cases did not result in proper FAXOPT values. FAX-203
+
+2010-07-07 16:21 +0000 [r274539] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Use the relatedpeer field of a sip_pvt
+ during INVITE processing. Review:
+ https://reviewboard.asterisk.org/r/629
+
+2010-07-07 07:07 +0000 [r274492] TransNexus OSP Development <support@transnexus.com>
+
+ * configs/osp.conf.sample, doc/osp.txt: Changed OSP TCP port from
+ 1080 to 5045.
+
+2010-07-07 06:32 +0000 [r274418-274491] Tilghman Lesher <tlesher@digium.com>
+
+ * CHANGES, apps/app_voicemail.c: Also run the externnotify script
+ when the pollmailboxes thread notices a change.
+
+ * /, configs/say.conf.sample: Merged revisions 274417 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r274417 | tilghman | 2010-07-07 01:13:54 -0500 (Wed, 07
+ Jul 2010) | 8 lines Correct how 100, 200, 300, etc. is said. Also
+ add the crazy British numbers. (closes issue #16102) Reported by:
+ Delvar Patches: say.conf.fix.patch uploaded by Delvar (license
+ 908) (plus a few additional fixes and simplifications by me)
+ ........
+
+2010-07-06 22:23 +0000 [r274316] Jeff Peeler <jpeeler@digium.com>
+
+ * /, configs/sip.conf.sample: Merged revisions 274283 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r274283 | jpeeler | 2010-07-06 17:15:21 -0500 (Tue, 06
+ Jul 2010) | 7 lines Correct sip.conf.sample comments for
+ prematuremedia option. (closes issue #17513) Reported by: festr
+ Patches: patch uploaded by festr (license 443) ........
+
+2010-07-06 22:15 +0000 [r274284] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c, UPGRADE.txt: Merged revisions 274280 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010)
+ | 9 lines Add option to not do a call forward on 482 Loop
+ Detected Asterisk has always set up a forwarded call when
+ receiving a 482 Loop Detected. This prevents handling the call
+ failure by just continuing on in the dialplan. Since this would
+ be a change in behavior, the new option to disable this behavior
+ is forwardloopdetected which defaults to 'yes'. Review:
+ https://reviewboard.asterisk.org/r/764/ ........ (no option for
+ trunk, just changing the behavior)
+
+2010-07-06 22:09 +0000 [r274281] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c: Status shows all non-CRC4 lines as
+ "yellow", even if "yellow" was not in the bitfield.
+
+2010-07-06 19:53 +0000 [r274243] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: Properly detect and report invalid maxrate and
+ maxrate values in the FAXOPT dialplan function. Also make
+ fax_rate_str_to_int() return an unsigned int and return 0 instead
+ of -1 in the event of an error. FAX-202
+
+2010-07-06 14:31 +0000 [r274164] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_rtp_asterisk.c, /: Merged revisions 274157 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r274157 | mmichelson | 2010-07-06 09:29:23 -0500 (Tue,
+ 06 Jul 2010) | 16 lines Fix problem with RFC 2833 DTMF not being
+ accepted. A recent check was added to ensure that we did not
+ erroneously detect duplicate DTMF when we received packets out of
+ order. The problem was that the check did not account for the
+ fact that the seqno of an RTP stream will roll over back to 0
+ after hitting 65535. Now, we have a secondary check that will
+ ensure that the seqno rolling over will not cause us to stop
+ accepting DTMF. (closes issue #17571) Reported by: mdeneen
+ Patches: rtp_seqno_rollover.patch uploaded by mmichelson (license
+ 60) Tested by: richardf, maxochoa, JJCinAZ ........
+
+2010-07-06 06:01 +0000 [r274053] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Uh, yeah.
+
+2010-07-05 13:53 +0000 [r273886] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, main/config.c: Merged revisions 273884 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273884 | pabelanger | 2010-07-05 09:51:29 -0400 (Mon, 05 Jul
+ 2010) | 8 lines Remove extra line breaks from 'core show config
+ mappings' (closes issue #17583) Reported by: pabelanger Patches:
+ issue17583.patch uploaded by pabelanger (license 224) Tested by:
+ lmadsen ........
+
+2010-07-03 02:36 +0000 [r273714-273830] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /, channels/chan_agent.c,
+ channels/chan_h323.c, include/asterisk/lock.h: Merged revisions
+ 273793 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010)
+ | 9 lines Have the DEADLOCK_AVOIDANCE macro warn when an unlock
+ fails, to help catch potentially large software bugs. (closes
+ issue #17407) Reported by: pdf Patches:
+ 20100527__issue17407.diff.txt uploaded by tilghman (license 14)
+ Review: https://reviewboard.asterisk.org/r/751/ ........
+
+ * main/autoservice.c, /: Merged revisions 273717 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273717 | tilghman | 2010-07-02 12:09:47 -0500 (Fri, 02 Jul 2010)
+ | 8 lines Autoservice loop optimization causes a busy loop, when
+ channels are serviced while in hangup. (closes issue #17564)
+ Reported by: ramonpeek Patches: 20100630__issue17564.diff.txt
+ uploaded by tilghman (license 14) Tested by: ramonpeek ........
+
+ * apps/app_queue.c: The switch fallthrough could create some
+ errorneous situations, so best to force directly to the default
+ case.
+
+2010-07-02 15:57 +0000 [r273641] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_dahdi.c, channels/chan_misdn.c,
+ channels/chan_sip.c, main/say.c, main/fixedjitterbuf.c,
+ res/res_agi.c, channels/chan_h323.c, main/utils.c,
+ channels/chan_iax2.c, addons/chan_mobile.c, apps/app_rpt.c,
+ channels/chan_mgcp.c, main/xmldoc.c, apps/app_voicemail.c,
+ apps/app_while.c: Fix various typos reported by Lintian (Also fix
+ the typos in the comments)
+
+2010-07-01 22:16 +0000 [r273566] Russell Bryant <russell@digium.com>
+
+ * /, main/datastore.c: Merged revisions 273565 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273565 | russell | 2010-07-01 17:09:19 -0500 (Thu, 01 Jul 2010)
+ | 7 lines Don't return a partially initialized datastore. If
+ memory allocation fails in ast_strdup(), don't return a partially
+ initialized datastore. Bad things may happen. (related to
+ ABE-2415) ........
+
+2010-07-01 20:28 +0000 [r273522] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 273474 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273474 | jpeeler | 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010)
+ | 14 lines Allow admin user to join conference without using
+ admin mode and no user pin. Configuring the conference in
+ meetme.conf like the following: conf => 2345,,6666 did not prompt
+ for pin when used without admin mode. This meant that the
+ conference could not be joined as an admin even if the user knew
+ the correct pin. The original bug report was submitted claiming
+ that the blank user pin should deny entry into the conference. I
+ think a better way to handle this would be with a feature
+ enhancement that used the following syntax: conf => 2345,X,6666 -
+ where X denotes no acceptable pin allowed (closes issue #15704)
+ Reported by: modelnine ........
+
+2010-07-01 19:34 +0000 [r273464] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c: Properly handle failures of fax->start_session()
+ FAX-177
+
+2010-07-01 16:40 +0000 [r273427] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c, channels/sip/include/sip.h: correct handling
+ of get_destination return values A failure when calling the
+ get_destination can mean multiple things. If the extension is not
+ found, a 404 error is appropriate, but if the URI scheme is
+ incorrect, a 404 is not approperiate. This patch adds the
+ get_destination_result enum to differentiate between these and
+ other failure types. The only logical difference in this patch is
+ that we now send a "416 Unsupported URI scheme" response instead
+ of a "404" when the scheme is not recognized. This indicates to
+ the initiator of the INVITE to retry the request with a correct
+ URI.
+
+2010-07-01 15:12 +0000 [r273355] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 273354 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273354 | jpeeler | 2010-07-01 10:05:43 -0500 (Thu, 01 Jul 2010)
+ | 12 lines Ensure channel placed in meetme in ringing state is
+ properly hung up. An outgoing channel placed in meetme while
+ still ringing which was then hung up would not exit meetme and
+ the channel was not properly destroyed. Specifically checking for
+ this scenario by looking at the appropriate control frames
+ resolves the issue. (closes issue #15871) Reported by: Ivan
+ Patches: meetme_congestion_trunk_v2.patch uploaded by Ivan
+ (license 229) ........
+
+2010-07-01 14:37 +0000 [r273270-273352] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/manager.c: Fixed whitespace problems
+
+ * main/manager.c: Altered my comment about TCP_NODELAY
+
+ * addons/chan_mobile.c: Don't free written frames in chan_mobile's
+ mbl_write() function. (closes issue #16430) Reported by: azbest
+ Tested by: azbest
+
+ * main/manager.c: Set TCP_NODELAY on manager TCP sockets to prevent
+ delays on outgoing packets. This regression was introduced in
+ r48338. AST-359
+
+2010-06-30 17:28 +0000 [r273233] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * res/res_rtp_asterisk.c: Fix rt(c)p set debug ip taking wrong
+ argument Also clean up some coding errors. (closes issue #17469)
+ Reported by: wdoekes Patches: astsvn-rtp-set-debug-ip.patch
+ uploaded by wdoekes (license 717) Tested by: wdoekes, pabelanger
+
+2010-06-30 17:17 +0000 [r273197-273198] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/config.h: Remove unnecessary if test in
+ CV_DSTR()
+
+ * include/asterisk/config.h: Misc doxygen cleanup in config.h
+
+2010-06-30 01:07 +0000 [r273054-273144] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c: Permission checking for the system application is
+ backwards. (closes issue #17550) Reported by: kenner Patches:
+ manager.c.diff uploaded by kenner (license 1040) Tested by:
+ kenner
+
+ * main/config.c: Don't attempt to proceed if our internal parser
+ indicates an invalid file. (closes issue #17560) Reported by:
+ Nick_Lewis
+
+ * /, channels/chan_sip.c: Merged revisions 273060 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273060 | tilghman | 2010-06-29 18:15:28 -0500 (Tue, 29 Jun 2010)
+ | 10 lines Allow the "useragent" value to be restored into memory
+ from the realtime backend. This value is purely informational. It
+ does not alter configuration at all. (closes issue #16029)
+ Reported by: Guggemand Patches: realtime-useragent.patch uploaded
+ by Guggemand (license 897) Tested by: Guggemand ........
+
+ * /: Recorded merge of revisions 273057 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r273057 | tilghman | 2010-06-29 17:58:58 -0500 (Tue, 29 Jun 2010)
+ | 4 lines _Really_ skip the channel... don't just retry for
+ another 200 cycles. (Closes issue SWP-1652, ABE-2240) ........
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Exclude libical for insufficient versions.
+
+ * main/pbx.c: Send DialPlanComplete as a response, not as a
+ separate event. Otherwise, it goes to all manager sessions and
+ may exclude the current session, if the Events mask excludes it.
+ (closes issue #17504) Reported by: rrb3942 Patches:
+ showdialplan_patch.diff uploaded by rrb3942 (license 1003) Tested
+ by: rrb3942
+
+2010-06-29 20:44 +0000 [r272981] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: send a 400 Bad Request on malformed sip
+ request RFC 2361 section 24.4.1 send a 400 Bad Request if the
+ request can not be understood due to malformed syntax. Currently
+ we simply ignore a packet with a missing callid, to, from, or via
+ header. Instead of ignoring we now send the 400 Bad request.
+
+2010-06-28 21:50 +0000 [r272923-272926] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/asterisk.c: Merged revisions 272925 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272925 | tilghman | 2010-06-28 16:50:02 -0500 (Mon, 28 Jun 2010)
+ | 8 lines Don't change ownership/group/permissions on run
+ directory, if it already exists. (closes issue #17076) Reported
+ by: stuarth Patches: 20100324__issue17076.diff.txt uploaded by
+ tilghman (license 14) Tested by: stuarth ........
+
+ * /, main/config.c: Merged revisions 272921-272922 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r272921 | tilghman | 2010-06-28 16:29:27 -0500 (Mon, 28
+ Jun 2010) | 8 lines Change the way that we read include files, to
+ accommodate for changes in GCC 4.4. (closes issue #17472)
+ Reported by: seandarcy Patches: config2.patch uploaded by nivan
+ (license 1066) Tested by: nivan ........ r272922 | tilghman |
+ 2010-06-28 16:38:49 -0500 (Mon, 28 Jun 2010) | 2 lines Also trim
+ trailing blanks on #includes ........
+
+2010-06-28 18:38 +0000 [r272880] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c, channels/sip/reqresp_parser.c,
+ channels/sip/include/sip.h,
+ channels/sip/include/reqresp_parser.h: rfc compliant sip option
+ parsing + new unit test RFC 3261 section 8.2.2.3 states that if
+ any unsupported options are found in the Require header field, a
+ "420 (Bad Extension)" response should be sent with an Unsupported
+ header field containing only the unsupported options. This is not
+ currently being done correctly. Right now, if Asterisk detects
+ any unsupported sip options in a Require header the entire list
+ of options are returned in the Unsupported header even if some of
+ those options are in fact supported. This patch fixes that by
+ building an unsupported options character buffer when parsing the
+ options that can be sent with the 420 response. A unit test
+ verifying this functionality has been created. Some code
+ refactoring was required. Review:
+ https://reviewboard.asterisk.org/r/680/
+
+2010-06-28 17:33 +0000 [r272805] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 272804 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272804 | mmichelson | 2010-06-28 12:31:40 -0500 (Mon, 28 Jun
+ 2010) | 5 lines Decode URI in contact header of 302 response.
+ ABE-2352 ........
+
+2010-06-28 15:33 +0000 [r272684] Russell Bryant <russell@digium.com>
+
+ * doc/tex/chan-mobile.tex (added), doc/tex/celdriver.tex,
+ doc/tex/chan_mobile.tex (removed), doc/tex/cdrdriver.tex,
+ doc/tex/asterisk.tex, doc/tex/cel-doc.tex: Use the underscore
+ package so that underscores do not need to be escaped.
+
+2010-06-28 14:55 +0000 [r272652] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: code guidelines cleanup for retrans_pkt()
+ function I am doing work in this function. I noticed a large
+ number of coding guidline fixes that needed to be made. Rather
+ than have those changes distract from my functional changes I
+ decided to separate these into a separate patch.
+
+2010-06-25 20:18 +0000 [r272568] Tilghman Lesher <tlesher@digium.com>
+
+ * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 272562 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272562 | tilghman | 2010-06-25 15:17:37 -0500 (Fri, 25 Jun 2010)
+ | 5 lines Make the structure of the table specified before match
+ the queries and results. (closes issue #17557) Reported by: cmaj
+ ........
+
+2010-06-25 19:42 +0000 [r272558] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c, include/asterisk/res_fax.h: Implemement support
+ for handling multiple documents when sending.
+
+2010-06-25 19:39 +0000 [r272557] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: chan_sip: more accurate retransmissions
+ RFC3261 states that Timer A should start at 500ms (T1) by
+ default. In chan_sip this value initially started at 1000ms and I
+ changed it to 500ms recently. After doing that I noticed in my
+ packet captures that it still occasionally retransmitted starting
+ at 1000ms instead of 500ms like I told it to. This occurs because
+ the scheduler runs in the do_monitor thread. If a new
+ retransmission is added while the do_monitor thread is sleeping
+ then it may not detect that retransmission for nearly 1000ms. To
+ fix this I just poke the do_monitor thread to wake up when a new
+ packet is sent reliably requiring retransmits. The thread then
+ detects the new scheduler entry and adjusts its sleep time to
+ account for it. Review: https://reviewboard.asterisk.org/r/747
+
+2010-06-25 19:17 +0000 [r272533] Tilghman Lesher <tlesher@digium.com>
+
+ * sounds/Makefile: Symlink sounds files, to save disk space, when
+ multiple tarballs/checkouts are on the same system.
+
+2010-06-24 22:11 +0000 [r272447] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 272446 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272446 | rmudgett | 2010-06-24 16:58:49 -0500 (Thu, 24 Jun 2010)
+ | 10 lines ss_thread calls pri_grab without lock during overlap
+ dial Recent changes to chan_dahdi with relation to overlap
+ dialing call pri_grab without first obtaining a lock. (closes
+ issue #17414) Reported by: pdf Patches: bug17414.patch uploaded
+ by jpeeler (license 325) ........
+
+2010-06-23 23:09 +0000 [r272370] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Resolve some errors produced during module
+ unload of chan_iax2. The external test suite stops Asterisk using
+ the "core stop gracefully" command. The logs from the tests show
+ that there are a number of problems with Asterisk trying to
+ cleanly shut down. This patch addresses the following type of
+ error that comes from chan_iax2: [Jun 22 16:58:11] ERROR[29884]:
+ lock.c:129 __ast_pthread_mutex_destroy: chan_iax2.c line 11371
+ (iax2_process_thread_cleanup): Error destroying mutex
+ &thread->lock: Device or resource busy For an example in the
+ context of a build, see:
+ http://bamboo.asterisk.org/browse/AST-TRUNK-739/log The primary
+ purpose of this patch is to change the thread pool shutdown
+ procedure to be more explicit to ensure that the thread exits
+ from a point where it is not holding a lock. While testing that,
+ I encountered various crashes due to the order of operations in
+ unload_module() being problematic. I reordered some things there,
+ as well. Review: https://reviewboard.asterisk.org/r/736/
+
+2010-06-23 22:36 +0000 [r272368] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 272367 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 This version
+ of the patch only adds AgentComplete for attended transfers. It
+ was already present for blind transfers. ........ r272367 |
+ mnicholson | 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) | 8
+ lines Send AgentComplete manager events in the event of blind and
+ attended transfers. (closes issue #16819) Reported by: elbriga
+ Patches: app_queue.diff uploaded by elbriga (license 482)
+ ........
+
+2010-06-23 21:53 +0000 [r272260-272332] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c: If there is realtime configuration, it
+ does not get re-read on reload unless the config file also
+ changes. (closes issue #16982) Reported by: dmitri Patches:
+ res_musiconhold.patch uploaded by dmitri (license 1001) Tested
+ by: atis
+
+ * res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael_lex.c,
+ res/ael/ael.flex: Ensure a NULL file while debugging cannot crash
+ AEL. (closes issue #17215) Reported by: vazir Patches:
+ 20100518__issue17215.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman
+
+2010-06-23 21:06 +0000 [r272257-272259] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * apps/app_meetme.c: Fix previous merge. ast_test_flag !=
+ ast_test_flag64
+
+ * /, apps/app_meetme.c: Merged revisions 272255 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272255 | pabelanger | 2010-06-23 16:57:01 -0400 (Wed, 23 Jun
+ 2010) | 12 lines First caller into a dynamic conference now enter
+ pin once. If MeetMe is configured to use dynamic conference
+ numbers, then the first caller (which creates the conference) had
+ to enter the PIN number twice. (closes issue #15878) Reported by:
+ shawkris Patches: issue15878.patch uploaded by pabelanger
+ (license 224) Tested by: pabelanger ........
+
+2010-06-23 20:59 +0000 [r272254-272256] Terry Wilson <twilson@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in: Update configure
+ when changing autconf m4 files...
+
+ * autoconf/ast_ext_tool_check.m4: Honor the --with-${library}=path
+ for AST_EXT_TOOL_CHECK (closes issue #16991) Reported by:
+ pprindeville Patches: with_netsnmp.patch.txt uploaded by twilson
+ (license 396) Tested by: twilson Review:
+ https://reviewboard.asterisk.org/r/739/
+
+2010-06-23 20:35 +0000 [r272243-272252] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * main/manager.c: Correct manager variable 'EventList' case.
+ (closes issue #17520) Reported by: kobaz Patches: manager.patch
+ uploaded by kobaz (license 834) Tested by: lmadsen
+
+ * configs/say.conf.sample: Add localization support for Spanish
+ (closes issue #17548) Reported by: cjacobsen Patches:
+ say.conf.sample.diff uploaded by cjacobsen (license 1029)
+
+2010-06-23 19:59 +0000 [r272218] Tim Ringenbach <tim.ringenbach@gmail.com>
+
+ * channels/chan_local.c: Add new AMI command LocalOptimizeAway.
+ This command lets you request a "/n" local channel optimize
+ itself out of the way anyway. Review:
+ https://reviewboard.asterisk.org/r/732/
+
+2010-06-23 18:45 +0000 [r272148-272150] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_mgcp.c: D'oh! Defaultenabled FTL.
+
+ * /: Recorded merge of revisions 272147 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272147 | tilghman | 2010-06-23 13:40:28 -0500 (Wed, 23 Jun 2010)
+ | 5 lines Backport part of revision 136715 to fix callerid in
+ voicemail text files (IMAP only). (closes issue #16945) Reported
+ by: mneuhauser ........
+
+2010-06-23 18:39 +0000 [r272146] Terry Wilson <twilson@digium.com>
+
+ * apps/app_meetme.c: Don't start the sla thread unless we realy
+ need it
+
+2010-06-23 18:25 +0000 [r272145] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_mgcp.c: Load all lines from realtime, not just the
+ first one. (closes issue #17144) Reported by: nahuelgreco
+ Patches: 20100513__issue17144__trunk.diff.txt uploaded by
+ tilghman (license 14) Tested by: tilghman
+
+2010-06-23 17:21 +0000 [r272109] Terry Wilson <twilson@digium.com>
+
+ * apps/app_meetme.c: Make sure reload updates SLA config Even if
+ there are no stations or trunks defined, we need to start the sla
+ thread to make sure we get the reload event. Also, when doing a
+ reload we need to remove the existing trunks and stations or they
+ end up hanging around. (closes issue #16818) Reported by: mbonin
+ Patches: sla_reload.patch uploaded by twilson (license 396)
+ Tested by: twilson
+
+2010-06-23 17:08 +0000 [r272090] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Add extra protection for reinvite glare
+ scenario. Testing proved that if Asterisk sent a connected line
+ reinvite, and the endpoint to which the reinvite were being sent
+ sent a reinvite, Asterisk would not properly respond with a 491
+ response. The reason is that on connected line reinvites, we set
+ the dialog's invitestate to INV_CALLING to prevent Asterisk from
+ sending a rapid flurry of connected line reinvites. For other
+ reinvites we do not do this. Because of the current invitestate,
+ when Asterisk received the reinvite, we interpreted this as a
+ spiraled INVITE, and thus did not behave properly. The fix for
+ this is to not enter the loop detection or spiral logic in
+ handle_request_invite if the channel state is currently up. This
+ way, no mid-call reinvites will be misinterpreted, no matter what
+ the nature of the reinvite may have been.
+
+2010-06-22 23:20 +0000 [r272052] Russell Bryant <russell@digium.com>
+
+ * channels/chan_dahdi.c: Don't try to lock/unlock an uninitialized
+ lock on a dahdi_pri. This small changes prevents
+ destroy_all_channels() from accessing a lock on an unused
+ dahdi_pri struct, resolving a ton of ERRORs that get spewed out
+ when shutting Asterisk down gracefully.
+
+2010-06-22 22:11 +0000 [r271905-272014] David Vossel <dvossel@digium.com>
+
+ * pbx/pbx_config.c: fixes issue with 'dialplan remove extension
+ blah' segfaulting with tab completion (closes issue #17440)
+ Reported by: kobaz
+
+ * channels/chan_sip.c: ignore CANCEL request after having already
+ received final response to INVITE RFC 3261 section 9 states that
+ a CANCEL has no effect on a request to a UAS that has already
+ given a final response. This patch checks to make sure there is a
+ pending invite before allowing a CANCEL request to be processed,
+ otherwise it responds to the CANCEL with a "481 Call/Transaction
+ Does Not Exist". Review: https://reviewboard.asterisk.org/r/697/
+
+ * main/manager.c: minor fixes for white/black event filters This
+ fixes a ref count leak in event filters and checks for a filter
+ container allocation failure during session creation.
+
+2010-06-22 17:35 +0000 [r271903] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 271902 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun
+ 2010) | 8 lines Decrease the module ref count in sip_hangup when
+ SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the
+ ref count correct. (closes issue #16815) Reported by: rain
+ Patches: chan_sip-unref-fix.diff uploaded by rain (license 327)
+ (modified) Tested by: rain ........
+
+2010-06-22 16:29 +0000 [r271868] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c, configs/manager.conf.sample, CHANGES: Add regular
+ expression filtering for manager events. This patch as documented
+ in the sample config allows one to optionally apply white, black,
+ or both types of filtering to manager events. The new
+ 'eventfilter' option is set per user. (closes issue #14861)
+ Reported by: fnordian Patches: eventfilter3.patch uploaded by
+ fnordian (license 110), modified by me Review:
+ https://reviewboard.asterisk.org/r/673/
+
+2010-06-22 16:28 +0000 [r271833-271867] Russell Bryant <russell@digium.com>
+
+ * res/ais/clm.c, res/ais/evt.c: Resolve some errors that occur on a
+ graceful shutdown. Don't Finalize() if Initialize() did not
+ succeed. This resulted in an error about trying to Finalize() an
+ invalid handle. Also trim some trailing whitespace while in the
+ area.
+
+ * res/res_fax.c: Change the method of retrieving the Asterisk
+ version string. Using this method makes it so res_fax doesn't
+ have to be rebuilt on every svn update.
+
+2010-06-22 15:46 +0000 [r271831] David Vossel <dvossel@digium.com>
+
+ * main/features.c: fixes attended transfer behavior when both
+ transferee and transferer hung up If both the transferer and
+ transferee of a attended transfer hangup before the new channel
+ picks up, the new channel should be hung up as well as it has no
+ endpoint to talk to. This mirrors the expected behavior used in
+ 1.4. (closes issue #17444) Reported by: corruptor
+
+2010-06-22 15:08 +0000 [r271690-271764] Matthew Nicholson <mnicholson@digium.com>
+
+ * CHANGES: Updated the CHANGES file documenting the addition of a
+ configurable port in the dundi config file.
+
+ * configs/dundi.conf.sample, /, pbx/pbx_dundi.c: Merged revisions
+ 271761 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r271761 | mnicholson | 2010-06-22 09:49:36 -0500 (Tue, 22 Jun
+ 2010) | 9 lines Allow users to specify a port for dundi peers.
+ (closes issue #17056) Reported by: klaus3000 Patches:
+ dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65)
+ Tested by: klaus3000 ........
+
+ * /, channels/chan_sip.c, include/asterisk/strings.h,
+ channels/sip/include/sip.h: Merged revisions 271689 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue,
+ 22 Jun 2010) | 8 lines Modify chan_sip's packet generation api to
+ automatically calculate the Content-Length. This is done by
+ storing packet content in a buffer until it is actually time to
+ send the packet, at which time the size of the packet is
+ calculated. This change was made to ensure that the
+ Content-Length is always correct. (closes issue #17326) Reported
+ by: kenner Tested by: mnicholson, kenner Review:
+ https://reviewboard.asterisk.org/r/693/ ........ This change also
+ adds an ast_str_copy_string() function (similar to
+ ast_copy_string), that copies one ast_str into another, properly
+ handling embedded nulls.
+
+2010-06-21 22:41 +0000 [r271657] Tilghman Lesher <tlesher@digium.com>
+
+ * build_tools/menuselect-deps.in, configure, configure.ac,
+ res/res_timing_kqueue.c: Conflict kqueue on OS X, since it
+ doesn't work there yet, anyway.
+
+2010-06-21 21:58 +0000 [r271625] David Vossel <dvossel@digium.com>
+
+ * codecs/codec_speex.c, codecs/ex_speex.h,
+ contrib/editors/asterisk.vim: add speex 16khz sample frame so
+ codec cost can be calculated (closes issue #17534) Reported by:
+ fabled Patches: speex-wb-sample.diff uploaded by fabled (license
+ 448)
+
+2010-06-21 20:46 +0000 [r271554] Jeff Peeler <jpeeler@digium.com>
+
+ * res/ael/pval.c, /: Merged revisions 271552 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r271552 | jpeeler | 2010-06-21 15:37:47 -0500 (Mon, 21 Jun 2010)
+ | 7 lines Do not use sizeof to calculate size of a heap allocated
+ character array. Change left out from 271399. (closes issue
+ #16053) Reported by: diLLec ........
+
+2010-06-21 20:46 +0000 [r271551-271553] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c, channels/sip/reqresp_parser.c: fixes crash
+ when From header URI is missing "sip:" (closes issue #17437)
+ Reported by: klaus3000 Patches: sip_crash uploaded by dvossel
+ (license 671) Tested by: klaus3000
+
+ * res/res_rtp_asterisk.c: fixes logic error introduced by slin16
+ sip support
+
+2010-06-21 05:10 +0000 [r271520] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_saycounted.c (added), CHANGES: Add new application for
+ declining counting words in multiple languages. (closes issue
+ #16869) Reported by: chappell Patches: app_say_counted-20100317.c
+ uploaded by chappell (license 8) Tested by: chappell
+
+2010-06-18 21:32 +0000 [r271483] Jeff Peeler <jpeeler@digium.com>
+
+ * res/ael/pval.c, /, include/asterisk/pval.h, pbx/pbx_ael.c: Merged
+ revisions 271399 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r271399 | jpeeler | 2010-06-18 14:28:24 -0500 (Fri, 18 Jun 2010)
+ | 11 lines Fix crash when parsing some heavily nested statements
+ in AEL on reload. Due to the recursion used when compiling AEL in
+ gen_prios, all the stack space was being consumed when parsing
+ some AEL that contained nesting 13 levels deep. Changing a few
+ large buffers to be heap allocated fixed the crash, although I
+ did not test how many more levels can now be safely used. (closes
+ issue #16053) Reported by: diLLec Tested by: jpeeler ........
+
+2010-06-18 18:59 +0000 [r271341] David Vossel <dvossel@digium.com>
+
+ * main/file.c: file.c was truncating audio file formats to the
+ lower 32bits.
+
+2010-06-18 18:36 +0000 [r271336] Jeff Peeler <jpeeler@digium.com>
+
+ * /: Recorded merge of revisions 271335 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r271335 | jpeeler | 2010-06-18 13:33:17 -0500 (Fri, 18 Jun 2010)
+ | 13 lines Eliminate deadlock potential in dahdi_fixup(). (This
+ is a backport of 269307, committed to trunk by rmudgett.) Calling
+ dahdi_indicate() when the channel private lock is already held
+ can cause a deadlock if the PRI lock is needed because
+ dahdi_indicate() will also get the channel private lock. The
+ pri_grab() function assumes that the channel private lock is held
+ once to avoid deadlock. (closes issue #17261) Reported by: aragon
+ ........
+
+2010-06-17 21:23 +0000 [r271231-271300] David Vossel <dvossel@digium.com>
+
+ * channels/sip/reqresp_parser.c: fixes some coding guideline issue
+
+ * channels/sip/include/dialog.h, channels/chan_sip.c,
+ channels/sip/include/sip.h: retransmit response to BYE requests
+ until timer J expires According to RFC 3261 section 17.2.2, which
+ describes non-INVITE server transaction, when a dialog enters the
+ Completed state it must destroy the dialog after Timer J (T1*64)
+ fires. For a BYE transaction Asterisk terminates the dialog
+ immediately during sip_hangup() when it should be waiting T1*64
+ ms. This results in some odd behavior. For instance if Asterisk
+ receives a BYE and transmits a 200ok in response, if the endpoint
+ never receives the 200ok it will retransmit the BYE to which
+ Asterisk responds with a "481 Call leg/transaction does not
+ exist" because the dialog is already gone. To resolve this I made
+ a function called sip_scheddestroy_final(). This differs slightly
+ from sip_schedestroy() in that it enables a flag that will
+ prevent the destruction from ever being rescheduled or canceled
+ afterwards. It also prevents the pvt's needdestroy flag from
+ being set which triggers the destruction of the dialog within the
+ do_monitor thread(). By using this function we are guaranteed
+ destruction will not occur until the scheduled time. This allows
+ Asterisk to respond to any possible retransmits for a dialog
+ after we process the initial BYE request for T1*64 ms. Other
+ changes: I removed two instances where sip_cancel_destroy is used
+ right before calling sip_scheddestroy. sip_scheddestroy always
+ calls sip_cancel_destroy before scheduling the new destruction so
+ it is completely unnecessary. Review:
+ https://reviewboard.asterisk.org/r/694/
+
+ * res/res_rtp_asterisk.c, main/rtp_engine.c, CHANGES: adds support
+ for slin16 in sip (closes issue #16153) Reported by: kfister
+ Patches: 16153-1.6.2.0-rc5.patch uploaded by kfister (license
+ 912) slin16.sip.patch.1 uploaded by malcolmd (license 924) Tested
+ by: kfister, malcolmd
+
+ * main/channel.c, res/res_rtp_asterisk.c, main/frame.c,
+ main/rtp_engine.c, codecs/codec_speex.c, CHANGES,
+ include/asterisk/frame.h: adds speex 16khz audio support (closes
+ issue #17501) Reported by: fabled Patches:
+ asterisk-trunk-speex-wideband-v2.patch uploaded by fabled
+ (license 448) Tested by: malcolmd, fabled, dvossel
+
+2010-06-17 15:34 +0000 [r271192] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/sig_analog.c: Change expected operation from error to
+ debug message
+
+2010-06-17 00:30 +0000 [r271089] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * apps/app_meetme.c: option w[(secs)] incorrectly capitalized in
+ xmldoc (closes issue #17516) Reported by: karlfife
+
+2010-06-16 22:37 +0000 [r271056] David Vossel <dvossel@digium.com>
+
+ * channels/sip/reqresp_parser.c: addition of more parse_uri test
+ cases
+
+2010-06-16 21:17 +0000 [r270987] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, configs/extensions.conf.sample: Merged revisions 270979 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r270979 | pabelanger | 2010-06-16 17:10:05 -0400 (Wed, 16 Jun
+ 2010) | 4 lines Fixed typo in macro-page Reported to
+ #asterisk-dev by a student of jsmith. ........
+
+2010-06-16 21:12 +0000 [r270981-270983] Jason Parker <jparker@digium.com>
+
+ * channels/chan_agent.c: Fix the actual place that was pointed out,
+ for previous commit.
+
+ * /, channels/chan_agent.c: Merged revisions 270980 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r270980 | qwell | 2010-06-16 16:10:09 -0500 (Wed, 16 Jun
+ 2010) | 4 lines Need to lock the agent chan before access its
+ internal bits. Pointed out by russellb on asterisk-dev mailing
+ list. ........
+
+2010-06-16 20:34 +0000 [r270974] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/dnsmgr.c, main/acl.c: Set sin_family to AF_INET when doing
+ lookups, also reset sin_port the first time the ip address
+ changes. (closes issue #17496) Reported by: ManChicken (closes
+ issue #15827) Reported by: DennisD Patches: dnsmgr_15827.patch
+ uploaded by chappell (license 8) Tested by: DennisD, gentlec,
+ damage, wimpy
+
+2010-06-16 19:03 +0000 [r270940] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, res/res_rtp_asterisk.c, main/frame.c,
+ main/rtp_engine.c, channels/chan_sip.c, CHANGES,
+ channels/chan_iax2.c, include/asterisk/frame.h,
+ formats/format_g719.c (added): addition of G.719 pass-through
+ support (closes issue #16293) Reported by: malcolmd Patches:
+ g719.passthrough.patch.7 uploaded by malcolmd (license 924)
+ format_g719.c uploaded by malcolmd (license 924)
+
+2010-06-16 18:43 +0000 [r270936] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * res/res_agi.c, CHANGES: MSG_OOB flag on HANGUP packet removed.
+ Per Tilghman's request on IRC (#asterisk-bugs). (closes issue
+ #17506) Reported by: brycebaril Tested by: pabelanger, tilghman
+
+2010-06-16 17:36 +0000 [r270867] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 270866 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r270866 | dvossel | 2010-06-16 12:35:29 -0500 (Wed, 16
+ Jun 2010) | 22 lines fixes chan_iax2 race condition There is code
+ in chan_iax2.c that attempts to guarantee that only a single
+ active thread will handle a call number at a time. This code
+ works once the thread is added to an active_list of threads, but
+ we are not currently guaranteed that a newly activated thread
+ will enter the active_list immediately because it is left up to
+ the thread to add itself after frames have been queued to it.
+ This means that if two frames come in for the same call number at
+ the same time, it is possible for them to grab two separate
+ threads because the first thread did not add itself to the
+ active_list fast enough. This causes some pretty complex
+ problems. This patch resolves this race condition by immediately
+ adding an activated thread to the active_list within the network
+ thread and only depending on the thread to remove itself once it
+ is done processing the frames queued to it. By doing this we are
+ guaranteed that if another frame for the same call number comes
+ in at the same time, that this thread will immediately be found
+ in the active_list of threads. Review:
+ https://reviewboard.asterisk.org/r/720/ ........
+
+2010-06-16 16:45 +0000 [r270836] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/sig_analog.c: Fix no call waiting caller ID Clearing the
+ callwaitcas flag in analog_call was causing the incoming D digit
+ to be ignored which triggers sending the caller ID.
+
+2010-06-16 15:05 +0000 [r270801] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * doc/tex/channelvariables.tex: Update formatting for
+ channelvariables.tex (closes issue #17511) Reported by: klaus3000
+ Patches: channelvariables.tex-patch.txt uploaded by klaus3000
+ (license 65) Tested by: pabelanger
+
+2010-06-15 22:48 +0000 [r270726] Russell Bryant <russell@digium.com>
+
+ * channels/sig_analog.c: Don't blow up if an ast_channel doesn't
+ get allocated.
+
+2010-06-15 21:42 +0000 [r270658-270692] Terry Wilson <twilson@digium.com>
+
+ * main/http.c: Don't continue sending the file when there has been
+ an error If there is a problem with a firmware file, Polycom
+ phones will close the connection. We were continuing to send the
+ file anyway. There should be no reason to continue sending a file
+ if there is an error writing it. (closes issue #16682) Reported
+ by: lmadsen
+
+ * res/res_phoneprov.c: Don't send files twice and remove extra \r\n
+ from header After the manager http auth changes, we forgot to
+ remove the manual sending of the file. Also, ast_http_send adds
+ two \r\n to the header that is passed to it, so a trailing \r\n
+ is removed from the Content-type header. It might be better to
+ change ast_http_send, but I don't like changing the behavior of
+ an API function. (closes issue #17239) Reported by: cjacobsen
+ Patches: patch2.diff uploaded by cjacobsen (license 1029) Tested
+ by: lathama, cjacobsen
+
+ * channels/chan_sip.c: Make contactdeny apply to src ip when
+ nat=yes chan_sip's "contactdeny" feature screens the "to be
+ registered contact". In case of nat=yes it should not use the
+ address information from the Contact header (which is not used at
+ all for routing), but the source IP address of the request. Thus,
+ if nat=yes and a client sends a request from a denied IP address
+ (e.g. by spoofing the src-IP address) it can bypass the
+ screening. This commit makes contactdeny apply to the src ip when
+ nat=yes instead. (closes issue #17276) Reported by: klaus3000
+ Patches: patch-asterisk-trunk-contactdeny.txt uploaded by
+ klaus3000 (license 65) Tested by: klaus3000
+
+2010-06-15 18:26 +0000 [r270519-270584] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 270583 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r270583 | tilghman | 2010-06-15 13:25:12 -0500 (Tue, 15 Jun 2010)
+ | 5 lines Variables have always been case-sensitive, so we should
+ not be removing case-insensitive matches. Bug reported via the
+ -dev list. See
+ http://lists.digium.com/pipermail/asterisk-dev/2010-June/044510.html
+ ........
+
+ * res/res_jabber.c: Argh, mixed declarations and code.
+
+ * configs/jabber.conf.sample, include/asterisk/jabber.h,
+ doc/distributed_devstate-XMPP.txt (added), CHANGES,
+ res/res_jabber.c: Add distributed devicestate via the XMPP
+ protocol. (closes issue #15757) Reported by: Marquis Patches:
+ distributed_devstate-XMPP.txt uploaded by lmadsen (license 10)
+ Tested by: Marquis, lmadsen, marcelloceschia Review:
+ https://reviewboard.asterisk.org/r/351/
+
+2010-06-15 12:51 +0000 [r270443] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/voicemail.conf.sample: Merged revisions 270442 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r270442 | lmadsen | 2010-06-15 07:47:03 -0500 (Tue, 15 Jun 2010)
+ | 1 line Move information about zonemessages into the
+ [zonemessages] section. ........
+
+2010-06-14 21:33 +0000 [r270332] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 270331 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r270331 | pabelanger | 2010-06-14 17:31:59 -0400 (Mon,
+ 14 Jun 2010) | 14 lines Properly play first file in sort list.
+ When using sort=alpha we would always skip the first file in the
+ list first time through. We now check for that properly. (closes
+ issue #17470) Reported by: pabelanger Patches: sort.aplha.patch
+ uploaded by pabelanger (license 224) Tested by: lmadsen Review:
+ https://reviewboard.asterisk.org/r/703/ ........
+
+2010-06-14 20:51 +0000 [r270298] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_ss7.h, channels/sig_ss7.c:
+ Extract sig_ss7_init_linkset() to sig_ss7. Also found a place
+ where sig_pri_init_pri() was inlined and called it instead.
+
+2010-06-14 19:41 +0000 [r270260] Jason Parker <jparker@digium.com>
+
+ * channels/chan_agent.c: Add option to get untruncated channel name
+ from AGENT function. The "channel" option would chop the channel
+ name at the last '-', which made it useless for something like a
+ channel transfer from the dialplan. The "fullchannel" option will
+ return the channel name as-is. ABE-2218
+
+2010-06-14 15:55 +0000 [r270219] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, channels/sig_pri.c: Add digit
+ manipulation tag support to chan_dahdi/sig_pri like chan_misdn.
+ Add the append_msn_to_cid_tag option to chan_dahdi like
+ chan_misdn. Review: https://reviewboard.asterisk.org/r/696/
+
+2010-06-13 09:16 +0000 [r270184] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * autoconf/ast_check_pwlib.m4, configure: bashism in configure
+ script Theoretically the ./configure script is a pure
+ bourne-shell script. Practically it may be run by bash if /bin/sh
+ is not good enough. But we should not count on it. See bug report
+ for the gory details. (closes issue #17485) Patches:
+ 0001-remove-bashism-from-ast_check_pwlib.m4.patch uploaded by
+ tzafrir (license 46)
+
+2010-06-13 01:53 +0000 [r270042-270151] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Reverting patch and reopening issue #16155, as patch breaks
+ FreeBSD / OSX builds.
+
+ * /, doc/HOWTO_collect_debug_information.txt: Merged revisions
+ 270078 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r270078 | pabelanger | 2010-06-12 14:54:20 -0400 (Sat, 12 Jun
+ 2010) | 2 lines Fix typo in example ........
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac: Use
+ pkg-config to find gmime libraries This way the libraries can be
+ found even if they are in non-standard locations. (closes issue
+ #16155) Reported by: jcollie Patches:
+ 0008-change-configure.ac-to-look-for-pkg-config-gmime-2.0.patch
+ uploaded by jcollie (license 412) Tested by: jsmith, tilghman,
+ pabelanger
+
+2010-06-11 18:31 +0000 [r269936-269976] Tilghman Lesher <tlesher@digium.com>
+
+ * main/frame.c, /: Merged revisions 269960 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r269960 | tilghman | 2010-06-11 13:23:05 -0500 (Fri, 11 Jun 2010)
+ | 8 lines For SpeeX, 0 bits remaining is valid and does not need
+ an emitted warning. (closes issue #15762) Reported by: nblasgen
+ Patches: issue15672.patch uploaded by pabelanger (license 224)
+ Tested by: nblasgen ........
+
+ * CHANGES, main/db.c: Add DBGetComplete event after a
+ DBGetResponse. (closes issue #16965) Reported by: rrb3942
+ Patches: DBGetComplete.patch uploaded by rrb3942 (license 1003)
+
+ * main/logger.c: Remove lines from the output related to the
+ backtrace itself.
+
+2010-06-10 20:30 +0000 [r269889] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * Makefile, makeopts.in: Remove ASTBINDIR variable (closes issue
+ #17031) Reported by: pabelanger Patches:
+ Makefile.ASTBINDIR.v2.patch uploaded by pabelanger (license 224)
+ Tested by: pabelanger, tilghman
+
+2010-06-10 19:34 +0000 [r269749-269822] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 269821 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r269821 | mmichelson | 2010-06-10 14:30:12 -0500 (Thu, 10 Jun
+ 2010) | 19 lines Fix potential crash when writing raw SLIN audio
+ on a PLC-enabled channel. The issue here was that the frame
+ created when adjusting for PLC had no offset to its audio data.
+ If this frame were translated to another format prior to being
+ sent out an RTP socket, all went well because the translation
+ code would put an appropriate offset into the frame. However, if
+ the SLIN audio were not translated before being sent out the RTP
+ socket, bad things would happen. Specifically, the
+ ast_rtp_raw_write makes the assumption that the frame has at
+ least enough of an offset that it can accommodate an RTP header.
+ This was not the case. As such, data was being written prior to
+ the allocation, likely corrupting the data the memory allocator
+ had written. Thus when the time came to free the data, all hell
+ broke loose. ....Well, Asterisk crashed at least. The fix was
+ just what one would expect. Offset the data in the frame by a
+ reasonable amount. The method I used is a bit odd since the data
+ in the frame is 16 bit integers and not bytes. I left a big ol'
+ comment about it. This can be improved on if someone is
+ interested. I was more interested in getting the crash resolved.
+ ........
+
+ * doc/tex/plc.tex (added), doc/tex/asterisk.tex: Add documentation
+ explaining PLC in Asterisk. Review:
+ https://reviewboard.asterisk.org/r/688/
+
+2010-06-10 13:17 +0000 [r269711] Russell Bryant <russell@digium.com>
+
+ * tests/test_heap.c: Fix an off by one error that caused a unit
+ test to occasionally crash.
+
+2010-06-10 12:28 +0000 [r269707] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/logger.c: Ensure that 'logger show channels' works properly
+ when wildcards are used in logger.conf.
+
+2010-06-10 08:15 +0000 [r269636] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/logger.c, utils/extconf.c, main/asterisk.c: Merged
+ revisions 269635 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r269635 | tilghman | 2010-06-10 02:52:34 -0500 (Thu, 10 Jun 2010)
+ | 9 lines Ensure restartable system calls can restart (BSD signal
+ semantics). This eliminates the annoying <beep> on the console.
+ (closes issue #17477) Reported by: jvandal Patches:
+ 20100610__issue17477.diff.txt uploaded by tilghman (license 14)
+ ........
+
+2010-06-10 00:32 +0000 [r269417-269602] Russell Bryant <russell@digium.com>
+
+ * channels/chan_dahdi.c: Attempt to fix a FreeBSD build error by
+ including sys/stat.h.
+ http://bamboo.asterisk.org/download/AST-TRUNKFREEBSD/build_logs/AST-TRUNKFREEBSD-187.log
+
+ * main/lock.c: Attempt to fix FreeBSD build problem.
+
+ * /, channels/chan_oss.c: Merged revisions 269495 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r269495 | russell | 2010-06-09 17:18:37 -0500 (Wed, 09 Jun 2010)
+ | 2 lines Don't stop Asterisk if chan_oss fails to register
+ 'Console' (due to another channel driver already claiming it).
+ ........
+
+ * include/asterisk/event.h, main/event.c: Resolve an invalid memory
+ read on an event. Valgrind pointed out that attempting to get an
+ IE value from an event that has no IEs produces an invalid memory
+ read past the end of the event. Thanks to mmichelson for pointing
+ the problem out to me and then testing the fix.
+
+2010-06-09 17:32 +0000 [r269346] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * contrib/init.d/rc.debian.asterisk, /, main/term.c: Merged
+ revisions 269334 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r269334 | pabelanger | 2010-06-09 13:24:53 -0400 (Wed, 09 Jun
+ 2010) | 12 lines Fix Debian init script to not use -c. When using
+ the init script as-is currently, it could cause issues on Debian
+ such as high CPU usage. This fix has worked for several people so
+ I'm implementing the change. We now handle color displays
+ properly. (closes issue #16784) Reported by: pabelanger Patches:
+ 20100530__issue16784__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: pabelanger, tilghman ........
+
+2010-06-09 17:06 +0000 [r269307-269308] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_ss7.h, channels/sig_ss7.c:
+ Add missing API function to sig_ss7: sig_ss7_fixup().
+
+ * channels/chan_dahdi.c: Eliminate deadlock potential in
+ dahdi_fixup(). Calling dahdi_indicate() within dahdi_fixup()
+ while the owner pointers are in a potentially inconsistent state
+ is a potentially bad thing in principle. However, calling
+ dahdi_indicate() when the channel private lock is already held
+ can cause a deadlock if the PRI lock is needed because
+ dahdi_indicate() will also get the channel private lock. The
+ pri_grab() function assumes that the channel private lock is held
+ once to avoid deadlock.
+
+2010-06-09 15:09 +0000 [r269271] David Vossel <dvossel@digium.com>
+
+ * res/res_musiconhold.c: fixes crash in moh when cachertclasses
+ flag is used The result for moh_register was not verified to
+ guarantee the mohclass as added to the container. (closes issue
+ #16993) Reported by: dmitri Patches:
+ res_musiconhold_rtclass2.patch uploaded by dmitri (license 1001)
+ moh_crash2.diff uploaded by dvossel (license 671) Tested by:
+ dmitri
+
+2010-06-09 13:17 +0000 [r269238] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, CHANGES:
+ dial by name in chan_dahdi * chan_dahdi supports dialing
+ configuring and dialing by device file name.
+ DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 .
+ Likewise it may appear in chan_dahdi.conf as 'channel =>
+ span-name!local!1'. * A new options for chan_dahdi.conf:
+ 'ignore_failed_channels'. Boolean. False by default. If set,
+ chan_dahdi will ignore failed 'channel' entries. Handy for the
+ above name-based syntax as it does not depend on initialization
+ order. * have my_pri_make_cc_dialstring() only manupulate
+ dial-strings of group (gGrR) dialing, which make it lsightly more
+ complicated. https://reviewboard.asterisk.org/r/535/
+
+2010-06-09 10:55 +0000 [r269187-269205] Russell Bryant <russell@digium.com>
+
+ * contrib/scripts/install_prereq: Add libjack-dev to
+ install_prereq.
+
+ * contrib/scripts/install_prereq: Add libpopt-dev, libical-dev, and
+ libspandsp-dev to install_prereq.
+
+ * contrib/scripts/install_prereq: Add libnewt-dev to
+ install-prereq.
+
+ * contrib/scripts/install_prereq: Add libopenais-dev to
+ install_prereq.
+
+ * contrib/scripts/install_prereq: Add an "install-unpackaged"
+ command to install_prereq for installing unpackaged dependencies
+ (such as NBS and libresample).
+
+ * contrib/scripts/install_prereq: Add libcurl to install_prereq.
+
+ * contrib/scripts/install_prereq: Add freetds-dev to
+ install_prereq.
+
+ * contrib/scripts/install_prereq: Add libradiusclient-ng-dev to
+ install_prereq.
+
+ * contrib/scripts/install_prereq: Add libbluetooth-dev to
+ install_prereq.
+
+ * contrib/scripts/install_prereq: Add libmysqlclient-dev to
+ install_prereq.
+
+ * contrib/scripts/install_prereq: Add libgtk2.0-dev to the packages
+ list for install_prereq.
+
+2010-06-08 23:48 +0000 [r269153] Bradley Latus <brad.latus@gmail.com>
+
+ * configs/cdr_custom.conf.sample, configs/cdr_tds.conf.sample,
+ cdr/cdr_sqlite.c, configs/cdr_sqlite3_custom.conf.sample,
+ funcs/func_cdr.c, configs/cdr_syslog.conf.sample, UPGRADE.txt,
+ cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c, cdr/cdr_pgsql.c,
+ CHANGES, cdr/cdr_odbc.c, cdr/cdr_tds.c,
+ configs/cdr_odbc.conf.sample: Add High Resolution Times to CDRs
+ for Asterisk People expressed an interest in having access to the
+ exact length of calls to a finer degree than seconds. See the
+ CHANGES and UPGRADE.txt for usage also updated the sample configs
+ to note the change. Patch by snuffy. (closes issue #16559)
+ Reported by: cianmaher Tested by: cianmaher, snuffy Review:
+ https://reviewboard.asterisk.org/r/461/
+
+2010-06-08 22:45 +0000 [r269119] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/localtime.h: Fix build on Mac OS X (and maybe
+ FreeBSD, too)
+
+2010-06-08 18:50 +0000 [r269083] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_fax.c: Don't pass null to manager_event() (closes issue
+ #17087) Reported by: bklang Patches: app-fax-null-sprintf1.diff
+ uploaded by mnicholson (license 96) Tested by: bklang
+
+2010-06-08 15:41 +0000 [r269008] Russell Bryant <russell@digium.com>
+
+ * Makefile.rules: Ensure CONFIG_FLAGS makes it into the build rules
+ when doing out of tree builds. (closes issue #16685) Reported by:
+ pprindeville
+
+2010-06-08 15:39 +0000 [r269007] Sean Bright <sean@malleable.com>
+
+ * /, cdr/cdr_tds.c: Merged revisions 269006 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r269006 | seanbright | 2010-06-08 11:28:49 -0400 (Tue, 08 Jun
+ 2010) | 11 lines Reduce startup time for cdr_tds with large CDR
+ tables. Since we are just checking for table existence, add a
+ WHERE clause that will return no rows but will raise an error if
+ the table doesn't exist. (closes issue #17380) Reported by:
+ kkwong Patches: issue17380-01.patch uploaded by seanbright
+ (license 71) Tested by: kkwong ........
+
+2010-06-08 15:23 +0000 [r268969-268988] Leif Madsen <lmadsen@digium.com>
+
+ * configs/sip.conf.sample: Update note in sip.conf.sample. Update
+ note in sip.conf.sample about externip and externhost with STUN.
+ (closes issue #16323) Reported by: klaus3000 Patches:
+ sip.conf.sample-patch.txt uploaded by klaus3000 (license 65)
+
+ * apps/app_meetme.c, main/ccss.c, include/asterisk/data.h,
+ res/res_jabber.c, res/res_config_sqlite.c,
+ include/asterisk/callerid.h, channels/chan_dahdi.c,
+ include/asterisk/bridging_technology.h,
+ include/asterisk/doxyref.h, include/asterisk/event.h,
+ include/asterisk/astmm.h, main/ast_expr2f.c, main/features.c,
+ include/asterisk/timing.h, include/asterisk/rtp_engine.h,
+ include/asterisk/ccss.h, include/asterisk/threadstorage.h,
+ include/asterisk/xml.h, main/pbx.c, channels/chan_sip.c,
+ include/asterisk/astobj2.h, include/asterisk/channel.h,
+ include/asterisk/calendar.h, include/asterisk/manager.h,
+ include/asterisk/features.h, include/asterisk/logger.h,
+ include/asterisk/http.h, channels/sig_pri.h,
+ include/asterisk/app.h, main/audiohook.c, include/asterisk/pbx.h,
+ include/asterisk/dnsmgr.h, include/asterisk/smdi.h,
+ apps/app_voicemail.c: Fix some doxygen warnings. (closes issue
+ #17336) Reported by: snuffy Patches: doxygen-fixes1.diff uploaded
+ by snuffy (license 35) Tested by: russell
+
+2010-06-08 06:57 +0000 [r268896-268933] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_sqlite.c: Release list lock before returning on
+ error.
+
+ * utils/extconf.c: Fix trunk build on Mac OS X.
+
+2010-06-08 05:29 +0000 [r268894] Terry Wilson <twilson@digium.com>
+
+ * channels/sip/sdp_crypto.c (added), res/res_rtp_asterisk.c,
+ main/global_datastores.c, main/rtp_engine.c,
+ include/asterisk/res_srtp.h (added), channels/sip/srtp.c (added),
+ channels/chan_sip.c, include/asterisk/autoconfig.h.in,
+ res/res_srtp.exports.in (added), configure.ac, CHANGES,
+ channels/chan_iax2.c, res/res_srtp.c (added), main/channel.c,
+ build_tools/menuselect-deps.in, main/asterisk.exports.in,
+ configure, funcs/func_channel.c,
+ channels/sip/dialplan_functions.c,
+ channels/sip/include/sdp_crypto.h (added),
+ doc/tex/secure-calls.tex (added),
+ include/asterisk/global_datastores.h, channels/sip/include/srtp.h
+ (added), makeopts.in, include/asterisk/rtp_engine.h,
+ include/asterisk/frame.h, doc/tex/asterisk.tex,
+ channels/sip/include/sip.h: Add SRTP support for Asterisk After 5
+ years in mantis and over a year on reviewboard, SRTP support is
+ finally being comitted. This includes generic CHANNEL dialplan
+ functions that work for getting the status of whether a call has
+ secure media or signaling as defined by the underlying channel
+ technology and for setting whether or not a new channel being
+ bridged to a calling channel should have secure signaling or
+ media. See doc/tex/secure-calls.tex for examples. Original patch
+ by mikma, updated for trunk and revised by me. (closes issue
+ #5413) Reported by: mikma Tested by: twilson, notthematrix,
+ hemanshurpatel Review: https://reviewboard.asterisk.org/r/191/
+
+2010-06-08 00:45 +0000 [r268857] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sip/dialplan_functions.c: Make SIP tests compile again.
+
+2010-06-07 22:56 +0000 [r268817-268818] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Use the mailbox destructor function,
+ instead.
+
+ * channels/chan_sip.c, channels/sip/include/sip.h: Mailbox list
+ would previously grow at each reload, containing duplicates.
+ Also, optimize the allocation of mailboxes to avoid additional
+ memory structures. (closes issue #16320) Reported by: Marquis
+ Patches: 20100525__issue16320.diff.txt uploaded by tilghman
+ (license 14)
+
+2010-06-07 20:04 +0000 [r268774] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_ss7.h
+ (added), channels/Makefile, channels/sig_pri.c,
+ channels/sig_ss7.c (added): Extract sig_ss7 out of chan_dahdi.
+ Extract the SS7 specific code out of chan_dahdi like what was
+ done to ISDN/PRI and analog signaling. The new SS7 structures
+ were modeled on sig_pri. The changes to sig_pri are an
+ enhancement and a bug fix made possible because SS7 was
+ extracted. 1) The sig_pri TRANSFERCAPABILITY channel variable
+ should have been set unconditionally in
+ sig_pri_new_ast_channel(). 2) SS7/PRI transfer capability
+ interaction in dahdi_new() fixed because of SS7 extraction. 3)
+ Module ref count error in dahdi_new() if startpbx failed to start
+ the PBX for some reason. Review:
+ https://reviewboard.asterisk.org/r/661/
+
+2010-06-07 19:52 +0000 [r268773] Tilghman Lesher <tlesher@digium.com>
+
+ * main/rtp_engine.c, channels/chan_sip.c,
+ channels/sip/dialplan_functions.c, include/asterisk/rtp_engine.h:
+ Seems strange (and the code backs up) that if the max and min of
+ a statistic is expressed as a double, the last value would not
+ also need to be a double. (closes issue #15807) Reported by:
+ klaus3000
+
+2010-06-07 19:06 +0000 [r268734] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Moved AOC request code out of the middle of
+ code parsing the dialed number.
+
+2010-06-07 18:59 +0000 [r268731] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c: Event well was going dry. (issue #17234)
+
+2010-06-07 17:34 +0000 [r268690] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * main/dsp.c: Set threshold for silence detection defaults to 256
+ (closes issue #15685) Reported by: david_s5 Patches:
+ dsp-silence-threshold-init.diff uploaded by dant (license 670)
+ issue15685.patch.v5 uploaded by pabelanger (license 224) Tested
+ by: danti Review: https://reviewboard.asterisk.org/r/670/
+
+2010-06-07 17:14 +0000 [r268653] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_smdi.c: Avoid unloading res_smdi twice. (closes issue
+ #17237) Reported by: pabelanger
+
+2010-06-07 15:51 +0000 [r268578] Richard Mudgett <rmudgett@digium.com>
+
+ * main/file.c: Suppress warning in waitstream_core(). Suppress the
+ warning about unexpected control subclass frames for
+ AST_CONTROL_CONNECTED_LINE, AST_CONTROL_REDIRECTING, and
+ AST_CONTROL_AOC in file.c:waitstream_core().
+
+2010-06-06 05:29 +0000 [r268454-268534] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/rc.redhat.asterisk: Take advantage of variable
+ substitution already in the Makefile to specify the correct
+ location for files in init.d. (closes issue #16979) Reported by:
+ jw-asterisk (issue #15691) Reported by: itamarjp
+
+ * channels/chan_iax2.c: Finally track down and eliminate the
+ "FRACK! warnings from chan_iax2".
+
+ * main/dsp.c: Fix crash in DTMF detection. What I did not
+ originally see in my previous commit was that even though the
+ next digit could be detected before the previous was considered
+ ended, the detection of the next digit effectively ends the
+ detection of the previous. Therefore, the length moves in
+ lockstep with the digit, and no separate counter is needed for
+ the length alone. (closes issue #17371) Reported by: alecdavis
+ (closes issue #17474) Reported by: kenner
+
+ * main/manager.c: Verify event is not NULL before attempting to
+ lower its usecount. (closes issue #17234) Reported by: mav3rick
+
+2010-06-05 05:23 +0000 [r268395-268417] Kevin P. Fleming <kpfleming@digium.com>
+
+ * CHANGES: Typo fix.
+
+ * CHANGES: Grammatical error fix.
+
+2010-06-05 02:51 +0000 [r268321] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configs/voicemail.conf.sample: Merged revisions 268320 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r268320 | tilghman | 2010-06-04 21:49:52 -0500 (Fri, 04 Jun 2010)
+ | 3 lines Rest In Peace
+ http://www.outandaboutnewspaper.com/article/4061 ........
+
+2010-06-04 22:37 +0000 [r268205-268281] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes compile error from uninitialized
+ variable
+
+ * channels/chan_sip.c: RFC3261 compliant sip unreliable retransmit
+ timing + 'registerattempts' option tweak Changes. 1. RFC 3261
+ states in section 17.1.2.2 and 17.1.1.2 that retransmission
+ timers should initially be set to timer T1. T1 by default is
+ 500ms. Asterisk was starting the retransmission timers at T1*2
+ which shouldn't cause any problems, but is not RFC compliant. 2.
+ RFC 3261 states in section 17.1.2.2 that for a non-INVITE client
+ transaction, if the retransmit timer fires while in the
+ proceeding state that the request must be retransmitted. Asterisk
+ currently ack's requests for both INVITE and non-INVITE
+ transactions when a 1XX response is received, this patch changes
+ this for non-INVITE requests. 3. The 'registerattempts' option in
+ sip.conf is supposed to set how many registry attempts will be
+ made before giving up. When this option is set to 1, I would
+ expect only one registry attempt to be made before stopping
+ because of a failure, but instead two are made. In my opinion
+ this is not expected behavior. This option does not indicate that
+ these are re-attempts. The logic behind this option has been
+ changed to only attempt registers the exact number of times this
+ option is set to. If this option is 0, it still continues to
+ re-attempt the registration forever. Review:
+ https://reviewboard.asterisk.org/r/687/
+
+2010-06-04 20:42 +0000 [r267972-268127] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 268126 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r268126 | tilghman | 2010-06-04 15:41:24 -0500 (Fri, 04
+ Jun 2010) | 2 lines AC_CONFIG_SUBDIRS has a bad side-effect on
+ cross-compiles. ........
+
+ * Makefile, /, makeopts.in: Merged revisions 268050 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r268050 | tilghman | 2010-06-04 14:38:57 -0500 (Fri, 04
+ Jun 2010) | 6 lines Build menuselect with the build environment's
+ compiler, not the host (target)'s compiler. (closes issue #17464)
+ Reported by: pprindeville Tested by: tilghman ........
+
+ * /, configure, configure.ac, autoconf/libcurl.m4: Merged revisions
+ 267971 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r267971 | tilghman | 2010-06-04 11:27:02 -0500 (Fri, 04 Jun 2010)
+ | 2 lines As-fixiate the build process ........
+
+2010-06-04 14:45 +0000 [r267928] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Incoming overlap dialing no longer works
+ after sig_pri extraction. The problem would manifest itself if
+ your dialplan matching could accept more digits to match than
+ were actually dialed. The time out waiting for overlap digits
+ disconnected the call instead of matching any accumulated digits
+ to the dialplan. Accidental conversion of a break out of loop as
+ a break out of switch. (closes issue #17401) Reported by:
+ avalentin Patches: issue17401_digit_timeout.patch uploaded by
+ rmudgett (license 664) Tested by: avalentin, rmudgett
+
+2010-06-04 03:20 +0000 [r267877] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/slin.h: As signed linear audio data is accessed
+ as 16-bit values, certain processors require the values to be
+ aligned in memory. (closes issue #16912) Reported by:
+ michaelevdokimov Patches: asterisk.patch uploaded by
+ michaelevdokimov (license 997) Tested by: michaelevdokimov
+
+2010-06-04 03:11 +0000 [r267863] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Send an ACK for every final response
+ received for an INVITE From issue ABE-2247. RFC 3261 compliance
+ for sections 13.2.24 and 17.1.1.2. Review:
+ https://reviewboard.asterisk.org/r/692/
+
+2010-06-04 02:58 +0000 [r267775-267862] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/slin.h: As signed linear audio data is accessed
+ as 16-bit values, certain processors require the values to be
+ aligned in memory. (closes issue #16912) Reported by:
+ michaelevdokimov
+
+ * configure, autoconf/ast_ext_lib.m4: If there's a default, turn it
+ on, even when the option isn't specified.
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 267759 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r267759 | tilghman | 2010-06-03 20:16:26 -0500 (Thu, 03 Jun 2010)
+ | 7 lines Make the default install path appear to be /usr on
+ Linux, instead of /usr/local. Also, reorganize the options, so
+ that they're more alphabetical. (closes issue #17013) Reported
+ by: klaus3000 ........
+
+2010-06-03 20:41 +0000 [r267714] Russell Bryant <russell@digium.com>
+
+ * main/ccss.c: Remove a LOG_WARNING. This came up when using the
+ sample configs, and just indicates expected behavior.
+
+2010-06-03 19:46 +0000 [r267669] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_odbc.c: Handle OOM errors more gracefully. (closes
+ issue #17084) Reported by: falves11 Patches:
+ issue17084_162_A.diff uploaded by falves11 (license 374) Tested
+ by: falves11
+
+2010-06-03 18:53 +0000 [r267624] Leif Madsen <lmadsen@digium.com>
+
+ * UPGRADE.txt, CHANGES: Update UPGRADE.txt and CHANGE for CDR
+ functionality changes. Updated the UPGRADE.txt and CHANGES file
+ stating that CDR records will not be explicity written unless
+ cdr.conf exists and is configured. (closes issue #17373) Reported
+ by: wdoekes Tested by: pabelanger
+
+2010-06-03 18:38 +0000 [r267622] Richard Mudgett <rmudgett@digium.com>
+
+ * codecs/codec_dahdi.c: Make compile again.
+
+2010-06-03 17:31 +0000 [r267537] Russell Bryant <russell@digium.com>
+
+ * channels/chan_usbradio.c: Don't stop Asterisk if chan_usbradio
+ isn't configured.
+
+2010-06-03 17:09 +0000 [r267492] Mark Michelson <mmichelson@digium.com>
+
+ * codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_adpcm.c,
+ codecs/codec_alaw.c, main/translate.c, codecs/codec_g726.c,
+ codecs/codec_gsm.c, codecs/codec_ulaw.c, codecs/codec_dahdi.c,
+ include/asterisk/translate.h: Remove unnecessary code relating to
+ PLC. The logic for handling generic PLC is now handled in
+ ast_write in channel.c instead of in translation code. Review:
+ https://reviewboard.asterisk.org/r/683/
+
+2010-06-03 17:05 +0000 [r267445-267490] Russell Bryant <russell@digium.com>
+
+ * channels/chan_usbradio.c: Remove a line that was killing Asterisk
+ on startup.
+
+ * channels/h323/Makefile.in: Comment out a rule that likes to run
+ implicitly unnecessarily, breaking builds
+
+2010-06-03 00:02 +0000 [r267399] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, CHANGES,
+ channels/sig_pri.c: Add ETSI Message Waiting Indication (MWI)
+ support. Add the ability to report waiting messages to ISDN
+ endpoints (phones). Relevant specification: EN 300 650 and EN 300
+ 745 Review: https://reviewboard.asterisk.org/r/599/
+
+2010-06-02 22:46 +0000 [r267352] Russell Bryant <russell@digium.com>
+
+ * channels/Makefile, channels/h323/Makefile.in: try to fix some
+ random chan_h323 compilation failures After some debugging, the
+ random chan_h323 build failures appear to be due to complications
+ introduced by some chan_h323 specific build stuff getting
+ triggered during a clean. Simplify this by moving the h323 clean
+ commands down into channels/makefile.
+
+2010-06-02 22:28 +0000 [r267350] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac, include/asterisk/channel.h, CHANGES,
+ channels/sig_pri.c: Add ETSI Malicious Call ID support. Add the
+ ability to report malicious callers as an AMI event in the call
+ event class. Relevant specification: EN 300 180 Review:
+ https://reviewboard.asterisk.org/r/576/
+
+2010-06-02 21:44 +0000 [r267303-267305] Russell Bryant <russell@digium.com>
+
+ * utils/extconf.c: Fix a build error on mac.
+
+ * main/Makefile: Ensure the -Wno-strict-aliasing flag makes it,
+ even if ASTCFLAGS has been specified. When ASTCFLAGS was
+ specified with the make command, Makefile.rules was using the
+ specified value from the command line and not the one here,
+ making it so this flag would go missing.
+
+2010-06-02 21:05 +0000 [r267261] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, CHANGES,
+ channels/sig_pri.c: Add ETSI Call Waiting support. Add the
+ ability to announce a call to an endpoint when there are no B
+ channels available. A call waiting call is a SETUP message with
+ no B channel selected. Relevant specification: EN 300 056, EN 300
+ 057, EN 300 058 For DAHDI/ISDN channels, the CHANNEL() dialplan
+ function now supports the "no_media_path" option. * Returns "0"
+ if there is a B channel associated with the call. * Returns "1"
+ if no B channel is associated with the call. The call is either
+ on hold or is a call waiting call. If you are going to allow
+ incoming call waiting calls then you need to use
+ CHANNEL(no_media_path) do determine if you must drop a call to
+ accept the new call. Review:
+ https://reviewboard.asterisk.org/r/568/
+
+2010-06-02 19:33 +0000 [r267181] David Vossel <dvossel@digium.com>
+
+ * CHANGES, doc/advice_of_charge.txt: Update CHANGES and aoc help
+ doc to reflect AOC additions
+
+2010-06-02 18:53 +0000 [r267138] Russell Bryant <russell@digium.com>
+
+ * main/cli.c: Add a CLI command that blocks until Asterisk has
+ fully booted. Review: https://reviewboard.asterisk.org/r/684/
+
+2010-06-02 18:13 +0000 [r267097] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Prevent use of uninitialized values. Two
+ struct sockaddr_ins are created when applying directmedia host
+ access rules. The addresses of these are passed to the RTP engine
+ to be filled in. However, the RTP engine inspects the fields of
+ the structs before actually taking action. This inspection caused
+ valgrind to be a bit unhappy.
+
+2010-06-02 18:10 +0000 [r267096] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, configs/chan_dahdi.conf.sample,
+ include/asterisk/aoc.h (added), channels/chan_sip.c,
+ configs/manager.conf.sample, main/aoc.c (added),
+ apps/app_queue.c, channels/sig_pri.c, doc/advice_of_charge.txt
+ (added), main/channel.c, channels/sig_pri.h,
+ channels/chan_dahdi.c, main/manager.c, main/features.c,
+ tests/test_aoc.c (added), configs/sip.conf.sample,
+ include/asterisk/frame.h, main/asterisk.c,
+ channels/sip/include/sip.h: Generic Advice of Charge. Asterisk
+ Generic AOC Representation - Generic AOC encode/decode routines.
+ (Generic AOC must be encoded to be passed on the wire in the
+ AST_CONTROL_AOC frame) - AST_CONTROL_AOC frame type to represent
+ generic encoded AOC data - Manager events for AOC-S, AOC-D, and
+ AOC-E messages Asterisk App Support - app_dial AOC-S pass-through
+ support on call setup - app_queue AOC-S pass-through support on
+ call setup AOC Unit Tests - AOC Unit Tests for encode/decode
+ routines - AOC Unit Test for manager event representation. SIP
+ AOC Support - Pass-through of generic AOC-D and AOC-E messages to
+ snom phones via the snom AOC specification. - Creation of
+ chan_sip page3 flags for the addition of the new
+ 'snom_aoc_enabled' sip.conf option. IAX AOC Support - Natively
+ supports AOC pass-through through the use of the new
+ AST_CONTROL_AOC frame type DAHDI AOC Support - ETSI PRI full AOC
+ Pass-through support - 'aoc_enable' chan_dahdi.conf option for
+ independently enabling pass-through of AOC-S, AOC-D, AOC-E. -
+ 'aoce_delayhangup' option for retrieving AOC-E on disconnect. -
+ DAHDI A() dial string option for requesting AOC services. example
+ usage: ;requests AOC-S, AOC-D, and AOC-E on call setup
+ exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e)) Review:
+ https://reviewboard.asterisk.org/r/552/
+
+2010-06-02 17:57 +0000 [r267093] Russell Bryant <russell@digium.com>
+
+ * apps/app_voicemail.c: Silence a compiler warning.
+
+2010-06-02 17:29 +0000 [r267065] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/slin.h: Fix infinite loop when loading codec
+ speex This changes the sample slinear frame data to contain
+ non-zero data so that translation calculations for speex works
+ when preprocessing and VAD is turned on. The encoder expects
+ samples to be returned, but when attempted with the mentioned two
+ options and silent sample frames everything was discarded.
+ (closes issue #17240) Reported by: seandarcy Review:
+ https://reviewboard.asterisk.org/r/682/
+
+2010-06-02 17:25 +0000 [r267041] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, main/ast_expr2.y: Merged revisions 267009 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r267009 | pabelanger | 2010-06-02 13:14:37 -0400 (Wed, 02 Jun
+ 2010) | 7 lines Cleanup error/warning messages in AEL2 parser
+ (closes issue #16684) Reported by: Silmaril Patches:
+ patch_ael2_logmsg.diff uploaded by Silmaril (license 979)
+ ........
+
+2010-06-02 17:13 +0000 [r266926-267008] Richard Mudgett <rmudgett@digium.com>
+
+ * main/manager.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac, configs/manager.conf.sample, CHANGES,
+ channels/sig_pri.c, include/asterisk/manager.h: Add ETSI Advice
+ Of Charge (AOC) event reporting. This feature generates AMI
+ events in the new aoc event class from the events passed up by
+ libpri. Review: https://reviewboard.asterisk.org/r/537/
+
+ * channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, CHANGES,
+ channels/sig_pri.c: Add ETSI Explicit Call Transfer (ECT)
+ support. Added ability to send and receive ETSI Explicit Call
+ Transfer (ECT) messages to eliminate tromboned calls. Note:
+ Asterisk already supported initiating the transfer of calls to
+ eliminate tromboned calls to libpri so there was nothing to do
+ for the asterisk portion. Review:
+ https://reviewboard.asterisk.org/r/520/
+
+2010-06-02 13:32 +0000 [r266877] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * main/bridging.c: pthread_join to assure the thread is really gone
+ (closes issue #15465) Reported by: fnordian Patches:
+ bridging.patch uploaded by fnordian (license 110) Tested by:
+ lmadsen, fnordian, peterh Review:
+ https://reviewboard.asterisk.org/r/679/
+
+2010-06-01 22:14 +0000 [r266832] Terry Wilson <twilson@digium.com>
+
+ * res/res_calendar_exchange.c: Use the correct ical.h file
+
+2010-06-01 21:28 +0000 [r266828] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, tests/test_locale.c
+ (added), configure.ac, configs/voicemail.conf.sample,
+ include/asterisk/localtime.h, main/stdtime/localtime.c, CHANGES,
+ apps/app_voicemail.c: Support setting locale per-mailbox (changes
+ date/time languages for email, pager messages). (closes issue
+ #14333) Reported by: klaus3000 Patches:
+ 20090515__issue14333.diff.txt uploaded by tilghman (license 14)
+ app_voicemail.c-svn-trunk-rev211675-patch.txt uploaded by
+ klaus3000 (license 65) Tested by: klaus3000
+
+2010-06-01 21:12 +0000 [r266786] Terry Wilson <twilson@digium.com>
+
+ * apps/app_dial.c, UPGRADE.txt: Set app and appdata fields when a
+ Dial is redirected (closes issue #17204) Reported by: one47
+ Tested by: twilson, one47
+
+2010-06-01 18:02 +0000 [r266592-266735] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_smdi.c: Don't register functions until the last possible
+ point, so they're not unloaded unnecessarily. (closes issue
+ #15996) Reported by: junky Patches: sdmi_wait.diff uploaded by
+ junky (license 177)
+
+ * main/manager.c: Eliminate stale manager events after a set
+ interval, even if AMI clients don't query for them. Actions (or
+ failures to act) by external clients should not cause memory
+ leaks in Asterisk, especially when those continued leaks could
+ cause Asterisk to misbehave later. (closes issue #17234) Reported
+ by: mav3rick Patches: 20100510__issue17234.diff.txt uploaded by
+ tilghman (license 14) 20100517__issue17234__trunk.diff.txt
+ uploaded by tilghman (license 14) Tested by: mav3rick, davidw
+ (closes issue #17365) Reported by: davidw
+
+ * /, main/asterisk.c: Merged revisions 266585 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r266585 | tilghman | 2010-06-01 10:17:46 -0500 (Tue, 01 Jun 2010)
+ | 11 lines Prevent CLI prompt from distorting output of lines
+ shorter than the prompt. Uses the VT100 method of clearing the
+ line from the cursor position to the end of the line: Esc-0K
+ (closes issue #17160) Reported by: coolmig Patches:
+ 20100531__issue17160.diff.txt uploaded by tilghman (license 14)
+ Tested by: coolmig ........
+
+2010-05-30 20:18 +0000 [r266438-266522] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_env.c: Needs to be wrapped in <para>
+
+ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 266437 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r266437 | tilghman | 2010-05-29 23:43:28 -0500 (Sat, 29 May 2010)
+ | 2 lines Reverting patch and reopening issue #16784, as patch
+ breaks color display. ........
+
+2010-05-28 22:54 +0000 [r266386] Terry Wilson <twilson@digium.com>
+
+ * res/res_calendar_icalendar.c, configure, configure.ac,
+ res/res_calendar_caldav.c: Fix ical library handling (again)
+ Newer versions of libical (which we require) store the header
+ file in a libical/ subfolder and include an ical.h file that does
+ a #warning for deprecation and then #includes <libical/ical.h>.
+ Since we now test for libical/ical.h, we can change the #includes
+ back to <libical/ical.h> and remove the test which specifically
+ adds /usr/include/libical as an include directory.
+
+2010-05-28 22:50 +0000 [r266337-266385] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_env.c, UPGRADE.txt, main/asterisk.c: Setup environment
+ variables for the benefit of child processes and disallow
+ changing them. (closes issue #14899) Reported by: jmls Patches:
+ 20090916__issue14899.diff.txt uploaded by tilghman (license 14)
+ Tested by: jmls
+
+ * main/asterisk.c: Only report swap on platforms which can examine
+ those statistics
+
+2010-05-28 17:55 +0000 [r266292] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes crash when creation of UDPTL fails
+ (closes issue #17264) Reported by: falves11 Patches:
+ issue_17264_reviewboard_fix.diff uploaded by dvossel (license
+ 671) issue_17264_1.6.2_reviewboard_fix.diff uploaded by dvossel
+ (license 671) Tested by: falves11
+
+2010-05-28 17:34 +0000 [r266289] Terry Wilson <twilson@digium.com>
+
+ * configure, configure.ac, makeopts.in: More build fixes for
+ ical/neon and res_calendar_ews
+
+2010-05-27 20:08 +0000 [r266240] Jeff Peeler <jpeeler@digium.com>
+
+ * pbx/pbx_realtime.c: fix compile error
+
+2010-05-27 19:25 +0000 [r266146-266238] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_realtime.c, CHANGES: Cache query results for one second.
+ Queries from the PBX core come in 3's. Caching avoids the
+ additional performance penalty from those two additional queries
+ hitting the database. (closes issue #16521) Reported by: tilghman
+ Patches: 20091229__issue16521.diff.txt uploaded by tilghman
+ (license 14) Tested by: Hubguru, tilghman
+
+ * /, main/logger.c, utils/extconf.c, main/asterisk.c: Merged
+ revisions 266142 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r266142 | tilghman | 2010-05-26 16:11:44 -0500 (Wed, 26 May 2010)
+ | 14 lines Use sigaction for signals which should persist past
+ the initial trigger, not signal. If you call signal() in a
+ Solaris signal handler, instead of just resetting the signal
+ handler, it causes the signal to refire, because the signal is
+ not marked as handled prior to the signal handler being called.
+ This effectively causes Solaris to immediately exceed the
+ threadstack in recursive signal handlers and crash. (closes issue
+ #17000) Reported by: rmcgilvr Patches:
+ 20100526__issue17000.diff.txt uploaded by tilghman (license 14)
+ Tested by: rmcgilvr ........
+
+2010-05-26 20:17 +0000 [r266092-266098] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c: Remove redundant ast_conntected_line_free call.
+ This wouldn't cause any problems, but it's certainly not needed
+ either.
+
+ * res/res_musiconhold.c: Remove unrelated MOH change from previous
+ commit. Thanks Kevin!
+
+ * main/channel.c, res/res_musiconhold.c: Fix misspelling of macro
+ args.
+
+2010-05-26 19:46 +0000 [r266006-266090] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c, main/app.c, channels/sip/config_parser.c,
+ channels/sip/include/sip.h: do all sip registry parsing before
+ transmit_register This patch breaks up every part of the sip
+ registry string during config parsing and removes all parsing
+ from transmit_register(). Thanks to Nick_Lewis for contributing
+ this patch! (closes issue #14331) Reported by: Nick_Lewis
+ Patches: chan_sip.c-domparse.patch uploaded by Nick Lewis
+ (license 657) chan_sip.c.patch uploaded by Nick Lewis (license
+ 657) chan_sip.c.domainparse3.patch uploaded by Nick Lewis
+ (license 657) chan_sip.c-domparse4.patch uploaded by Nick Lewis
+ (license 657) chan_sip.c-domparse5.patch uploaded by Nick Lewis
+ (license 657) nicklewispatch.diff uploaded by dvossel (license
+ 671) Tested by: Nick_Lewis, dvossel Review:
+ https://reviewboard.asterisk.org/r/628/
+
+ * channels/chan_sip.c: fixes failed SIP Directed pickup resulting
+ in dead channel (closes issue #17339) Reported by: one47 Patches:
+ sip_magic_pickup2 uploaded by one47 (license 23) Tested by:
+ one47, dvossel
+
+2010-05-26 16:23 +0000 [r265894-265923] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_pgsql.c, /: Merged revisions 265910 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r265910 | tilghman | 2010-05-26 11:21:00 -0500 (Wed, 26
+ May 2010) | 7 lines Not finding rows in the DB does not rise to
+ the level of a warning. (closes issue #17062) Reported by:
+ drookie Patches: 20100525__issue17062.diff.txt uploaded by
+ tilghman (license 14) ........
+
+ * res/res_config_pgsql.c, configs/res_pgsql.conf.sample: Construct
+ socket name, according to the Postgres docs, and document as
+ such. (closes issue #17392) Reported by: dps Patches:
+ 20100525__issue17392.diff.txt uploaded by tilghman (license 14)
+ Tested by: dps
+
+2010-05-26 14:45 +0000 [r265842-265844] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: .......
+
+ * channels/chan_sip.c: Re-enable "always" option for videosupport
+ option in sip.conf. (closes issue #17016) Reported by: twilson
+ Patches: 17016.patch uploaded by mmichelson (license 60) Tested
+ by: devmod
+
+2010-05-26 05:33 +0000 [r265793] Terry Wilson <twilson@digium.com>
+
+ * build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ res/res_calendar_ews.c: Ensure that libneon > 0.29.0 is installed
+ for res_calendar_ews This uses a modified version of pabelanger's
+ patch that checks for NTLM support instead, which was added in
+ 0.29.0 which is what is required for res_calendar_ews. (closes
+ issue #17391) Reported by: loloski Patches: issue17391.patch.v2
+ uploaded by pabelanger (license 224) Tested by: twilson
+
+2010-05-26 00:29 +0000 [r265747] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
+ configure, include/asterisk/autoconfig.h.in, configure.ac,
+ pbx/pbx_lua.c, res/res_calendar_caldav.c, res/res_calendar_ews.c:
+ Use configure to determine the prefixes and include directories
+ properly. This ensures cross-platform compatibility, even among
+ Linux distributions, which don't always put headers in the same
+ place. (closes issue #17391) Reported by: loloski
+
+2010-05-25 20:59 +0000 [r265698] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Properly use peer's outboundproxy for
+ outbound REGISTERs. The logic used in transmit_register to get
+ the outboundproxy for a peer was flawed since this value would be
+ overridden shortly afterwards when create_addr was called. In
+ addition, this also fixes some logic used when parsing users.conf
+ so that the peer name is placed in the internally-generated
+ register string so that an outboundproxy set in the Asterisk GUI
+ will be used for outbound REGISTERs.
+
+2010-05-25 17:00 +0000 [r265611] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 265610 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May
+ 2010) | 8 lines Don't mark the cdr records of unanswered queue
+ calls with "NOANSWER". This restores the behavior prior to
+ r258670. (closes issue #17334) Reported by: jvandal Patches:
+ queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested
+ by: aragon, jvandal ........
+
+2010-05-25 16:23 +0000 [r265608] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Memory leak in connected line data when SIP blond
+ transfer done. The handling of the control subclass
+ AST_CONTROL_READ_ACTION frame leaked connected line string memory
+ in __ast_read(). Also in __ast_read() the frame type switch
+ should not have had a case for AST_CONTROL_READ_ACTION.
+ AST_CONTROL_READ_ACTION is not a frame type.
+
+2010-05-25 08:31 +0000 [r265525] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * addons/ooh323c/src/oochannels.c: Typos: 'succesful' (lintian)
+
+2010-05-24 22:21 +0000 [r265467] Terry Wilson <twilson@digium.com>
+
+ * doc/manager_1_1.txt, main/manager.c, main/asterisk.c: Merge the
+ rest of the FullyBooted patch
+
+2010-05-24 22:16 +0000 [r265449-265453] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_senddtmf.c: Allow SendDTMF to play digits to a specified
+ channel. Patch supplied by reporter was modified to use
+ autoservice and prevent a potential channel ref leak but is
+ otherwise as the reporter uploaded it. (closes issue #17182)
+ Reported by: rcasas Patches: app_senddtmf.c.patch_trunk uploaded
+ by rcasas (license 641)
+
+ * channels/h323/ast_h323.cxx: Print openh323 log to the Asterisk
+ console. (closes issue #17109) Reported by: under Patches:
+ logstream.diff uploaded by under (license 914)
+
+ * channels/chan_sip.c: Allow type=user SIP endpoints to be loaded
+ properly from realtime. (closes issue #16021) Reported by:
+ Guggemand Patches: realtime-type-fix.patch uploaded by Guggemand
+ (license 897) (altered by me slightly to avoid ref leaks) Tested
+ by: Guggemand
+
+2010-05-24 20:08 +0000 [r265367] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_rpt.c: Make app_rpt.c able to compile again.
+
+2010-05-24 19:42 +0000 [r265366] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: reverses incorrect logic introduced by
+ r243200 The decoding of the replace_id did not need to be broken
+ up in this instance. This was brought to my attention again
+ because it caused a segfault when the from or to tags were not
+ present in the "Replaces" header.
+
+2010-05-24 19:06 +0000 [r265317-265320] Terry Wilson <twilson@digium.com>
+
+ * doc/tex/manager.tex: Add the FullyBooted AMI event It is possible
+ to connect to the manager interface before all Asterisk modules
+ are loaded. To ensure that an application does not send AMI
+ actions that might require a module that has not yet loaded, the
+ application can listen for the FullyBooted manager event. It will
+ be sent upon connection if all modules have been loaded, or as
+ soon as loading is complete. The event: Event: FullyBooted
+ Privilege: system,all Status: Fully Booted Review:
+ https://reviewboard.asterisk.org/r/639/
+
+ * CREDITS, configs/calendar.conf.sample, CHANGES,
+ res/res_calendar_ews.c (added), res/res_calendar.c: Calendaring
+ support for Exchange Server 2007+ via EWS This commit adds
+ support for calendaring with Exchange Server 2007+ via Exchange
+ Web Services. Full write support and for querying attendees. Many
+ thanks to Jan Kaláb for the feature. (closes issue #17022)
+ Reported by: pitel Patches: res_calendar_ews.c uploaded by pitel
+ (license 1008) Tested by: pitel, twilson Review:
+ https://reviewboard.asterisk.org/r/557/ Review:
+ https://reviewboard.asterisk.org/r/668/
+
+2010-05-24 18:19 +0000 [r265316] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c: On systems with a LOT of RAM, a signed integer
+ sometimes printed negative. (closes issue #16837) Reported by:
+ jlpedrosa Patches: 20100504__issue16837.diff.txt uploaded by
+ tilghman (license 14)
+
+2010-05-24 16:10 +0000 [r265273] David Vossel <dvossel@digium.com>
+
+ * main/channel.c: fixes segfault when using generic plc
+
+2010-05-23 18:23 +0000 [r265227] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c: small changes to avoiding 'freeing unused
+ memory...'
+
+2010-05-21 22:46 +0000 [r265174] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Channel initialization failure causes crashes.
+ __ast_channel_alloc_ap() has several points in the initialization
+ of a new channel structure where it could fail. Since the channel
+ structure is now an ao2 object, the destructor callback needs to
+ be able to handle clean up when the structure setup is
+ incomplete. Problems corrected: 1) Failing to setup the alertpipe
+ would not unreference the structure but free it directly. Doing
+ this to an ao2_object is very bad. 2) File descriptors need to be
+ initialized to -1 before a construction failure could occur so
+ the destructor will not close unopened descriptors. 3) The
+ destructor needs to check that the string field has been
+ initialized before using any string field values. Crashes
+ expected. 4) The destructor should not notify devstate if the
+ device name is empty. It is a waste of cycles and a couple ERROR
+ log messages are generated. Review:
+ https://reviewboard.asterisk.org/r/675/
+
+2010-05-21 21:08 +0000 [r264953-265090] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/file.h, /, apps/app_queue.c: Merged revisions
+ 265089 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May
+ 2010) | 8 lines Don't hang up on a queue caller if the file we
+ attempt to play does not exist. This also fixes a documentation
+ mistake in file.h that made my original attempt to correct this
+ problem not work correctly. (closes issue #17061) Reported by:
+ RoadKill ........
+
+ * channels/chan_sip.c: Be sure to set the sin_family on the proxy
+ when allocating. (closes issue #17157) Reported by: stuarth
+
+ * /, include/asterisk/channel.h: Merged revisions 264999 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r264999 | mmichelson | 2010-05-21 11:53:53 -0500 (Fri, 21 May
+ 2010) | 3 lines Fix grammatical error in comment. ........
+
+ * main/channel.c, main/autoservice.c, /,
+ include/asterisk/channel.h: Merged revisions 264996 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri,
+ 21 May 2010) | 32 lines Allow ast_safe_sleep to defer specific
+ frames until after the sleep has concluded. From reviewboard
+ Background: A Digium customer discovered a somewhat odd bug. The
+ setup is that parties A and B are bridged, and party A places
+ party B on hold. While party B is listening to hold music, he
+ mashes a bunch of DTMF. Party A takes party B off hold while this
+ is happening, but party B continues to hear hold music. I could
+ reproduce this about 1 in 5 times. The issue: When DTMF features
+ are enabled and a user presses keys, the channel that the DTMF is
+ streamed to is placed in an ast_safe_sleep for 100 ms, the
+ duration of the emulated tone. If an AST_CONTROL_UNHOLD frame is
+ read from the channel during the sleep, the frame is dropped.
+ Thus the unhold indication is never made to the channel that was
+ originally placed on hold. The fix: Originally, I discussed with
+ Kevin possible ways of fixing the specific problem reported.
+ However, we determined that the same type of problem could happen
+ in other situations where ast_safe_sleep() is used. Using
+ autoservice as a model, I modified ast_safe_sleep_conditional()
+ to defer specific frame types so they can be re-queued once the
+ sleep has finished. I made a common function for determining if a
+ frame should be deferred so that there are not two identical
+ switch blocks to maintain. Review:
+ https://reviewboard.asterisk.org/r/674/ ........
+
+ * res/res_fax.c, include/asterisk/res_fax.h,
+ res/res_fax.exports.in, res/res_fax_spandsp.c: Log spandsp's fax
+ debug output to the FAX logger level. Review:
+ https://reviewboard.asterisk.org/r/658
+
+2010-05-21 01:00 +0000 [r264905] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Take dup'd code for directmedia ACLs and
+ make utility func The same code was repeated in lots of different
+ places, so I made a utility fuction for it. This should make the
+ merge in the v6-new branch easier.
+
+2010-05-20 23:29 +0000 [r264828] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/callerid.c: Merged revisions 264820 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r264820 | rmudgett | 2010-05-20 18:23:21 -0500 (Thu, 20 May 2010)
+ | 6 lines ast_callerid_parse() had a path that left name
+ uninitialized. Several callers of ast_callerid_parse() do not
+ initialize the name parameter before calling thus there is the
+ potential to use an uninitialized pointer. ........
+
+2010-05-20 22:23 +0000 [r264752-264779] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Let ExtensionState resolve dynamic hints. (closes
+ issue #16623) Reported by: tilghman Patches:
+ 20100116__issue16623.diff.txt uploaded by tilghman (license 14)
+ Tested by: lmadsen
+
+ * apps/app_stack.c: Error message fix. (closes issue #17356)
+ Reported by: kenner Patches: app_stack.c.diff uploaded by kenner
+ (license 1040)
+
+2010-05-20 20:49 +0000 [r264669-264711] Richard Mudgett <rmudgett@digium.com>
+
+ * main/ccss.c: Avoid crash in generic CC agent init if caller name
+ or number is NULL.
+
+ * apps/app_dial.c, apps/app_queue.c: Dial and queue connected line
+ update macro not always run when expected. The connected line
+ update macro would not get run if the connected line number
+ string was empty. The number could be empty if the connected line
+ update did not update a number but the name. It should be run if
+ there was an AST_CONTROL_CONNECTED_LINE frame received for
+ pending dials and queues. Renamed and added some more comments
+ for some confusing identifiers directly connected to the related
+ code. Also fixed a memory leak in app_queue. Review:
+ https://reviewboard.asterisk.org/r/669/
+
+2010-05-20 17:54 +0000 [r264626] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
+ channels/sip/include/sip.h: Add support for direct media ACLs
+ directmediapermit/directmediadeny support to restrict which peers
+ can do directmedia based on ip address. In some networks not all
+ phones are fully routed, i.e. not all phones can ping each other.
+ This patch adds a way to restrict directmedia for certain peers
+ between certain networks. (closes issue #16645) Reported by:
+ raarts Patches: directmediapermit.patch uploaded by raarts
+ (license 937) Tested by: raarts Review:
+ https://reviewboard.asterisk.org/r/467/
+
+2010-05-20 15:30 +0000 [r264497-264540] Kevin P. Fleming <kpfleming@digium.com>
+
+ * addons/ooh323c/src/h323, addons/ooh323c/src: Ignore pre-processed
+ source files generated during DONT_OPTIMIZE dev-mode builds.
+
+ * main/logger.c: Correct 'all logger levels' patch to work
+ properly. Nick Lewis pointed out that the patch as committed
+ wouldn't actually include dynamic logger levels, which was missed
+ by the other reviewers. Thanks!
+
+2010-05-19 21:29 +0000 [r264452] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, channels/chan_sip.c, include/asterisk/_private.h,
+ include/asterisk/options.h, main/asterisk.c, main/loader.c: Fix
+ transcode_via_sln option with SIP calls and improve PLC usage.
+ From reviewboard: The problem here is a bit complex, so try to
+ bear with me... It was noticed by a Digium customer that generic
+ PLC (as configured in codecs.conf) did not appear to actually be
+ having any sort of benefit when packet loss was introduced on an
+ RTP stream. I reproduced this issue myself by streaming a file
+ across an RTP stream and dropping approx. 5% of the RTP packets.
+ I saw no real difference between when PLC was enabled or disabled
+ when using wireshark to analyze the RTP streams. After analyzing
+ what was going on, it became clear that one of the problems faced
+ was that when running my tests, the translation paths were being
+ set up in such a way that PLC could not possibly work as
+ expected. To illustrate, if packets are lost on channel A's read
+ stream, then we expect that PLC will be applied to channel B's
+ write stream. The problem is that generic PLC can only be done
+ when there is a translation path that moves from some codec to
+ SLINEAR. When I would run my tests, I found that every single
+ time, read and write translation paths would be set up on channel
+ A instead of channel B. There appeared to be no real way to
+ predict which channel the translation paths would be set up on.
+ This is where Kevin swooped in to let me know about the
+ transcode_via_sln option in asterisk.conf. It is supposed to work
+ by placing a read translation path on both channels from the
+ channel's rawreadformat to SLINEAR. It also will place a write
+ translation path on both channels from SLINEAR to the channel's
+ rawwriteformat. Using this option allows one to predictably set
+ up translation paths on all channels. There are two problems with
+ this, though. First and foremost, the transcode_via_sln option
+ did not appear to be working properly when I was placing a SIP
+ call between two endpoints which did not share any common
+ formats. Second, even if this option were to work, for PLC to be
+ applied, there had to be a write translation path that would go
+ from some format to SLINEAR. It would not work properly if the
+ starting format of translation was SLINEAR. The one-line change
+ presented in this review request in chan_sip.c fixed the first
+ issue for me. The problem was that in sip_request_call, the
+ jointcapability of the outbound channel was being set to the
+ format passed to sip_request_call. This is nativeformats of the
+ inbound channel. Because of this, when
+ ast_channel_make_compatible was called by app_dial, both channels
+ already had compatibly read and write formats. Thus, no
+ translation path was set up at the time. My change is to set the
+ jointcapability of the sip_pvt created during sip_request_call to
+ the intersection of the inbound channel's nativeformats and the
+ configured peer capability that we determined during the earlier
+ call to create_addr. Doing this got the translation paths set up
+ as expected when using transcode_via_sln. The changes presented
+ in channel.c fixed the second issue for me. First and foremost,
+ when Asterisk is started, we'll read codecs.conf to see the value
+ of the genericplc option. If this option is set, and ast_write is
+ called for a frame with no data, then we will attempt to fill in
+ the missing samples for the frame. The implementation uses a
+ channel datastore for maintaining the PLC state and for creating
+ a buffer to store PLC samples in. Even when we receive a frame
+ with data, we'll call plc_rx so that the PLC state will have
+ knowledge of the previous voice frame, which it can use as a
+ basis for when it comes time to actually do a PLC fill-in. So,
+ reviewers, now I ask for your help. First off, there's the one
+ line change in chan_sip that I have put in. Is it right? By my
+ logic it seems correct, but I'm sure someone can tell me why it
+ is not going to work. This is probably the change I'm least
+ concerned about, though. What concerns me much more is the set of
+ changes in channel.c. First off, am I even doing it right? When I
+ run tests, I can clearly see that when PLC is activated, I see a
+ significant increase in RTP traffic where I would expect it to
+ be. However, in my humble opinion, the audio sounds kind of
+ crappy whenever the PLC fill-in is done. It sounds worse to me
+ than when no PLC is used at all. I need someone to review the
+ logic I have used to be sure that I'm not misusing anything. As
+ far as I can see my pointer arithmetic is correct, and my use of
+ AST_FRIENDLY_OFFSET should be correct as well, but I'm sure
+ someone can point out somewhere where I've done something
+ incorrectly. As I was writing this review request up, I decided
+ to give the code a test run under valgrind, and I find that for
+ some reason, calls to plc_rx are causing some invalid reads.
+ Apparently I'm reading past the end of a buffer somehow. I'll
+ have to dig around a bit to see why that is the case. If it's
+ obvious to someone reviewing, speak up! Finally, I have one other
+ proposal that is not reflected in my code review. Since without
+ transcode_via_sln set, one cannot predict or control where a
+ translation path will be up, it seems to me that the current
+ practice of using PLC only when transcoding to SLINEAR is not
+ useful. I recommend that once it has been determined that the
+ method used in this code review is correct and works as expected,
+ then the code in translate.c that invokes PLC should be removed.
+ Review: https://reviewboard.asterisk.org/r/622/
+
+2010-05-19 20:30 +0000 [r264400] David Vossel <dvossel@digium.com>
+
+ * main/udptl.c: fixes infinite loop during udptl.c's
+ decode_open_type When decode_length returns the length there is a
+ check to see if that length is negative, if so the decode loop
+ breaks as this means the limit has been reached. The problem here
+ is that length is an unsigned int, so length can never be
+ negative. This resulted in an infinite loop. (issue #17352)
+
+2010-05-19 20:26 +0000 [r264335-264379] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/udptl.c: Cast an unsigned int to a signed int when comparing
+ it with 0. (AST-377)
+
+ * /, apps/app_speech_utils.c: Merged revisions 264334 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r264334 | mnicholson | 2010-05-19 15:01:38 -0500 (Wed,
+ 19 May 2010) | 5 lines Set quieted flag when receiving a dtmf
+ tone during playback in speechbackground. (closes issue #16966)
+ Reported by: asackheim ........
+
+2010-05-19 19:21 +0000 [r264331] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes crash in check_rtp_timeout During
+ deadlock avoidance the sip dialog pvt is locked and unlocked.
+ When this occurs we have no guarantee the pvt's owner is still
+ valid. We were trying to access the pvt's owner after this
+ without checking to see if it still existed first. (closes issue
+ #17271) Reported by: under Patches: check_rtp_timeout.diff
+ uploaded by under (license 914) Tested by: dvossel
+
+2010-05-19 17:48 +0000 [r264204-264249] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/options.h: Merged revisions 264248 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r264248 | tilghman | 2010-05-19 12:41:29 -0500 (Wed, 19
+ May 2010) | 17 lines Internal timing is now on by default, if
+ you're using DAHDI 2.3 or above. The reason for ensuring DAHDI
+ 2.3 or above is that this version ensures that a timer is always
+ available, whereas in previous versions, it was possible for
+ DAHDI to be loaded, but have no drivers to actually generate
+ timing. If internal_timing was turned on in this circumstance, a
+ complete lack of audio would result. This is the reason why
+ internal_timing was not on by default. However, now that DAHDI
+ ensures the availability of a timer, there is no reason for this
+ setting to be off (and in fact, it solves a great many initial
+ user problems). (closes issue #15932) Reported by: dimas Patches:
+ 20100519__issue15932.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman ........
+
+ * main/dsp.c: Keep track of digit duration, when we're decoding
+ inband to pass DTMF frames. (closes issue #17235) Reported by:
+ frawd Patches: new_dtmf_dsp_len.patch uploaded by frawd (license
+ 610) 20100518__issue17235.diff.txt uploaded by tilghman (license
+ 14) Tested by: frawd
+
+2010-05-19 15:39 +0000 [r264161] Leif Madsen <lmadsen@digium.com>
+
+ * main/cli.c: Fix compilation problem with previous commit. (issue
+ #16009)
+
+2010-05-19 15:29 +0000 [r264160] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/logger.c, configs/logger.conf.sample: Add ability for logger
+ channels to include *all* levels. Now that Asterisk modules can
+ dynamically create and destroy logger levels on demand, it's
+ useful to be able to configure a logger channel (console, file,
+ whatever) to be able to accept log messages from *all* levels,
+ even levels created dynamically. This patch adds support for
+ this, by allowing the '*' level name to be used in logger.conf.
+ Review: https://reviewboard.asterisk.org/r/663/
+
+2010-05-19 15:12 +0000 [r264117] Leif Madsen <lmadsen@digium.com>
+
+ * CHANGES, main/cli.c: Add ability to hangup all channels from the
+ CLI. Added the keyword 'all' to the 'channel hangup request' CLI
+ command so that you can request all channels to be hungup without
+ having to restart Asterisk. (closes issue #16009) Reported by:
+ moy Patches: hangup-all-rev-221688.patch uploaded by moy (license
+ 222) Tested by: moy, russell
+
+2010-05-19 14:38 +0000 [r264114] David Vossel <dvossel@digium.com>
+
+ * res/res_rtp_asterisk.c: fixes crash during dtmf During the
+ processing of Cisco dtmf the dtmf samples were not being
+ calculated correctly. In an attempt to determine what sample rate
+ was being used, a NULL frame was processed which caused a crash.
+ This patch resolves this. (closes issue #17248) Reported by:
+ falves11 Patches: issue_17248.diff uploaded by dvossel (license
+ 671)
+
+2010-05-19 08:09 +0000 [r264031] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * configs/indications.conf.sample: fix incorrectly typed
+ indications for [nz] stutter and dialrecall (closes issue #17359)
+ Reported by: alecdavis Patches: bug17359.diff.txt uploaded by
+ alecdavis (license 585)
+
+2010-05-19 06:41 +0000 [r263905-263950] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/dsp.c: Merged revisions 263949 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r263949 | tilghman | 2010-05-19 01:32:27 -0500 (Wed, 19 May 2010)
+ | 8 lines Because progress is called multiple times, across
+ several frames, we must persist states when detecting multitone
+ sequences. (closes issue #16749) Reported by: dant Patches:
+ dsp.c-bug16749-1.patch uploaded by dant (license 670) Tested by:
+ dant ........
+
+ * configure, configure.ac, build_tools/sha1sum-sh (added),
+ makeopts.in, sounds/Makefile: Add an sha1sum-workalike for
+ platforms which don't have it (like Mac OS X)
+
+2010-05-18 22:48 +0000 [r263904] David Vossel <dvossel@digium.com>
+
+ * main/strings.c: fixes segfault on logging (closes issue #17331)
+ Reported by: under Patches: utils.diff uploaded by under (license
+ 914) segfault_on_logging.diff uploaded by dvossel (license 671)
+ Tested by: under, dvossel
+
+2010-05-18 21:09 +0000 [r263860] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Be sure to heap-allocate the redirecting to
+ tag so as not to cause crashiness.
+
+2010-05-18 20:49 +0000 [r263858] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_timing_kqueue.c: Make happy green color come back
+
+2010-05-18 20:09 +0000 [r263810] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix memory leaks in redirecting structures
+ in chan_sip.c Thanks to Richard for pointing this out.
+
+2010-05-18 19:30 +0000 [r263807-263808] Jeff Peeler <jpeeler@digium.com>
+
+ * CHANGES: put changes with the correct version
+
+ * /, CHANGES, apps/app_directory.c: Merged revisions 263769 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r263769 | jpeeler | 2010-05-18 13:54:58 -0500 (Tue, 18 May 2010)
+ | 10 lines Modify directory name reading to be interrupted with
+ operator or pound escape. In the case of accidentally entering
+ the wrong first three letters for the reading, users could be
+ very frustrated if the name listing is very long. This allows
+ interrupting the reading by pressing 0 or #. 0 will attempt to
+ execute a configured operator (o) extension and # will exit and
+ proceed in the dialplan. ABE-2200 ........
+
+2010-05-17 23:49 +0000 [r263724] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ makeopts.in, sounds/Makefile, autoconf/ast_ext_lib.m4: Cache
+ sound tarfiles in a common directory, such that a clean reinstall
+ does not force a re-download of the tarballs. (closes issue
+ #15370) Reported by: pprindeville Patches:
+ asterisk-trunk-bugid15370.patch uploaded by pprindeville (license
+ 347) Tested by: pprindeville, tilghman, seanbright
+
+2010-05-17 22:08 +0000 [r263640] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/devicestate.c: Merged revisions 263639 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r263639 | mmichelson | 2010-05-17 17:00:28 -0500 (Mon, 17 May
+ 2010) | 10 lines Fix logic error when checking for a devstate
+ provider. When using strsep, if one of the list of specified
+ separators is not found, it is the first parameter to strsep
+ which is now NULL, not the pointer returned by strsep. This issue
+ isn't especially severe in that the worst it is likely to do is
+ waste some cycles when a device with no '/' and no ':' is passed
+ to ast_device_state. ........
+
+2010-05-17 19:31 +0000 [r263589] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: With IMAP backend, messages in INBOX were
+ counted twice for MWI. (closes issue #17135) Reported by:
+ edhorton Patches: 20100513__issue17135.diff.txt uploaded by
+ tilghman (license 14) 17135_2.diff uploaded by ebroad (license
+ 878) Tested by: edhorton, ebroad
+
+2010-05-17 15:36 +0000 [r263541] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, channels/chan_local.c, main/rtp_engine.c,
+ channels/chan_sip.c, include/asterisk/channel.h,
+ configs/misdn.conf.sample, apps/app_queue.c,
+ funcs/func_redirecting.c, channels/misdn_config.c,
+ main/channel.c, main/dial.c, channels/chan_dahdi.c,
+ channels/misdn/isdn_lib.h, channels/chan_misdn.c,
+ channels/misdn/chan_misdn_config.h, main/features.c,
+ funcs/func_connectedline.c, include/asterisk/frame.h,
+ funcs/func_callerid.c, channels/sip/include/sip.h: Enhancements
+ to connected line and redirecting work. From reviewboard: Digium
+ has a commercial customer who has made extensive use of the
+ connected party and redirecting information present in later
+ versions of Asterisk Business Edition and which is to be in the
+ upcoming 1.8 release. Through their use of the feature, new
+ problems and solutions have come about. This patch adds several
+ enhancements to maximize usage of the connected party and
+ redirecting information functionality. First, Asterisk trunk
+ already had connected line interception macros. These macros
+ allow you to manipulate connected line information before it was
+ sent out to its target. This patch adds the same feature except
+ for redirecting information instead. Second, the ast_callerid and
+ ast_party_id structures have been enhanced to provide a "tag."
+ This tag can be set with func_callerid, func_connectedline,
+ func_redirecting, and in the case of DAHDI, mISDN, and SIP
+ channels, can be set in a configuration file. The idea behind the
+ callerid tag is that it can be set to whatever value the
+ administrator likes. Later, when running connected line and
+ redirecting macros, the admin can read the tag off the
+ appropriate structure to determine what action to take. You can
+ think of this sort of like a channel variable, except that
+ instead of having the variable associated with a channel, the
+ variable is associated with a specific identity within Asterisk.
+ Third, app_dial has two new options, s and u. The s option lets a
+ dialplan writer force a specific caller ID tag to be placed on
+ the outgoing channel. The u option allows the dialplan writer to
+ force a specific calling presentation value on the outgoing
+ channel. Fourth, there is a new control frame subclass called
+ AST_CONTROL_READ_ACTION added. This was added to correct a very
+ specific situation. In the case of SIP semi-attended (blond)
+ transfers, the party being transferred would not have the
+ opportunity to run a connected line interception macro to
+ possibly alter the transfer target's connected line information.
+ The issue here was that during a blond transfer, the SIP transfer
+ code has no bridged channel on which to queue the connected line
+ update. The way this was corrected was to add this new control
+ frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on
+ the channel on which the connected line interception macro should
+ be run. When ast_read is called to read the frame, ast_read
+ responds by calling a callback function associated with the
+ specific read action the control frame describes. In this case,
+ the action taken is to run the connected line interception macro
+ on the transferee's channel. Review:
+ https://reviewboard.asterisk.org/r/652/
+
+2010-05-17 15:14 +0000 [r263375-263460] Leif Madsen <lmadsen@digium.com>
+
+ * main/manager.c: Missing newlines added to Set-Cookie line in
+ manager.c Sean Bright pointed out that we lost a set of newline
+ characters in commit 190349 on a line I had recently changed. Yay
+ for code review on commits. (issue #17231, #10961)
+
+ * main/manager.c, /: Recorded merge of revisions 263456 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r263456 | lmadsen | 2010-05-17 09:35:18 -0500 (Mon, 17 May 2010)
+ | 11 lines Manager cookies are not compatible with RFC2109. The
+ Version field in the cookies we're setting contain quotes around
+ the version number which is not compatible with RFC2109 and
+ breaks some implementations. (closes issue #17231) Reported by:
+ ecarruda Patches: manager_rfc2109-trunk-v1.patch uploaded by
+ ecarruda (license 559) manager_rfc2109-1.6.2-v1.patch uploaded by
+ ecarruda (license 559) Tested by: ecarruda, russell ........
+
+ * /, sounds/Makefile: Merged revisions 263374 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r263374 | lmadsen | 2010-05-17 09:04:57 -0500 (Mon, 17 May 2010)
+ | 8 lines Update link to new version of core sounds. The latest
+ version of the core sounds files 1.4.19 now includes the missing
+ queue-minute sound file which is called by app_queue but which
+ has been missing. (closes issue #17123) Reported by: n8ideas
+ ........
+
+2010-05-17 13:05 +0000 [r263294] David Vossel <dvossel@digium.com>
+
+ * CHANGES: Update CHANGES to reflect DAHDI buffer dialstring option
+ backport to 1.6.2
+
+2010-05-16 16:31 +0000 [r263250] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * contrib/scripts/live_ast: live_ast: add commands 'rsync' and
+ 'gen-live-asterisk' This adds the following two commands to
+ live_ast: * rsync [user]@host directory Copy over all generated
+ files to <directory> at remote host. Would allow running live_ast
+ there. Hence allows separating a build machine from a test
+ machine. * gen-live-asteris: regenerate live/asterisk . Useful if
+ copying over files to a different directory.
+
+2010-05-16 11:14 +0000 [r263208] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/astobj2.c: Improve some very confusing structure names in
+ astobj2.c As pointed out by 'akshayb' on #asterisk-dev, the code
+ here called a list of bucket entries a 'bucket', and the entries
+ within the bucket were called 'bucket_list'. This made the code
+ very hard to understand without reading all of it... so I've
+ renamed 'bucket_list' to 'bucket_entry' to clarify the purpose of
+ the structure.
+
+2010-05-14 18:53 +0000 [r263151] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: fix iax_frame double free Very unfortunate
+ things happen if we add an iax_frame to the frame queue and let
+ go of the lock before scheduling the frame's transmit... There is
+ a race condition that exists where the frame can be removed from
+ the frame_queue and freed before the transmit is scheduled if we
+ do not hold on to that lock. This results in a freed frame being
+ scheduled for transmit later.
+
+2010-05-13 22:01 +0000 [r263069] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Fix inverted logic in cli command: ss7 set
+ debug on/off
+
+2010-05-13 20:25 +0000 [r263028] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * configure, configure.ac: Remove "untested" feature PRI_VERSION
+ Nobody seems to actually test PRI_VERSION. It is only useful for
+ failing PRI support in chan_dahdi.
+
+2010-05-13 17:49 +0000 [r262940-262987] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_timing_kqueue.c: For FreeBSD
+
+ * res/res_timing_kqueue.c: Hmmm, probably should have read the
+ manpage more thoroughly.
+
+2010-05-13 15:36 +0000 [r262895-262897] Russell Bryant <russell@digium.com>
+
+ * channels/chan_console.c: Fix an off by one error that causes a
+ crash. Thanks to Raymond Burke for pointing it out.
+
+ * main/stdtime/localtime.c: Fix build on linux.
+
+ * pbx/pbx_spool.c: Fix build on linux.
+
+2010-05-13 05:37 +0000 [r262852] Tilghman Lesher <tlesher@digium.com>
+
+ * Makefile, pbx/pbx_spool.c, tests/test_time.c,
+ build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ main/stdtime/localtime.c, res/res_timing_kqueue.c (added): Add
+ kqueue(2) implementation to Asterisk in various places. This will
+ save a considerable amount of CPU on the BSDs, including Mac OS
+ X, as it eliminates several places in the code that we previously
+ used a busy loop. Additionally, this adds a res_timing interface,
+ using kqueue timers. Review:
+ https://reviewboard.asterisk.org/r/543/
+
+2010-05-12 19:59 +0000 [r262800] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * main/loader.c, main/cli.c: Notify CLI when modules is loaded /
+ unloaded (closes issue #17308) Reported by: pabelanger Patches:
+ cli.modules.patch uploaded by pabelanger (license 224) Tested by:
+ pabelanger, russell
+
+2010-05-12 19:53 +0000 [r262796-262798] Leif Madsen <lmadsen@digium.com>
+
+ * res/ael/pval.c: Revert previous WARNING message removal.
+ Marquis42 suggested a better method of doing what I wanted
+ because I ended up removing the WARNING message for all instances
+ when really I just wanted to remove it for the 'return' keyword,
+ not everything. (issue #17145)
+
+ * res/ael/pval.c: Remove unnecessary WARNING message in ael/pval.c
+ (closes issue #17145) Reported by: okrief
+
+2010-05-12 18:01 +0000 [r262744] David Vossel <dvossel@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 262662 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010)
+ | 11 lines fixes app_meetme dsp error We attempted to detect
+ silence after translating a frame from signed linear. This caused
+ a flooding of errors. To resolve this the code to detect silence
+ was moved before the translation. (closes issue #17133) Reported
+ by: jsdyer ........
+
+2010-05-12 17:57 +0000 [r262661-262743] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Don't crash when destroying chan_dahdi
+ pseudo channels. Must do a deep copy of the cc_params in
+ duplicate_pseudo(). Otherwise, when the duplicate pseudo channel
+ is destroyed, it frees the original pseudo channel cc_params. The
+ original pseudo channel is then left with a dangling pointer for
+ when the next duplicated pseudo channel is created.
+
+ * channels/chan_misdn.c: Merged revisions 262657,262660 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r262660 | rmudgett | 2010-05-12 11:46:47 -0500 (Wed,
+ 12 May 2010) | 4 lines Forgot some conditionals around the
+ callrerouting facility help text. JIRA ABE-2223 ..........
+ r262657 | rmudgett | 2010-05-12 11:26:49 -0500 (Wed, 12 May 2010)
+ | 22 lines Add mISDN Call rerouting facility for point-to-point
+ ISDN lines (exchange line) In the case of ISDN
+ point-to-multipoint (multidevice) you can use the mISDN "facility
+ calldeflect" application for call diversions from external (PSTN)
+ to external (PSTN). In that case this is the only way to get rid
+ of the two call legs to the PBX and let the calling number at the
+ C party become the number of the A party. In the case of ISDN
+ point-to-point (exchange line) the call deflection facility may
+ not be used. Instead a call rerouting facility has to be used.
+ This patch for chan_misdn.c is an extension to realize this
+ service (facility rerouting application). It can accept either
+ spelling: "callrerouting" or "callrerouteing". The patch is
+ tested towards Deutsche Telekom and requires a modified version
+ of mISDN from Digium, Inc. Patches:
+ misdn_rerouteing_corrected.patch (Slightly modified.) JIRA
+ ABE-2223
+
+2010-05-12 16:23 +0000 [r262656] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_privacy.c: Ensure the arguments are initialized. Also
+ miscellaneous CG cleanup. (closes issue #16576) Reported by:
+ uxbod Patches: 20100505__issue16576.diff.txt uploaded by tilghman
+ (license 14) Tested by: uxbod
+
+2010-05-12 01:00 +0000 [r262613] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * channels/chan_sip.c, include/asterisk/cli.h: Convert to
+ AST_CLI_YESNO and AST_CLI_ONOFF Clean up chan_sip.c to use new
+ AST_CLI functions (closes issue #17287) Reported by: pabelanger
+ Patches: issue17287.patch uploaded by pabelanger (license 224)
+ Tested by: russell
+
+2010-05-11 23:18 +0000 [r262569] Richard Mudgett <rmudgett@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ channels/sig_pri.c: Dialing an invalid extension causes
+ incomplete hangup sequence. Revision -r1489 of the libpri 1.4
+ branch corrected a deviation from Q.931 Section 5.3.2. However,
+ this resulted in an unexpected behaviour change to the upper
+ layer (Asterisk). This change uses pri_hangup_fix_enable() to
+ follow Q.931 Section 5.3.2 call hangup better if the version of
+ libpri supports it. (issue #17104) Reported by: shawkris Tested
+ by: rmudgett
+
+2010-05-11 21:25 +0000 [r262513] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/causes.h: Move cause 200 to cause 26, as
+ specified in Q.850. Also cleanup the formatting and add a few
+ more that seem like good candidates. (closes issue #16157)
+ Reported by: wimpy
+
+2010-05-11 19:57 +0000 [r262422] Jason Parker <jparker@digium.com>
+
+ * /, res/Makefile: Merged revisions 262421 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r262421 | qwell | 2010-05-11 14:55:42 -0500 (Tue, 11 May 2010) |
+ 11 lines Use a less silly method for modifying a flex-generated
+ file. The sed syntax that was used wasn't actually valid, causing
+ some versions to choke. This is the method that is used in 1.6.x+
+ for similar changes. (closes issue #16696) Reported by: bklang
+ Patches: 16696-sedfix.diff uploaded by qwell (license 4) Tested
+ by: qwell ........
+
+2010-05-11 19:40 +0000 [r262414-262419] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * pbx/pbx_config.c: Improve logging by displaying line number
+ (closes issue #16303) Reported by: dant Patches:
+ issue16303.patch.v2 uploaded by pabelanger (license 224) Tested
+ by: dant, lmadsen, pabelanger
+
+ * channels/chan_sip.c: Improve logging information for
+ misconfigured contexts (closes issue #17238) Reported by:
+ pprindeville Patches: chan_sip-bug17238.patch uploaded by
+ pprindeville (license 347) Tested by: pprindeville
+
+2010-05-11 17:23 +0000 [r262330] Tilghman Lesher <tlesher@digium.com>
+
+ * /, Makefile.rules, apps/app_voicemail.c: Merged revisions 262321
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r262321 | tilghman | 2010-05-11 12:22:07 -0500 (Tue, 11 May 2010)
+ | 2 lines Fix issue #17302 a slightly different way (mad props to
+ Qwell) ........
+
+2010-05-11 16:43 +0000 [r262299] Jason Parker <jparker@digium.com>
+
+ * bootstrap.sh: Allow bootstrap script to work on Solaris. As
+ usual, the way they do things is different, so we need to account
+ for that. automake is versioned ala BSD/Linux, but autoconf is
+ not. We don't actually need to specify a version there, since
+ AC_PREREQ will cover it for us. Things will fail pretty loudly if
+ AC_PREREQ isn't met. (closes issue #16341) Reported by: bklang
+ Patches: opensolaris_bootstrap.sh uploaded by bklang (license
+ 919)
+
+2010-05-10 19:06 +0000 [r262236-262240] David Vossel <dvossel@digium.com>
+
+ * apps/app_directed_pickup.c: fixes PickupChan application (closes
+ issue #16863) Reported by: schern Patches:
+ app_directed_pickup.c.patch uploaded by schern (license 995)
+ for_trunk.diff uploaded by cjacobsen (license 1029) Tested by:
+ Graber, cjacobsen, lathama, rickead2000, dvossel
+
+ * channels/chan_console.c: fixes crash in chan_console There is a
+ race condition between console_hangup() and start_stream(). It is
+ possible for console_hangup() to be called and then the stream
+ thread to begin after the hangup. To avoid this a check in
+ start_stream() to make sure the pvt-owner still exists while the
+ pvt lock is held is made. If the owner is gone that means the
+ channel hung up and start_stream should be aborted.
+
+2010-05-10 16:36 +0000 [r262152] Tilghman Lesher <tlesher@digium.com>
+
+ * /, Makefile.rules: Merged revisions 262151 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r262151 | tilghman | 2010-05-10 11:34:21 -0500 (Mon, 10 May 2010)
+ | 10 lines Allow compilation on Mac OS X 10.4 (Tiger) (closes
+ issue #17297) Reported by: jcovert Patches:
+ 20100506__issue17297.diff.txt uploaded by tilghman (license 14)
+ (closes issue #17302) Reported by: jcovert ........
+
+2010-05-09 02:14 +0000 [r262048-262102] Tilghman Lesher <tlesher@digium.com>
+
+ * autoconf/ast_c_define_check.m4, configure,
+ include/asterisk/autoconfig.h.in, autoconf/ast_ext_lib.m4,
+ autoconf/ast_c_compile_check.m4: Cleanup a bit more by getting
+ rid of useless version defines. Also make library detection use
+ passed CFLAGS. (closes issue #17309) Reported by: stuarth
+
+ * configure, configure.ac: Use CPPFLAGS to pass PTHREAD_CFLAGS for
+ vpb only
+
+2010-05-07 23:54 +0000 [r262005] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * UPGRADE.txt, apps/app_voicemail.c: VoicemailMain and
+ VMauthenticate, allow escape to the 'a' extension when a single
+ '*' is entered Where a site uses VoicemailMain(mailbox) the users
+ have to be at their own extension to clear their voicemail, they
+ have no way of escaping VoicemailMain to allow entry of new
+ boxnumber. This patch, allows a site to include to 'a' priority
+ in the VoicemailMain context, to allow an escape. If the 'a'
+ priority doesn't exist in the context that VoicemailMain was
+ called from then it acts as the old behaviour. Reported by:
+ alecdavis Tested by: alecdavis Patch vm_a_extension.diff2.txt
+ uploaded by alecdavis (license 585) Review:
+ https://reviewboard.asterisk.org/r/489/
+
+2010-05-07 22:09 +0000 [r261913-261964] Tilghman Lesher <tlesher@digium.com>
+
+ * addons/ooh323c/src/ooh323.c: Fix build on Linux
+
+ * funcs/func_odbc.c: Double free crash (closes issue #17245)
+ Reported by: thedavidfactor Patches:
+ 20100426__issue17245.diff.txt uploaded by tilghman (license 14)
+ Tested by: murraytm
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac: Use
+ the detected pthread building flags in every place, instead of
+ hardcoding -lpthread. We nicely detect the right flags on each
+ system for building Asterisk with pthreads, then ignore it for
+ every other build option that requires us to build with pthreads.
+ This caused some items to return a false negative. Also cleanup
+ some minor naming issues that caused "library library" redundancy
+ in the output. (closes issue #17303) Reported by: stuarth
+ Patches: 20100507__issue17303.diff.txt uploaded by tilghman
+ (license 14) Tested by: stuarth
+
+2010-05-07 16:05 +0000 [r261867] Leif Madsen <lmadsen@digium.com>
+
+ * UPGRADE-1.6.txt: Update UPGRADE-1.6.txt stating insecure=very has
+ been removed. (closes issue #17282) Reported by: stuarth Tested
+ by: stuarth
+
+2010-05-07 15:33 +0000 [r261866] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/sig_pri.c: Fix deadlock in sig_pri when hanging up. The
+ pri_dchannel thread currently violates locking order by locking
+ the private and then attempting to queue a frame, which needs to
+ lock the channel. Queueing a frame is unneccesary though and is
+ actually a regression since sig_pri. All the places that
+ currently use ast_softhangup_nolock now will just set the
+ softhangup value directly as before. (closes issue #17216)
+ Reported by: lmsteffan Patches: bug17216.patch uploaded by
+ jpeeler (license 325)
+
+2010-05-06 23:41 +0000 [r261822] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Some code optimizations. * Made more places
+ use pri_queue_control() instead of pri_queue_frame() and a local
+ frame variable. * Made pri_queue_frame() use
+ sig_pri_lock_owner(). pri_queue_frame() no longer releases the
+ libpri access lock unless it is required. * Made the
+ pri_queue_frame() and pri_queue_control() parameter list similar
+ to sig_pri_lock_owner().
+
+2010-05-06 20:11 +0000 [r261736] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 261735 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r261735 | jpeeler | 2010-05-06 15:10:59 -0500 (Thu, 06
+ May 2010) | 8 lines Only allow the operator key to be accepted
+ after leaving a voicemail. Or rather disallow the operator key
+ from being accepted when not offered, such as after finishing a
+ recording from within the mailbox options menu. ABE-2121 SWP-1267
+ ........
+
+2010-05-06 17:06 +0000 [r261609] Jason Parker <jparker@digium.com>
+
+ * /, sounds/Makefile: Merged revisions 261608 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r261608 | qwell | 2010-05-06 11:56:02 -0500 (Thu, 06 May 2010) |
+ 4 lines Use the versioned MOH tarballs, now that we have them.
+ This makes for more reproducibility. Prompted by a discussion in
+ #asterisk-dev ........
+
+2010-05-06 15:39 +0000 [r261560] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/sip/include/sip.h: Permit more lines within a SIP body
+ to be parsed. The example given within the related issue showed
+ 120 lines, which was mostly a result of the body being XML.
+ (closes issue #17179) Reported by: khw
+
+2010-05-06 14:15 +0000 [r261496-261500] Russell Bryant <russell@digium.com>
+
+ * tests/test_heap.c: Add test case for removing random elements
+ from a heap. I modified the original patch for trunk to use the
+ unit test API. (issue #17277) Reported by: cappucinoking Patches:
+ test_heap.diff uploaded by cappucinoking (license 1036) Tested
+ by: cappucinoking, russell
+
+ * main/heap.c: Fix handling of removing nodes from the middle of a
+ heap. This bug surfaced in 1.6.2 and does not affect code in any
+ other released version of Asterisk. It manifested itself as SIP
+ qualify not happening when it should, causing peers to go
+ unreachable. This was debugged down to scheduler entries
+ sometimes not getting executed when they were supposed to, which
+ was in turn caused by an error in the heap code. The problem only
+ sometimes occurs, and it is due to the logic for removing an
+ entry in the heap from an arbitrary location (not just popping
+ off the top). The scheduler performs this operation frequently
+ when entries are removed before they run (when ast_sched_del() is
+ used). In a normal pop off of the top of the heap, a node is
+ taken off the bottom, placed at the top, and then bubbled down
+ until the max heap property is restored (see max_heapify()). This
+ same logic was used for removing an arbitrary node from the
+ middle of the heap. Unfortunately, that logic is full of fail.
+ This patch fixes that by fully restoring the max heap property
+ when a node is thrown into the middle of the heap. Instead of
+ just pushing it down as appropriate, it first pushes it up as
+ high as it will go, and _then_ pushes it down. Lastly, fix a
+ minor problem in ast_heap_verify(), which is only used for
+ debugging. If a parent and child node have the same value, that
+ is not an error. The only error is if a parent's value is less
+ than its children. A huge thanks goes out to cappucinoking for
+ debugging this down to the scheduler, and then producing an
+ ast_heap test case that demonstrated the breakage. That made it
+ very easy for me to focus on the heap logic and produce a fix.
+ Open source projects are awesome. (closes issue #16936) Reported
+ by: ib2 Tested by: cappucinoking, crjw (closes issue #17277)
+ Reported by: cappucinoking Patches: heap-fix.rev2.diff uploaded
+ by russell (license 2) Tested by: cappucinoking, russell
+
+2010-05-06 07:27 +0000 [r261451] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_dahdi.c: When failing to configure, don't destroy
+ 'cfg' twice Fixes a crash when some config section had an
+ incorrect channel config.
+
+2010-05-05 22:22 +0000 [r261405] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Avoid a crash on SS7 channels.
+
+2010-05-05 20:48 +0000 [r261364] Russell Bryant <russell@digium.com>
+
+ * Makefile, configs/asterisk.conf.sample: Restore previous
+ asterisk.conf syntax, where the directories aren't commented out.
+ This fixes some breakage in the test suite, that uses the
+ contents of asterisk.conf to discover the install layout on the
+ system.
+
+2010-05-05 19:13 +0000 [r261316] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes sip native transfer The Refer-To
+ header field containing the Replaces header in the URI was not
+ being decoded properly. This caused invalid parsing between the
+ caller id field and the domain resulting in a failed transfer.
+ (closes issue #17284) Reported by: dvossel
+
+2010-05-05 18:43 +0000 [r261314] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, channels/chan_sip.c: Merged revisions 261274 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r261274 | pabelanger | 2010-05-05 12:42:22 -0400 (Wed, 05 May
+ 2010) | 12 lines Registration fix for SIP realtime. Make sure
+ realtime fields are not empty. (closes issue #17266) Reported by:
+ Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick
+ Lewis (license 657) Tested by: Nick_Lewis, sberney Review:
+ https://reviewboard.asterisk.org/r/643/ ........
+
+2010-05-05 18:28 +0000 [r261313] Mark Michelson <mmichelson@digium.com>
+
+ * channels/sip/dialplan_functions.c: Prevent unnecessary warnings
+ when getting rtpsource or rtpdest. If a recognized media type was
+ present, but the media type was not enabled for the channel, then
+ a warning would be emitted. For instance, attempting to get
+ CHANNEL(rtpsource,video) on a call with no video would cause a
+ warning message to appear. With this change, the warning will
+ only appear if the stream argument is not recognized as being a
+ media type that can be specified.
+
+2010-05-05 15:42 +0000 [r261124-261232] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * apps/app_queue.c: 'queue reset stats' erroneously clears
+ wrapuptime configuration. Resets each member's lastcall to 0 now.
+ (closes issue #17262) Reported by: rain Patches:
+ wrapuptime_reset_fix.diff uploaded by rain (license 327) Tested
+ by: rain
+
+ * main/manager.c, include/asterisk/cli.h, CHANGES,
+ include/asterisk/manager.h: New 'manager show settings' CLI
+ command. See the CHANGES file for more details. (closes issue
+ #16343) Reported by: pabelanger Patches: issue16343.patch.v5
+ uploaded by pabelanger (license 224) Tested by: pabelanger,
+ tilghman, lmadsen Review: https://reviewboard.asterisk.org/r/630/
+
+ * Makefile, configs/asterisk.conf.sample (added): New static
+ asterisk.conf.sample file. This simply moves the functionality
+ from the Makefile (cleaning it up) into an external
+ asterisk.conf.samples file. Also updates formatting (easier to
+ read) and grammar changes to asterisk.conf.samples. (closes issue
+ #17027) Reported by: pabelanger Patches:
+ 0017027.asterisk.conf.v6.patch uploaded by pabelanger (license
+ 224) Tested by: qwell, lmadsen, pabelanger, chappell Review:
+ https://reviewboard.asterisk.org/r/616/
+
+2010-05-04 23:51 +0000 [r261095] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 261093-261094 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r261093 | tilghman | 2010-05-04 18:36:53 -0500 (Tue, 04
+ May 2010) | 7 lines Protect against overflow, when calculating
+ how long to wait for a frame. (closes issue #17128) Reported by:
+ under Patches: d.diff uploaded by under (license 914) ........
+ r261094 | tilghman | 2010-05-04 18:47:08 -0500 (Tue, 04 May 2010)
+ | 2 lines Add a tiny corner case to the previous commit ........
+
+2010-05-04 22:46 +0000 [r261051] Mark Michelson <mmichelson@digium.com>
+
+ * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add new
+ possible value to autopause option to allow members to be
+ autopaused in all queues. See the CHANGES file and
+ queues.conf.sample for more details. (closes issue #17008)
+ Reported by: jlpedrosa Patches: queues.autopause_en_review.diff
+ uploaded by jlpedrosa (license 1002) Review:
+ https://reviewboard.asterisk.org/r/581/
+
+2010-05-04 21:10 +0000 [r261007] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h, channels/sig_pri.c: The inalarm flag is
+ not passed up from the sig_analog and sig_pri submodules. The CLI
+ "dahdi show channel" command was not correctly reporting the
+ InAlarm status. The inalarm flag is now consistently passed
+ between chan_dahdi and submodules.
+
+2010-05-04 18:51 +0000 [r260924] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 260923 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r260923 | jpeeler | 2010-05-04 13:46:46 -0500 (Tue, 04
+ May 2010) | 12 lines Voicemail transfer to operator should occur
+ immediately, not after main menu. There were two scenarios in the
+ advanced options that while using the operator=yes and review=yes
+ options, the transfer occurred only after exiting the main menu
+ (after sending a reply or leaving a message for an extension).
+ Now after the audio is processed for the reply or message the
+ transfer occurs immediately as expected. ABE-2107 ABE-2108
+ ........
+
+2010-05-04 15:49 +0000 [r260802] Jason Parker <jparker@digium.com>
+
+ * /, build_tools/make_build_h: Merged revisions 260801 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r260801 | qwell | 2010-05-04 10:49:27 -0500 (Tue, 04 May
+ 2010) | 1 line Fix fallout from removing from configure script.
+ Pointed out by philipp64 on #asterisk-dev ........
+
+2010-05-03 22:13 +0000 [r260757] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_meetme.c, CHANGES: Add new admin features to meetme:
+ Roll call, eject all, mute all, record in-conf This patch adds
+ the following in-conference admin DTMF features: *81 - Roll call
+ (or simply user count if INTROUSER isn't enabled) *82 - Eject all
+ non-admins *83 - Mute/unmute all non-admins *84 - Start recording
+ the conference on the fly FWIW, this code uses newly recorded
+ prompts. (closes issue #16379) Reported by: rfinnie Patches:
+ meetme-enhancements-232771-v1.patch uploaded by rfinnie (license
+ 940) modified slightly by me
+
+2010-05-03 17:06 +0000 [r260663] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * Makefile, /: Merged revisions 260661-260662 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260661 | pabelanger | 2010-05-03 12:41:30 -0400 (Mon, 03 May
+ 2010) | 10 lines non-root make install PREFIX=/tmp fails. Prepend
+ libdir when executing mkpkgconfig allowing non-root installs to
+ work. (closes issue #17268) Reported by: pabelanger Patches:
+ issue17268.patch uploaded by pabelanger (license 224) Tested by:
+ pabelanger ........ r260662 | pabelanger | 2010-05-03 12:54:41
+ -0400 (Mon, 03 May 2010) | 3 lines Should have removed /usr/lib/
+ part. Thanks Qwell. ........
+
+2010-05-03 14:58 +0000 [r260570] Leif Madsen <lmadsen@digium.com>
+
+ * doc/HOWTO_collect_debug_information.txt: Merged revisions 260569
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260569 | lmadsen | 2010-05-03 09:57:39 -0500 (Mon, 03 May 2010)
+ | 1 line Minor typo pointed out by pabelanger on IRC. ........
+
+2010-05-02 02:52 +0000 [r260521] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/data.c, include/asterisk/data.h: Avoid making AstData depend
+ on libxml2 to compile. We have some functions inside the AstData
+ API to get the tree in XML form, but it is not required at the
+ moment to compile asterisk and we can disable that part of the
+ API if we don't have libxml2 support.
+
+2010-04-30 22:36 +0000 [r260437] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/sig_analog.h: Merged revisions 260434 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010)
+ | 11 lines Ensure channel state is not incorrectly set in the
+ case of a very early answer. The needringing bit was being read
+ in dahdi_read after answering thereby setting the state to
+ ringing from up. This clears needringing upon answering so that
+ is no longer possible. (closes issue #17067) Reported by: tzafrir
+ Patches: needringing.diff uploaded by tzafrir (license 46)
+ ........
+
+2010-04-30 22:24 +0000 [r260435] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
+ Separate the uses of NUM_DCHANS and MAX_CHANNELS into PRI, SS7,
+ and MFCR2 users. Created SIG_PRI_MAX_CHANNELS, SIG_PRI_NUM_DCHANS
+ SIG_SS7_MAX_CHANNELS, SIG_SS7_NUM_DCHANS SIG_MFCR2_MAX_CHANNELS
+ Also fixed the declaration of pollers[] in mfcr2_monitor(). It
+ was dimensioned to the number of bytes in struct
+ dahdi_mfcr2.pvts[] and not to the same dimension of the struct
+ dahdi_mfcr2.pvts[].
+
+2010-04-30 20:11 +0000 [r260344-260346] Mark Michelson <mmichelson@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 260345 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r260345 | mmichelson | 2010-04-30 15:08:15 -0500 (Fri,
+ 30 Apr 2010) | 18 lines Fix potential crash from race condition
+ due to accessing channel data without the channel locked. In
+ res_musiconhold.c, there are several places where a channel's
+ stream's existence is checked prior to calling ast_closestream on
+ it. The issue here is that in several cases, the channel was not
+ locked while checking the stream. The result was that if two
+ threads checked the state of the channel's stream at
+ approximately the same time, then there could be a situation
+ where both threads attempt to call ast_closestream on the
+ channel's stream. The result here is that the refcount for the
+ stream would go below 0, resulting in a crash. I have added
+ proper channel locking to res_musiconhold.c to ensure that we do
+ not try to check chan->stream without the channel locked. A
+ Digium customer has been using this patch for several weeks and
+ has not had any crashes since applying the patch. ABE-2147
+ ........
+
+ * apps/app_queue.c: Fix logic reversal error when queue callers
+ join the queue. When a specific position is specified for the
+ queue, the idea was that the caller cannot be placed ahead of
+ higher-priority callers. Unfortunately, the logic was reversed so
+ that the caller could ONLY be placed ahead of higher priority
+ callers. Discovered while writing a unit test.
+
+2010-04-30 06:19 +0000 [r260280-260292] Tilghman Lesher <tlesher@digium.com>
+
+ * main/strcompat.c: Don't allow file descriptors to go above 64k,
+ when we're closing them in a fork(2). This saves time, when, even
+ though the system allows the process limit to be that high, the
+ practical limit is much lower. Also introduce an additional
+ optimization, in the form of using the CLOEXEC flag to close
+ descriptors at the right time. (closes issue #17223) Reported by:
+ dbackeberg Patches: 20100423__issue17223.diff.txt uploaded by
+ tilghman (license 14) Tested by: dbackeberg
+
+ * configs/extensions.conf.sample: Logic fixups for a sample FREENUM
+ dialplan context. (closes issue #17263) Reported by: pprindeville
+ Patches: freenum-dialplan.patch#3 uploaded by pprindeville
+ (license 347)
+
+2010-04-29 22:44 +0000 [r260231] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+ 260195 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010)
+ | 26 lines DTMF CallerID detection problems. The code handling
+ DTMF CallerID drops digits on long CallerID numbers and may
+ timeout waiting for the first ring with shorter numbers. The DTMF
+ emulation mode was not turned off when processing DTMF CallerID.
+ When the emulation code gets behind in processing the DTMF digits
+ it can skip a digit. For shorter numbers, the timeout may have
+ been too short. I increased it from 2 seconds to 4 seconds. Four
+ seconds is a typical time between rings for many countries.
+ (closes issue #16460) Reported by: sum Patches: issue16460.patch
+ uploaded by rmudgett (license 664) issue16460_v1.6.2.patch
+ uploaded by rmudgett (license 664) Tested by: sum, rmudgett
+ Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA
+ AST-334 JIRA SWP-901 ........
+
+2010-04-29 18:15 +0000 [r260148] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/extensions.conf.sample: Pattern match fail.
+
+2010-04-29 15:33 +0000 [r260050] David Vossel <dvossel@digium.com>
+
+ * /, include/asterisk/audiohook.h, main/audiohook.c: Merged
+ revisions 260049 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260049 | dvossel | 2010-04-29 10:31:02 -0500 (Thu, 29 Apr 2010)
+ | 14 lines Fixes crash in audiohook_write_list The middle_frame
+ in the audiohook_write_list function was being freed if a
+ audiohook manipulator returned a failure. This is incorrect
+ logic. This patch resolves this and adds detailed descriptions of
+ how this function should work and why manipulator failures must
+ be ignored. (closes issue #17052) Reported by: dvossel Tested by:
+ dvossel (closes issue #16196) Reported by: atis Review:
+ https://reviewboard.asterisk.org/r/623/ ........
+
+2010-04-29 00:35 +0000 [r260007] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/extconf.h: Fix comment.
+
+2010-04-28 22:34 +0000 [r259957] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, channels/sip/include/sip.h: Don't override
+ peer context with domain context. (closes issue #17040) Reported
+ by: pprindeville Patches: asterisk-1.6-bugid17040.patch uploaded
+ by pprindeville (license 347) Tested by: pprindeville Review:
+ https://reviewboard.asterisk.org/r/565/
+
+2010-04-28 21:20 +0000 [r259870] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, channels/chan_local.c, /: Merged revisions 259858
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010)
+ | 33 lines resolves deadlocks in chan_local Issue_1. In the
+ local_hangup() 3 locks must be held at the same time... pvt,
+ pvt->chan, and pvt->owner. Proper deadlock avoidance is done when
+ the channel to hangup is the outbound chan_local channel, but
+ when it is not the outbound channel we have an issue... We
+ attempt to do deadlock avoidance only on the tech pvt, when both
+ the tech pvt and the pvt->owner are locked coming into that loop.
+ By never giving up the pvt->owner channel deadlock avoidance is
+ not entirely possible. This patch resolves that by doing deadlock
+ avoidance on both the pvt->owner and the pvt when trying to get
+ the pvt->chan lock. Issue_2. ast_prod() is used in
+ ast_activate_generator() to queue a frame on the channel and make
+ the channel's read function get called. This function is used in
+ ast_activate_generator() while the channel is locked, which
+ mean's the channel will have a lock both from the generator code
+ and the frame_queue code by the time it gets to chan_local.c's
+ local_queue_frame code... local_queue_frame contains some of the
+ same crazy deadlock avoidance that local_hangup requires, and
+ this recursive lock prevents that deadlock avoidance from
+ happening correctly. This patch removes ast_prod() from the
+ channel lock so only one lock is held during the
+ local_queue_frame function. (closes issue #17185) Reported by:
+ schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel
+ (license 671) issue_17185_v2.diff uploaded by dvossel (license
+ 671) Tested by: schmoozecom, GameGamer43 Review:
+ https://reviewboard.asterisk.org/r/631/ ........
+
+2010-04-28 21:08 +0000 [r259853] Leif Madsen <lmadsen@digium.com>
+
+ * /, config.guess: Merged revisions 259852 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259852 | lmadsen | 2010-04-28 16:07:48 -0500 (Wed, 28 Apr 2010)
+ | 6 lines Update config.guess. Updating config.guess because
+ after installing Ubuntu Server 9.10 and running all the update
+ scripts, running ./configure would not continue because it was
+ unable to determine what kind of system I had. After updating
+ config.guess things started working again. ........
+
+2010-04-28 20:32 +0000 [r259760-259848] Jason Parker <jparker@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 259847 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r259847 | qwell | 2010-04-28 15:30:21 -0500 (Wed, 28 Apr
+ 2010) | 1 line Add AC_CONFIG_AUX_DIR to configure script, so
+ systems without install can use install-sh from our source dir.
+ ........
+
+ * /, makeopts.in: Merged revisions 259833 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259833 | qwell | 2010-04-28 15:25:36 -0500 (Wed, 28 Apr 2010) |
+ 1 line Missed this when removing $ID ........
+
+ * Makefile, /, configure, configure.ac: Merged revisions 259748 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259748 | qwell | 2010-04-28 14:17:38 -0500 (Wed, 28 Apr 2010) |
+ 7 lines Remove usage of `id` since it isn't useful and was
+ causing breakge. Solaris `id` doesn't support the -u argument.
+ Instead of figuring out how to fix this to work on Solaris, I
+ decided to check why it was necessary and where else it was used.
+ It was only used in one place, and it hasn't been needed for a
+ very long time (I question whether it was ever needed). ........
+
+2010-04-28 17:18 +0000 [r259672] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 259664 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r259664 | jpeeler | 2010-04-28 12:13:29 -0500 (Wed, 28
+ Apr 2010) | 4 lines Do not play goodbye prompt after timeout of
+ message review. ABE-2124 ........
+
+2010-04-27 22:47 +0000 [r259587-259617] Jason Parker <jparker@digium.com>
+
+ * res/res_agi.c: Fix compile on systems without
+ HAVE_NULLSAFE_PRINTF defined.
+
+ * channels/sip/dialplan_functions.c: Be more explicit about field
+ naming in a test.
+
+2010-04-27 22:18 +0000 [r259538] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 259531 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27
+ Apr 2010) | 11 lines DAHDI "WARNING" message is confusing and
+ vague "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed
+ failed: Success" Changed the warning to "Failed to decode
+ CallerID on channel 'name'". The message before it is likely more
+ specific about why the CallerID decode failed. SWP-501 AST-283
+ ........
+
+2010-04-27 22:11 +0000 [r259533] Mark Michelson <mmichelson@digium.com>
+
+ * main/ccss.c: Shuffle some casts to make builds on bamboo happier.
+
+2010-04-27 21:49 +0000 [r259527] Leif Madsen <lmadsen@digium.com>
+
+ * /, sounds/Makefile: Merged revisions 259526 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259526 | lmadsen | 2010-04-27 16:48:47 -0500 (Tue, 27 Apr 2010)
+ | 15 lines Update sounds files. * Add additional sounds prompts
+ for say_enumeration * Update the English conference sounds
+ prompts so they are better quality and all sound more consistent
+ * Clean up the core-sounds-XX.txt and extra-sounds-XX.txt files
+ to include all present sound files Both core (en, fr, es) and
+ extra (en, fr) sounds files have been updated. (closes issue
+ #16200) Reported by: murf (closes issue #17137) Reported by:
+ lmadsen ........
+
+2010-04-27 21:18 +0000 [r259439-259451] Jason Parker <jparker@digium.com>
+
+ * /: Block 259441 instead of recording it as merged.
+
+ * /: Recorded merge of revisions 259441 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259441 | qwell | 2010-04-27 16:15:46 -0500 (Tue, 27 Apr 2010) |
+ 1 line Add gar to the check for AR for those silly OSes (Solaris)
+ that don't have ar. ........
+
+ * main/editline/configure, main/editline/Makefile.in,
+ main/editline/configure.in: Add gar to the check for AR for those
+ silly OSes (Solaris) that don't have ar. autoconf2.13 couldn't
+ handle AC_PROG_GREP, so I removed it. This is fine, since we
+ don't need to use anything that the configure script doesn't.
+
+2010-04-27 21:10 +0000 [r259438] Leif Madsen <lmadsen@digium.com>
+
+ * include/asterisk/doxygen/mantisworkflow.h: Update the Mantis
+ Workflow document in doxygen. (closes issue #17175) Reported by:
+ lmadsen Patches: Bug_Tracker_Workflow.v2.txt uploaded by
+ pabelanger (license 224) Tested by: pabelanger, lmadsen
+
+2010-04-27 19:52 +0000 [r259357] Mark Michelson <mmichelson@digium.com>
+
+ * main/ccss.c: Change cc_ref and cc_unref from macros to inline
+ functions. The hope is that Solaris won't be as whiny after this
+ change.
+
+2010-04-27 19:31 +0000 [r259353] Jason Parker <jparker@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 259352 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r259352 | qwell | 2010-04-27 14:29:26 -0500 (Tue, 27 Apr
+ 2010) | 5 lines Support the silly OSes that don't have ar and
+ strip. Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when path
+ isn't specified, and AC_PATH_TOOLS doesn't exist, we'll just
+ switch to AC_CHECK_TOOLS. ........
+
+2010-04-27 18:29 +0000 [r259229-259307] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
+ revisions 259270 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010)
+ | 14 lines hidecalleridname parameter in chan_dahdi.conf Issue
+ #7321 implements a new chan_dahdi configuration option. However,
+ a change mentioned in the issue was never implemented. This is
+ the change that will allow the feature to work. I added a note to
+ chan_dahdi.conf.sample about the feature. (closes issue #17143)
+ Reported by: djensen99 Patches: diff.txt uploaded by djensen99
+ (license NA) (One line change) Tested by: djensen99 ........
+
+ * channels/chan_dahdi.c: Re-fix dahdi_request() iflist locking
+ since CCSS merged.
+
+2010-04-27 15:25 +0000 [r259189] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/etc_default_asterisk (added): Add missing file
+ (pointed out by TheDavidFactor on #asterisk-dev) referenced by
+ revision 239231.
+
+2010-04-26 21:45 +0000 [r259023-259105] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 259104 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259104 | mmichelson | 2010-04-26 16:44:43 -0500 (Mon, 26 Apr
+ 2010) | 3 lines Let compilation succeed warning-free when
+ DONT_OPTIMIZE is turned off. ........
+
+ * main/channel.c, /: Merged revisions 259018 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259018 | mmichelson | 2010-04-26 16:03:08 -0500 (Mon, 26 Apr
+ 2010) | 13 lines Prevent Newchannel manager events for dummy
+ channels. No Newchannel manager event will be fired for channels
+ that are allocated to not match a registered technology type.
+ Thus bogus channels allocated solely for variable substitution or
+ CDR operations do not result in a Newchannel event. (closes issue
+ #16957) Reported by: atis Review:
+ https://reviewboard.asterisk.org/r/601 ........
+
+2010-04-26 19:05 +0000 [r258974] David Ruggles <thedavidfactor@gmail.com>
+
+ * contrib/valgrind.supp: Line 24 missed in compatibility fix in
+ revision 233577 added a "fun:" prefix line 24
+
+2010-04-26 15:59 +0000 [r258934] Leif Madsen <lmadsen@digium.com>
+
+ * channels/chan_sip.c: Small error in the T.140 RTP port verbose
+ log. (closes issue #16988) Reported by: frawd Patches:
+ chan_sip_sdp_verbose_fix.diff uploaded by frawd (license 610)
+ Tested by: russell
+
+2010-04-26 14:18 +0000 [r258896] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c, include/asterisk/res_fax.h, res/res_fax_spandsp.c:
+ Update res_fax and res_fax_spandsp to be compatible with Fax For
+ Asterisk 1.2. The fax session initilization code for T.38 faxes
+ has been rewritten. T.38 session initialization was removed from
+ generic_fax_exec, and split into two different code paths for
+ receive and send. Also the 'z' option (to send a T.38 reinvite if
+ we do not receive one) was added to sendfax. In the output of
+ 'fax show sessions', the 'Type' column has been renamed to 'Tech'
+ and replaced with a new 'Tech' column that will report 'G.711' or
+ 'T.38'. Control of ECM defaults has been added to res_fax A 'fax
+ show settings' CLI command has been added. Support of the new
+ AST_T38_REQUEST_PARMS control method request to handle channels
+ that have already received a T.38 reinvite before the FAX
+ application is start has been added. Support for the 'fax show
+ settings' command has been added to res_fax_spandsp and handling
+ of the ECM flag has been slightly altered.
+
+2010-04-25 18:51 +0000 [r258838-258855] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c: additional checking related to issue 17186
+
+ * addons/chan_ooh323.c: Don't pass zero length callerid to ooh323
+ stack Don't pass zero callerid string to ooh323 stack because it
+ can't encode this properly and can't generate setup message.
+ (closes issue #17186) Reported by: vmikhelson Patches:
+ zero_callerid_num.patch uploaded by may213 (license 454) Tested
+ by: may213
+
+2010-04-25 18:12 +0000 [r258776] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_monitor.c: Merged revisions 258775 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010)
+ | 6 lines When StopMonitor is called, ensure that it will not be
+ restarted by a channel event. (closes issue #16590) Reported by:
+ kkm Patches: resmonitor-16590-trunk.239289.diff uploaded by kkm
+ (license 888) ........
+
+2010-04-22 22:19 +0000 [r258685] Jason Parker <jparker@digium.com>
+
+ * utils/extconf.c: Add another random function that does nothing to
+ make the utils/ dir happy.
+
+2010-04-22 22:11 +0000 [r258675] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c: Fix previous commit.
+
+2010-04-22 22:10 +0000 [r258673-258674] Jason Parker <jparker@digium.com>
+
+ * utils/Makefile, utils/extconf.c: Make utils/ stuff *actually*
+ compile this time.
+
+ * utils/Makefile, utils/extconf.c: Let utils/ dir compile when
+ DEBUG_THREADS is not enabled.
+
+2010-04-22 21:57 +0000 [r258671] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/cdr.c, main/channel.c, /, main/features.c: Merged revisions
+ 193391,258670 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193391 | mnicholson | 2009-05-08 16:01:25 -0500 (Fri, 08 May
+ 2009) | 8 lines Set the proper disposition on originated calls.
+ (closes issue #14167) Reported by: jpt Patches:
+ call-file-missing-cdr2.diff uploaded by mnicholson (license 96)
+ Tested by: dlotina, rmartinez, mnicholson ........ r258670 |
+ mnicholson | 2010-04-22 16:49:07 -0500 (Thu, 22 Apr 2010) | 11
+ lines Fix broken CDR behavior. This change allows a CDR record
+ previously marked with disposition ANSWERED to be set as BUSY or
+ NO ANSWER. Additionally this change partially reverts r235635 and
+ does not set the AST_CDR_FLAG_ORIGINATED flag on CDRs generated
+ from ast_call(). To preserve proper CDR behavior, the
+ AST_CDR_FLAG_DIALED flag is now cleared from all brige CDRs in
+ ast_bridge_call(). (closes issue #16797) Reported by:
+ VarnishedOtter Tested by: mnicholson ........ (closes issue
+ #16222) Reported by: telles Tested by: mnicholson
+
+2010-04-22 21:06 +0000 [r258632] Russell Bryant <russell@digium.com>
+
+ * tests/test_event.c, main/event.c: Add ast_event subscription unit
+ test and fix some ast_event API bugs. This patch introduces
+ another test in test_event.c that exercises most of the
+ subscription related ast_event API calls. I made some minor
+ additions to the existing event allocation test to increase API
+ coverage by the test code. Finally, I made a list in a comment of
+ API calls not yet touched by the test module as a to-do list for
+ future test development. During the development of this test
+ code, I discovered a number of bugs in the event API. 1)
+ subscriptions to AST_EVENT_ALL were not handled appropriately in
+ a couple of different places. The API allows a subscription to
+ all event types, but with IE parameters, just as if it was a
+ subscription to a specific event type. However, the parameters
+ were being ignored. This affected ast_event_check_subscriber()
+ and event distribution to subscribers. 2) Some of the logic in
+ ast_event_check_subscriber() for checking subscriptions against
+ query parameters was wrong. Review:
+ https://reviewboard.asterisk.org/r/617/
+
+2010-04-22 20:04 +0000 [r258595] Eliel C. Sardanons <eliels@gmail.com>
+
+ * apps/app_voicemail.c: Pass interactive = 0 and fix a compile
+ error.
+
+2010-04-22 19:08 +0000 [r258557] Jason Parker <jparker@digium.com>
+
+ * main/lock.c (added), include/asterisk/res_odbc.h,
+ include/asterisk/astobj2.h, main/heap.c, include/asterisk/lock.h,
+ main/astobj2.c, res/res_odbc.c, include/asterisk/heap.h: Remove
+ ABI differences that occured when compiling with DEBUG_THREADS.
+ "Bad Things" would happen if Asterisk was compiled with
+ DEBUG_THREADS, but a loaded module was not (or vice versa). This
+ also immensely simplifies the lock code, since there are no
+ longer 2 separate versions of them. Review:
+ https://reviewboard.asterisk.org/r/508/
+
+2010-04-22 18:07 +0000 [r258517] Eliel C. Sardanons <eliels@gmail.com>
+
+ * doc/manager_1_1.txt, main/channel.c, include/asterisk/doxyref.h,
+ include/asterisk/xml.h, main/data.c (added), main/xml.c,
+ include/asterisk/channel.h, include/asterisk/_private.h,
+ include/asterisk/data.h (added), CHANGES, apps/app_queue.c,
+ main/asterisk.c, apps/app_voicemail.c: Asterisk data retrieval
+ API. This module implements an abstraction for retrieving and
+ exporting asterisk data. Developed by: Brett Bryant
+ <brettbryant@gmail.com> Eliel C. Sardanons (LU1ALY)
+ <eliels@gmail.com> For the Google Summer of code 2009 Project.
+ Documentation can be found in doxygen format and inside the
+ header include/asterisk/data.h Review:
+ https://reviewboard.asterisk.org/r/275/
+
+2010-04-22 17:36 +0000 [r258515] Russell Bryant <russell@digium.com>
+
+ * doc/tex/channelvariables.tex: Add MEETMEBOOKID from r256019.
+
+2010-04-21 21:56 +0000 [r258433] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 258432 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r258432 | jpeeler | 2010-04-21 16:45:36 -0500 (Wed, 21
+ Apr 2010) | 8 lines Fix looping forever when no input received in
+ certain voicemail menu scenarios. Specifically, prompting for an
+ extension (when leaving or forwarding a message) or when
+ prompting for a digit (when saving a message or changing
+ folders). ABE-2122 SWP-1268 ........
+
+2010-04-21 19:45 +0000 [r258351-258387] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/asterisk.tex: Missed this when reverting the bad version
+ change in asterisk.tex.
+
+ * doc/tex/asterisk.tex: Fix change in asterisk.tex that got merged
+ in after testing. (issue #17220)
+
+ * Makefile, doc/tex/security-events.tex, configure,
+ include/asterisk/autoconfig.h.in, doc/tex/Makefile, configure.ac,
+ doc/tex/phoneprov.tex, doc/tex, doc/tex/ael.tex,
+ build_tools/prep_tarball, doc/tex/localchannel.tex,
+ doc/tex/enum.tex, makeopts.in, doc/tex/asterisk.tex,
+ doc/tex/cel-doc.tex: Add ability to generate ASCII documentation
+ from the TeX files. These changes add the ability to run 'make
+ asterisk.txt' just like the existing 'make asterisk.pdf' commands
+ to generate a text document from the TeX files we have in the
+ doc/tex/ directory. I've also updated a few of the .tex files
+ because they weren't properly escaping certain characters so they
+ would show up as Unicode characters (like [U+021C]). Made changes
+ to the configure scripts so it would detect the catdvi program
+ which is required to convert the .dvi file generated by latex.
+ I've also added a few lines to the build_tools/prep_tarball
+ script so that the text documentation gets generated and added to
+ future tarballs of Asterisk releases. (closes issue #17220)
+ Reported by: lmadsen Patches: asterisk.txt.patch uploaded by
+ lmadsen (license 10) asterisk.txt.patch-v4 uploaded by pabelanger
+ (license 224) Tested by: lmadsen, pabelanger
+
+2010-04-21 19:07 +0000 [r258345] Mark Michelson <mmichelson@digium.com>
+
+ * funcs/func_callcompletion.c: Add small documentation update to
+ func_callcompletion.c. This directs users to documents which can
+ help explain the concepts and configuration options settable with
+ the function.
+
+2010-04-21 19:02 +0000 [r258344] Leif Madsen <lmadsen@digium.com>
+
+ * UPGRADE.txt, CHANGES, channels/chan_iax2.c: IAXpeers output now
+ matches SIPpeers format for manager (AMI). (closes issue #17100)
+ Reported by: secesh Tested by: pabelanger Review:
+ https://reviewboard.asterisk.org/r/594/
+
+2010-04-21 18:13 +0000 [r258305] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes issue with double "sip:" in header
+ field This is a clear mistake in logic. Future discussions about
+ how to avoid having to handle uri's like this should take place
+ in the future, but this fix needs to go in for now. (closes issue
+ #15847) Reported by: ebroad Patches: doublesip.patch uploaded by
+ ebroad (license 878)
+
+2010-04-21 13:26 +0000 [r258265] Leif Madsen <lmadsen@digium.com>
+
+ * res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
+ res/res_calendar_caldav.c: Fix the \brief description in the
+ res_calendar_*.c files.
+
+2010-04-21 13:24 +0000 [r258190-258256] Julian Lyndon-Smith <julian@dotr.com>
+
+ * doc/manager_1_1.txt: fix whitespace issue
+
+ * doc/manager_1_1.txt, doc/tex/manager.tex: Added NEW ACTIONS entry
+ for new MixMonitorMute AMI command. Added State and Direction
+ variables for new MixMonitorMute AMI command.
+
+ * CHANGES: Added CHANGES entry for new MixMonitorMute AMI command.
+
+ * main/frame.c, include/asterisk/audiohook.h, main/audiohook.c,
+ include/asterisk/frame.h, apps/app_mixmonitor.c,
+ res/res_mutestream.c: Added MixMonitorMute manager command Added
+ a new manager command to mute/unmute MixMonitor audio on a
+ channel. Added a new feature to audiohooks so that you can mute
+ either read / write (or both) types of frames - this allows for
+ MixMonitor to mute either side of the conversation without
+ affecting the conversation itself. (closes issue #16740) Reported
+ by: jmls Review: https://reviewboard.asterisk.org/r/487/
+
+2010-04-20 19:02 +0000 [r258106-258149] Leif Madsen <lmadsen@digium.com>
+
+ * configs/cli_aliases.conf.sample: Add 'soft hangup' alias per
+ Steve Johnson on asterisk-users.
+
+ * configs/extensions.conf.sample: Add example dialplan for dialing
+ ISN numbers (http://www.freenum.org). Minor tweaks and
+ documentation added by me. (closes issue #17058) Reported by:
+ pprindeville Patches: freenum.patch#5 uploaded by pprindeville
+ (license 347) Tested by: lmadsen
+
+ * contrib/scripts/sip-friends.sql: Add missing 'useragent' field to
+ sip-friends.sql file. (closes issue #17171) Reported by: thehar
+ Patches: sip-friends.patch uploaded by thehar (license 831)
+ Tested by: pabelanger, thehar
+
+2010-04-20 17:06 +0000 [r258065] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 258029 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r258029 | jpeeler | 2010-04-20 11:16:33 -0500 (Tue, 20
+ Apr 2010) | 11 lines Play correct prompt when voicemail store
+ failure occurs after attempted forward. If a user's mailbox was
+ full and a message was attempted to be forwarded to said box,
+ warnings on the console would indicate failure. However, the
+ played prompt was that of success (vm-msgsaved). Now storage
+ failure is taken into account and the correct prompt
+ (vm-mailboxfull) is played when appropriate. ABE-2123 SWP-1262
+ ........
+
+2010-04-20 12:38 +0000 [r257988] Leif Madsen <lmadsen@digium.com>
+
+ * formats/format_pcm.c: Update supported file extensions in
+ doxygen. Updated the doxygen \arg line after looking at the file
+ for some other Asterisk documentation and noticing they weren't
+ up to date. Thanks to seanbright for looking at the code for me
+ :)
+
+2010-04-19 21:57 +0000 [r257947-257949] Jason Parker <jparker@digium.com>
+
+ * main/indications.c: Change log message to match severity.
+
+ * main/indications.c: Don't consider a missing indications.conf to
+ be a critical error. There were many changes in revision 176627
+ which would avoid the error that a missing config would have
+ caused. Other than this, there are no other config files
+ (including asterisk.conf, surprisingly) that are required.
+
+2010-04-19 19:23 +0000 [r257883] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Bad merge fix
+
+2010-04-19 18:42 +0000 [r257851] Mark Michelson <mmichelson@digium.com>
+
+ * funcs/func_srv.c: Commit compromise I suggested on review 608.
+ This allows for multiple SRV queries to be done from the dialplan
+ for the same service on a single call while still allowing one to
+ bypass the call to SRVQUERY if they so please. Taking action
+ since no comments had been left for a while. This can easily be
+ reverted if needed. External tests still pass.
+
+2010-04-19 17:57 +0000 [r257810] Terry Wilson <twilson@digium.com>
+
+ * main/features.c: Fix incomplete CDR merge from r195881 Because
+ res/res_features.c was removed and main/cdr.c added, these
+ changes didn't make it to trunk and the 1.6.x branches
+
+2010-04-18 17:25 +0000 [r257768] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/cdr_odbc.conf.sample: Removing unused configuration
+ parameters
+
+2010-04-16 21:22 +0000 [r257713] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * /, apps/app_mixmonitor.c: Merged revisions 257686 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16
+ Apr 2010) | 21 lines Make the mixmonitor thread process audio
+ frames faster Mantis issue 17078 reports MixMonitor recordings
+ have shorter durations than the call duration. This was because
+ the mixmonitor thread was not processing frames from the
+ audiohook fast enough. The mixmonitor thread would slowly fall
+ behind the most recent audio frame and when the channel hangs up,
+ the mixmonitor thread would exit without processing the same
+ number of frames as the channel; leaving the mixmonitor recording
+ shorter than actual call duration. This revision fixes this issue
+ by moving the ast_audiohook_trigger_wait() and the subsequent
+ audiohook.status check into the block where the
+ ast_audiohook_read_frame() function returns NULL. (closes issue
+ #17078) Reported by: geoff2010 Patches: dw-M17078.patch uploaded
+ by dhubbard (license 733) Tested by: dhubbard, geoff2010 Review:
+ https://reviewboard.asterisk.org/r/611/ ........
+
+2010-04-16 19:50 +0000 [r257646] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Make sure to fail a monitor if we receive a
+ negative response for a CC SUBSCRIBE.
+
+2010-04-16 19:25 +0000 [r257642] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * channels/chan_dahdi.c: Enable PRI SERVICE message support in
+ chan_dahdi for the 'national' switchtype Revision 1072 of libpri
+ added SERVICE message support for the 'national' switchtype. The
+ attached patch enables the use of 'pri service' CLI commands on
+ dahdi channels that are configured for the 'national' switchtype.
+ (closes issue #17142) Reported by: dhubbard Patches: dw-ni2.patch
+ uploaded by dhubbard (license 733) Tested by: elguero, dhubbard
+ Review: https://reviewboard.asterisk.org/r/612/
+
+2010-04-15 21:26 +0000 [r257493-257560] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/app.h, /, tests/test_app.c, main/app.c: Merged
+ revisions 257544 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257544 | tilghman | 2010-04-15 16:23:24 -0500 (Thu, 15 Apr 2010)
+ | 6 lines Allow application options with arguments to contain
+ parentheses, through a variety of escaping techniques. Fixes
+ SWP-1194 (ABE-2143). Review:
+ https://reviewboard.asterisk.org/r/604/ ........
+
+ * /, channels/chan_sip.c: Merged revisions 257467 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010)
+ | 13 lines Don't recreate peer, when responding to a repeated
+ deregistration attempt. When a reply to a deregistration is lost
+ in transmit, the client retries the deregistration. Previously,
+ this would cause a realtime/autocreate peer to be loaded back
+ into memory, after it had already been correctly purged. Instead,
+ we just want to resend the reply without loading the peer.
+ (closes issue #16908) Reported by: kkm Patches:
+ 20100412__issue16908.diff.txt uploaded by tilghman (license 14)
+ Tested by: kkm ........
+
+2010-04-15 19:41 +0000 [r257343-257427] Leif Madsen <lmadsen@digium.com>
+
+ * /, doc/backtrace.txt: Merged revisions 257426 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257426 | lmadsen | 2010-04-15 14:40:33 -0500 (Thu, 15 Apr 2010)
+ | 13 lines Update backtrace.txt documentation. Update the
+ backtrace.txt documentation so it conforms to the same layout as
+ other documents we've been working on recently. Additionally, add
+ a bunch of new information about gathering backtraces for crashes
+ and deadlocks, along with ways of verifying your file before
+ uploading it. Create a couple of one line commands for people to
+ generate the files we need. (closes issue #17190) Reported by:
+ lmadsen Patches: backtrace.txt.patch-2 uploaded by lmadsen
+ (license 10) Tested by: lmadsen, pabelanger ........
+
+ * /, doc/backtrace.txt: Merged revisions 257342 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257342 | lmadsen | 2010-04-15 08:41:45 -0500 (Thu, 15 Apr 2010)
+ | 1 line Update address of the bug tracker. ........
+
+2010-04-14 22:57 +0000 [r257262] Tilghman Lesher <tlesher@digium.com>
+
+ * main/features.c, configs/features.conf.sample: Yet another issue
+ where the conversion of the application delimiter to comma caused
+ an issue. Application arguments within the feature map could
+ possibly contain a comma, which conflicts with the syntax of the
+ features.conf configuration file. This patch allows the argument
+ to be wrapped in parentheses or quoted, to allow the application
+ arguments to be interpreted as a single configuration parameter.
+ (closes issue #16646) Reported by: pinga-fogo Patches:
+ 20100414__issue16646.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman Review:
+ https://reviewboard.asterisk.org/r/547/
+
+2010-04-13 19:17 +0000 [r257191] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Also unref the pvt when we delete the
+ provisional keepalive job. (closes issue #16774) Reported by:
+ kowalma Patches: 20100315__issue16774.diff.txt uploaded by
+ tilghman (license 14) Tested by: falves11, jamicque Review:
+ https://reviewboard.asterisk.org/r/591/
+
+2010-04-13 18:10 +0000 [r257146] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/manager.c, /, configs/manager.conf.sample: Merged revisions
+ 257070 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr
+ 2010) | 9 lines Add an option to restore past broken behavor of
+ the Events manager action Before r238915, certain values for the
+ EventMask parameter of the Events action would result in no
+ response being returned. This patch adds an option to restore
+ that broken behavior. Also while fixing this bug I discovered
+ that passing an empty EventMasks parameter would also result in
+ no response being returned, this has been fixed as well while
+ being preserved when the broken behavior is requested. (closes
+ issue #17023) Reported by: nblasgen Review:
+ https://reviewboard.asterisk.org/r/602/ ........
+
+2010-04-13 16:33 +0000 [r257065] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c: Ensure that we can have commas within
+ cdr values. (closes issue #17001) Reported by: snuffy Patches:
+ 20100412__issue17001.diff.txt uploaded by tilghman (license 14)
+ Tested by: snuffy
+
+2010-04-13 16:18 +0000 [r256985-257032] Mark Michelson <mmichelson@digium.com>
+
+ * configs/sip.conf.sample: Update sample dialstrings in
+ sip.conf.sample file.
+
+ * funcs/func_srv.c: Address Russell's comments on func_srv from
+ reviewboard. * Change copyright date * Place channel in
+ autoservice when doing SRV lookup * Get rid of trailing
+ whitespace * Change logic in load_module function
+
+ * main/ccss.c: Fix issue where recall would not happen when it
+ should. Specifically, the situation would happen when multiple
+ callers would request CC for a single generically-monitored
+ device. If the monitored device became available but the caller
+ did not answer the recall, then there was nothing that would poke
+ the CC core to let it know that it should attempt to recall
+ someone else instead. After careful consideration, I came to the
+ conclusion that the only area of Asterisk that needed to be
+ touched was the generic CC monitor. All other types of CC would
+ require something outside of Asterisk to invoke a recall for a
+ separate device. This was accomplished by changing the generic
+ monitor destructor to poke other generic monitor instances if the
+ device is currently available and the specific instance was
+ currently not suspended. In order to not accidentally trigger
+ recalls at bad times, the fit_for_recall flag was also added to
+ the generic_monitor_instance_list struct. This gets set as soon
+ as a monitored device becomes available. It gets cleared if a
+ CCNR request triggers the creation of a new generic monitor
+ instance. By doing this, we don't accidentally try to recall a
+ device when the monitored device was being monitored for CCNR and
+ never actually became available for recall in the first place.
+ This error was discovered by Steve Pitts during in-house testing
+ at Digium.
+
+2010-04-12 17:29 +0000 [r256860-256901] Leif Madsen <lmadsen@digium.com>
+
+ * /, doc/HOWTO_collect_debug_information.txt (added): Merged
+ revisions 256900 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r256900 | lmadsen | 2010-04-12 12:29:26 -0500 (Mon, 12 Apr 2010)
+ | 15 lines Add How-To document on collecting debugging info for
+ issues.asterisk.org Paul Belanger has been helping a lot with bug
+ tracking recently and created this document that we can now point
+ to when additional debugging information is required. This
+ document will help those filing issues to know how to get the
+ information required when filing their issues. This will make
+ things easier on the developers. Initial text and changes by
+ pabelanger. Tweaks and editing by myself. (closes issue #17159)
+ Reported by: pabelanger Patches:
+ HOWTO_collect_debug_information.txt.patch uploaded by lmadsen
+ (license 10) Tested by: tzafrir, pabelanger, lmadsen ........
+
+ * apps/app_voicemail.c: Remove silly debug message that is not
+ useful. (issue #17159)
+
+2010-04-12 14:47 +0000 [r256823] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: gives channel reference before unlocking it
+ and using setvar helper. To guarantee the channel is valid when
+ calling setvar on the MASTER_CHANNEL dialplan function, a channel
+ reference must be taken before unlocking. Thanks to russell for
+ pointing out the error.
+
+2010-04-12 14:39 +0000 [r256821] Leif Madsen <lmadsen@digium.com>
+
+ * main/logger.c: CLI command logger set level auto complete. A
+ simple patch to enable auto tab complete. (closes issue #17152)
+ Reported by: pabelanger Patches: 0017152.patch uploaded by
+ pabelanger (license 224)
+
+2010-04-12 02:19 +0000 [r256745-256783] Russell Bryant <russell@digium.com>
+
+ * tests/test_substitution.c: test_substitution expects func_curl to
+ be present to work.
+
+ * tests/test_pbx.c: Add ASTERISK_FILE_VERSION() macro
+
+2010-04-10 08:33 +0000 [r256704] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * contrib/scripts/safe_asterisk.8, doc/asterisk.8,
+ contrib/scripts/autosupport.8, contrib/scripts/astgenkey.8: fix
+ hyphen vs. minus in man pages In troff '-' is used for a hyphen.
+ A minus is denoted by '\-' . This is normally also used for a
+ dash. This patch converts all '-'-s that are minuses or dashes to
+ '\-'.
+
+2010-04-09 22:20 +0000 [r256646-256661] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, main/ccss.c: Remove status_response
+ callbacks where they are not needed.
+
+ * channels/chan_local.c: Prevent crash when originating a call to a
+ local channel. Call completion code tries to grab the call
+ completion parameters from the requesting channel during
+ local_request. When originating a call to a local channel,
+ however, this channel is NULL. This was causing an issue for me
+ when trying to run a test script.
+
+2010-04-09 19:46 +0000 [r256569-256608] Richard Mudgett <rmudgett@digium.com>
+
+ * doc/CCSS_architecture.pdf (added): Merge CCSS architecture
+ document from CCSS branch.
+
+ * channels/sig_pri.h, configure, include/asterisk/autoconfig.h.in:
+ Remove PRI CCSS BUGBUG message and update configure script.
+
+2010-04-09 16:04 +0000 [r256485-256530] Mark Michelson <mmichelson@digium.com>
+
+ * channels/sip/reqresp_parser.c, channels/sip/include/sip.h,
+ channels/sip/include/reqresp_parser.h: Add routines for parsing
+ SIP URIs consistently. From the original issue report opened by
+ Nick Lewis: Many sip headers in many sip methods contain the ABNF
+ structure name-andor-addr = name-addr / addr-spec Examples
+ include the to-header, from-header, contact-header,
+ replyto-header At the moment chan_sip.c makes various different
+ attempts to parse this name-andor-addr structure for each header
+ type and for each sip method with sometimes limited degrees of
+ success. I recommend that this name-andor-addr structure be
+ parsed by a dedicated function and that it be used irrespective
+ of the specific method or header that contains the
+ name-andor-addr structure Nick has also included unit tests for
+ verifying these routines as well, so...heck yeah. (closes issue
+ #16708) Reported by: Nick_Lewis Patches:
+ reqresp_parser-nameandoraddr2.patch uploaded by Nick Lewis
+ (license 657 Review: https://reviewboard.asterisk.org/r/549
+
+ * channels/chan_sip.c, tests/test_gosub.c, funcs/func_srv.c: Fix
+ some compiler errors that popped up after the CCSS merge.
+
+ * apps/app_dial.c, configs/chan_dahdi.conf.sample,
+ include/asterisk/devicestate.h, include/asterisk/xml.h,
+ channels/chan_local.c, doc/tex/ccss.tex (added), main/ccss.c
+ (added), channels/chan_sip.c, configure.ac, main/xml.c,
+ include/asterisk/channel.h, configs/manager.conf.sample,
+ include/asterisk/channelstate.h (added),
+ include/asterisk/manager.h, CHANGES, channels/sig_pri.c,
+ channels/sig_pri.h, main/channel.c, channels/chan_dahdi.c,
+ main/manager.c, funcs/func_callcompletion.c (added),
+ channels/sig_analog.c, channels/sig_analog.h,
+ configs/ccss.conf.sample (added), include/asterisk/rtp_engine.h,
+ include/asterisk/frame.h, include/asterisk/ccss.h (added),
+ doc/tex/asterisk.tex, main/asterisk.c,
+ channels/sip/include/sip.h: Merge Call completion support into
+ trunk. From Reviewboard: CCSS stands for Call Completion
+ Supplementary Services. An admittedly out-of-date overview of the
+ architecture can be found in the file doc/CCSS_architecture.pdf
+ in the CCSS branch. Off the top of my head, the big differences
+ between what is implemented and what is in the document are as
+ follows: 1. We did not end up modifying the Hangup application at
+ all. 2. The document states that a single call completion monitor
+ may be used across multiple calls to the same device. This proved
+ to not be such a good idea when implementing protocol-specific
+ monitors, and so we ended up using one monitor per-device
+ per-call. 3. There are some configuration options which were
+ conceived after the document was written. These are documented in
+ the ccss.conf.sample that is on this review request. For some
+ basic understanding of terminology used throughout this code, see
+ the ccss.tex document that is on this review. This implements
+ CCBS and CCNR in several flavors. First up is a "generic"
+ implementation, which can work over any channel technology
+ provided that the channel technology can accurately report device
+ state. Call completion is requested using the dialplan
+ application CallCompletionRequest and can be canceled using
+ CallCompletionCancel. Device state subscriptions are used in
+ order to monitor the state of called parties. Next, there is a
+ SIP-specific implementation of call completion. This method uses
+ the methods outlined in draft-ietf-bliss-call-completion-06 to
+ implement call completion using SIP signaling. There are a few
+ things to note here: * The agent/monitor terminology used
+ throughout Asterisk sometimes is the reverse of what is defined
+ in the referenced draft. * Implementation of the draft required
+ support for SIP PUBLISH. I attempted to write this in a
+ generic-enough fashion such that if someone were to want to write
+ PUBLISH support for other event packages, such as dialog-state or
+ presence, most of the effort would be in writing callbacks
+ specific to the event package. * A subportion of supporting
+ PUBLISH reception was that we had to implement a PIDF parser. The
+ PIDF support added is a bit minimal. I first wrote a validation
+ routine to ensure that the PIDF document is formatted properly.
+ The rest of the PIDF reading is done in-line in the
+ call-completion-specific PUBLISH-handling code. In other words,
+ while there is PIDF support here, it is not in any state where it
+ could easily be applied to other event packages as is. Finally,
+ there are a variety of ISDN-related call completion protocols
+ supported. These were written by Richard Mudgett, and as such I
+ can't really say much about their implementation. There are notes
+ in the CHANGES file that indicate the ISDN protocols over which
+ call completion is supported. Review:
+ https://reviewboard.asterisk.org/r/523
+
+ * main/srv.c, channels/chan_sip.c, funcs/func_srv.c (added),
+ CHANGES, include/asterisk/srv.h: func_srv and explicit
+ specification of a remote IP for SIP. From Review Board: There
+ are two interrelated changes here. First, there is the
+ introduction of func_srv. This adds two new read-only dialplan
+ functions, SRVQUERY and SRVRESULT. They work very similarly to
+ the ENUMQUERY and ENUMRESULT functions, except that this allows
+ one to query SRV records instead. In order to facilitate this
+ work, I added a couple of new API calls to srv.h.
+ ast_srv_get_record_count tells the number of records returned by
+ an SRV lookup. This number is calculated at the time of the SRV
+ lookup. ast_srv_get_nth_record allows one to get a numbered SRV
+ record. Second, there is the modification to chan_sip that allows
+ one to specify a hostname or IP address (along with a port) to
+ send an outgoing INVITE to when dialing a SIP peer. This goes
+ hand-in-hand with func_srv. You can query SRV records and then
+ use the host and port from the results to dial via a specific
+ host instead of what is configured in sip.conf. Review:
+ https://reviewboard.asterisk.org/r/608 SWP-1200
+
+2010-04-08 16:35 +0000 [r256428] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, Makefile.rules, build_tools/make_linker_version_script: Ensure
+ that linker version scripts (used for symbol export control)
+ always exist. Using wildcard matching in the Makefile is not
+ adequate to determine whether an export file should exist for a
+ module or not, so instead we'll just create one if the module
+ needs one, or copy the default one if it does not.
+
+2010-04-06 19:28 +0000 [r256370] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/lock.h: Mac OS X does not support comparing a
+ mutex to its initializer. Create a test for this.
+
+2010-04-06 14:42 +0000 [r256319] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes deadlock in chan_sip caused by usage
+ of MASTER_CHANNEL dialplan function (closes issue #16767)
+ Reported by: lmsteffan Patches: deadlock_16767v3.diff uploaded by
+ dvossel (license 671) Review:
+ https://reviewboard.asterisk.org/r/606/
+
+2010-04-06 00:39 +0000 [r256265] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 256225 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05
+ Apr 2010) | 5 lines DAHDI/PRI call to pri_channel_bridge() not
+ protected by PRI lock. SWP-1231 ABE-2163 ........
+
+2010-04-05 15:14 +0000 [r256161] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/localchannel.tex: Fix for localchannel.tex to allow PDFs
+ to be generated again.
+
+2010-04-03 02:12 +0000 [r256103-256104] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, channels/chan_local.c, channels/chan_sip.c,
+ include/asterisk/channel.h, main/cel.c, channels/sig_pri.c,
+ channels/chan_iax2.c, apps/app_queue.c, channels/chan_oss.c,
+ funcs/func_redirecting.c, main/channel.c, main/dial.c,
+ channels/chan_dahdi.c, channels/chan_misdn.c,
+ apps/app_dumpchan.c, res/res_agi.c, channels/chan_h323.c,
+ res/snmp/agent.c, apps/app_amd.c, funcs/func_callerid.c:
+ Consolidate ast_channel.cid.cid_rdnis into
+ ast_channel.redirecting.from.number. SWP-1229 ABE-2161 * Ensure
+ chan_local.c:local_call() will not leak cid.cid_dnid when
+ copying.
+
+ * apps/app_dial.c: Using the Dial application f option when the
+ call is forwarded will likely crash. Fix app_dial.c:do_forward()
+ OPT_FORCECLID setting cid.cid_num with a stack allocated string
+ instead of a heap allocated string.
+
+2010-04-02 23:55 +0000 [r256010-256019] Russell Bryant <russell@digium.com>
+
+ * apps/app_meetme.c: Export MEETMEBOOKID and fix pin-less
+ conferences with realtime conferences (closes issue #16866)
+ Reported by: DEA Patches: rt-meetme-options.txt uploaded by DEA
+ (license 3) Tested by: DEA Review:
+ https://reviewboard.asterisk.org/r/582/
+
+ * channels/chan_local.c, /: Merged revisions 256014 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r256014 | russell | 2010-04-02 18:45:56 -0500 (Fri, 02
+ Apr 2010) | 9 lines Resolve a deadlock that occurs due to a
+ pointless call to ast_bridged_channel() (closes issue #16840)
+ Reported by: bzing2 Patches: patch.txt uploaded by bzing2
+ (license 902) issue_16840.rev1.diff uploaded by russell (license
+ 2) Tested by: bzing2, russell ........
+
+ * main/channel.c, /: Merged revisions 256009 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r256009 | russell | 2010-04-02 18:30:15 -0500 (Fri, 02 Apr 2010)
+ | 2 lines Remove extremely verbose debug message. ........
+
+2010-04-02 20:19 +0000 [r255952] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c: Pass the PID of the Asterisk process, not the
+ PID of the canary. (closes issue #17065) Reported by:
+ globalnetinc Patches: astcanary.patch uploaded by makoto (license
+ 38) Tested by: frawd, globalnetinc
+
+2010-04-02 18:57 +0000 [r255906] Kevin P. Fleming <kpfleming@digium.com>
+
+ * res/res_ael_share.exports.in (added), codecs,
+ res/res_pktccops.exports.in (added), utils,
+ res/res_monitor.exports.in (added), Makefile.moddir_rules,
+ res/res_smdi.exports.in (added), Makefile.rules, cdr,
+ res/res_agi.exports.in (added), formats, main/asterisk.exports
+ (removed), res/res_odbc.exports (removed),
+ res/res_calendar.exports (removed), apps/app_voicemail.exports
+ (removed), bridges, res/res_odbc.exports.in (added),
+ main/asterisk.exports.in (added), apps/app_voicemail.exports.in
+ (added), res/res_calendar.exports.in (added),
+ res/res_features.exports (removed), res/res_fax.exports.in
+ (added), pbx, res/res_adsi.exports.in (added),
+ res/res_jabber.exports (removed), res/res_pktccops.exports
+ (removed), channels, res/res_jabber.exports.in (added),
+ main/Makefile, res/res_smdi.exports (removed), tests, apps, cel,
+ res/res_agi.exports (removed), addons, res/res_speech.exports
+ (removed), Makefile, funcs, res/res_speech.exports.in (added),
+ res/res_fax.exports (removed), main, res/res_adsi.exports
+ (removed), res/res_features.exports.in (added),
+ res/res_ael_share.exports (removed),
+ build_tools/make_linker_version_script (added), res,
+ res/res_monitor.exports (removed): Allow symbol export filtering
+ to work properly on platforms that have symbol prefixes. Some
+ platforms prefix externally-visible symbols in object files
+ generated from C sources (most commonly, '_' is the prefix). On
+ these platforms, the existing symbol export filtering process
+ ends up suppressing all the symbols that are supposed to be left
+ visible. This patch allows the prefix string to be supplied to
+ the top-level Makefile in the LINKER_SYMBOL_PREFIX variable, and
+ then generates the linker scripts as required to include the
+ prefix supplied.
+
+2010-04-02 06:45 +0000 [r255850-255851] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: Ignore Redial softkey when no previous
+ dialed number is known (closes issue #17126) Reported by: wedhorn
+ Patches: skinny79xx_redial1.diff uploaded by wedhorn (license 30)
+
+ * channels/chan_skinny.c: Cleanup transmit_* functions Bulk lot of
+ generally trivial changes for cleaning up the transmit stuff.
+ Line state request has been modified for line only responses.
+ (closes issue #16994) Reported by: wedhorn Patches:
+ skinny-clean07.diff uploaded by wedhorn (license 30) Tested by:
+ wedhorn
+
+2010-04-01 18:16 +0000 [r255796] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/lock.h: Fix DEBUG_THREADS build on Darwin.
+ (closes issue #16828) Reported by: oej Patches:
+ 20100331__issue16828.diff.txt uploaded by tilghman (license 14)
+
+2010-04-01 16:09 +0000 [r255751] Matthew Nicholson <mnicholson@digium.com>
+
+ * configs/sip.conf.sample: Removed documentation of the non
+ existent 'both' option to 'faxdetect' in sip.conf
+
+2010-03-31 22:35 +0000 [r255701] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix improper comaparison of anonymous URI
+ when getting P-Asserted-Identity. There was a bug where we split
+ the URI on the @ sign and then attempted to compare to
+ "anonymous@anonymous.invalid" afterwards. This comparison could
+ never evaluate true. So now we keep a copy of the URI prior to
+ the split so that the comparison is valid.
+
+2010-03-31 19:13 +0000 [r255592] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: Recorded merge of revisions 255591 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r255591 | tilghman | 2010-03-31 14:09:46 -0500 (Wed, 31 Mar 2010)
+ | 15 lines Ensure line terminators in email are consistent. Fixes
+ an issue with certain Mail Transport Agents, where attachments
+ are not interpreted correctly. (closes issue #16557) Reported by:
+ jcovert Patches: 20100308__issue16557__1.4.diff.txt uploaded by
+ tilghman (license 14) 20100308__issue16557__1.6.0.diff.txt
+ uploaded by tilghman (license 14)
+ 20100308__issue16557__trunk.diff.txt uploaded by tilghman
+ (license 14) Tested by: ebroad, zktech Reviewboard:
+ https://reviewboard.asterisk.org/r/544/ ........
+
+2010-03-31 17:48 +0000 [r255504] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_dial.c, /, configs/sip.conf.sample: Add documentation
+ clarifying when 't' and 'T' can be used. (closes issue #17021)
+ Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad
+
+2010-03-30 20:56 +0000 [r255323-255410] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_h323.c: Merged revisions 255409 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r255409 | russell | 2010-03-30 15:56:00 -0500 (Tue, 30
+ Mar 2010) | 2 lines Don't kill Asterisk if the H323 listener does
+ not start. ........
+
+ * /, pbx/pbx_dundi.c: Merged revisions 255322 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r255322 | russell | 2010-03-30 11:06:06 -0500 (Tue, 30 Mar 2010)
+ | 2 lines Don't make Asterisk not start if pbx_dundi fails to
+ initialize. ........
+
+2010-03-29 14:07 +0000 [r255281] Jared Smith <jaredsmith@jaredsmith.net>
+
+ * apps/app_confbridge.c, CHANGES: This patch adds custom device
+ state handling for ConfBridge conferences, matching the devstate
+ handling of the MeetMe conferences. Review:
+ https://reviewboard.asterisk.org/r/572/ Closes issue #16972
+
+2010-03-29 05:10 +0000 [r255240] Russell Bryant <russell@digium.com>
+
+ * main/event.c: Remove a debugging log entry.
+
+2010-03-27 23:51 +0000 [r255199] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c,
+ addons/chan_ooh323.c, addons/ooh323c/src/ooh323.h,
+ addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c:
+ corrections in gk interface, small fixes in call clearing.
+
+2010-03-27 14:44 +0000 [r255158] Sean Bright <sean@malleable.com>
+
+ * apps/app_voicemail.c: We need to inclde sys/wait.h on OpenBSD to
+ get WEXITSTATUS.
+
+2010-03-27 06:09 +0000 [r255117] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_spool.c: inotify support for pbx_spool This should give a
+ good speed boost, in that one particular thread isn't waking up
+ once a second to read directory contents. Reviewboard:
+ https://reviewboard.asterisk.org/r/137/
+
+2010-03-26 19:27 +0000 [r255021-255066] Leif Madsen <lmadsen@digium.com>
+
+ * configs/sip.conf.sample: Replace some documentation from 1.6.x
+ back into trunk. This documentation associated wth tlsbindaddr is
+ still useful so lets synchronize it between trunk and 1.6.x
+ branches. (issue #17054)
+
+ * configs/sip.conf.sample: Update confusing documentation for
+ tlsbindaddr. Update some confusing documentation for the
+ tlsbindaddr option in sip.conf.sample. Point at a link instead
+ which has better documentation. (closes issue #17054) Reported
+ by: klaus3000
+
+2010-03-26 16:27 +0000 [r254976] Sean Bright <sean@malleable.com>
+
+ * contrib/scripts/live_ast: Work around a bug in dash on Ubuntu by
+ checking the number of arguments before shift'ing. Reported and
+ tested by pabelanger.
+
+2010-03-25 23:38 +0000 [r254931] Kevin P. Fleming <kpfleming@digium.com>
+
+ * addons/chan_ooh323.h, addons/ooh323c/src/ooasn1.h,
+ addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooStackCmds.c,
+ addons/ooh323c/src/errmgmt.c, addons/ooh323c/src/ooTimer.c,
+ addons/ooh323c/src/dlist.c, addons/ooh323c/src/eventHandler.c,
+ addons/ooh323c/src/ooCapability.c, addons/ooh323cDriver.c,
+ addons/mp3/interface.c, addons/ooh323cDriver.h,
+ addons/ooh323c/src/rtctype.c, addons/ooh323c/src/ooCalls.c,
+ addons/ooh323c/src/encode.c, addons/ooh323c/src/ooUtils.c,
+ addons/ooh323c/src/ooGkClient.c, addons/ooh323c/src/ooh323ep.c,
+ addons/ooh323c/src/ooports.c, addons/mp3/decode_ntom.c,
+ addons/ooh323c/src/memheap.c, addons/ooh323c/src/ooh323.c,
+ addons/ooh323c/src/ooh245.c, addons/mp3/common.c,
+ addons/ooh323c/src/decode.c, addons/ooh323c/src/context.c,
+ addons/ooh323c/src/perutil.c, addons/mp3/layer3.c,
+ addons/ooh323c/src/oochannels.c,
+ addons/ooh323c/src/ooCmdChannel.c,
+ addons/ooh323c/src/printHandler.c, addons/ooh323c/src/ooq931.c,
+ addons/ooh323c/src/ootrace.c: Use "local" instead of "system"
+ header file inclusion. Now that these files are in the tree, they
+ should prefer the tree's local copy of all Asterisk headers over
+ any that may be installed.
+
+2010-03-25 21:39 +0000 [r254884] Russell Bryant <russell@digium.com>
+
+ * addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/ooSocket.h: Fix
+ a number of other build problems on Mac OS X.
+
+2010-03-25 20:41 +0000 [r254802] Jason Parker <jparker@digium.com>
+
+ * utils/Makefile, /: Merged revisions 254800 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r254800 | qwell | 2010-03-25 15:41:15 -0500 (Thu, 25 Mar 2010) |
+ 1 line Don't remove local copies of utils in uninstall. ........
+
+2010-03-25 20:41 +0000 [r254718-254801] Russell Bryant <russell@digium.com>
+
+ * addons/chan_ooh323.h: Resolve compiler warning on FreeBSD.
+
+ * addons/ooh323c/src/ooh323.c, addons/Makefile,
+ addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ootrace.c: Fix
+ chan_ooh323 so it works on Mac OS X, as well.
+
+ * channels/chan_usbradio.c: chan_usbradio depends on alsa.
+
+2010-03-25 18:38 +0000 [r254636-254638] Kevin P. Fleming <kpfleming@digium.com>
+
+ * .cleancount: Bump cleancount due to ast_channel change.
+
+ * include/asterisk/channel.h: Remove no-longer-used (and unsafe)
+ field in ast_channel for linked lists. The ast_channel structure
+ had a field used for linking a channel into a linked list, but
+ now that ast_channel structures are ao2 objects, this is no
+ longer needed, and could be harmful as ao2 objects really
+ shouldn't ever be placed into linked lists (since those lists
+ don't assist with reference count management on the objects).
+
+ * addons/Makefile: Get chan_ooh323 building again after recent
+ build system changes.
+
+2010-03-25 17:52 +0000 [r254454-254557] Mark Michelson <mmichelson@digium.com>
+
+ * tests/test_acl.c (added): Add unit test for testing ACL
+ functionality. There are two unit tests contained here. 1.
+ "Invalid ACL" This attempts to read a bunch of badly formatted
+ ACL entries and add them to a host access rule. The goal of this
+ test is to be sure that all invalid entries are rejected as they
+ should be. 2. "ACL" This sets up four ACLs. One is a permit all,
+ one is a deny all, and the other two have specific rules about
+ which subnets are allowed and which are not. Then a set of test
+ addresses is used to determine whether we would allow those
+ addresses to access us when each ACL is applied. This test, by
+ the way, was what resulted in AST-2010-003's creation. Review:
+ https://reviewboard.asterisk.org/r/532
+
+ * include/asterisk/acl.h, /: Merged revisions 254552 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r254552 | mmichelson | 2010-03-25 12:33:35 -0500 (Thu,
+ 25 Mar 2010) | 5 lines Add doxygen for acl.h Review:
+ https://reviewboard.asterisk.org/r/528 ........
+
+ * channels/sip/dialplan_functions.c: Add new rtpsource options to
+ the CHANNEL function. This adds rtpsource options analogous to
+ the rtpdest functions that already exist. In addition, this fixes
+ potential crashes which could result due to trying to read values
+ from nonexistent RTP streams.
+
+ * res/res_rtp_asterisk.c, /: Recorded merge of revisions 254452 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r254452 | mmichelson | 2010-03-25 10:59:56 -0500 (Thu, 25 Mar
+ 2010) | 44 lines Several fixes regarding RFC2833 DTMF detection.
+ Here is a copy and paste of the details from my request on
+ reviewboard that dealt with these changes: Fix 1. The first
+ change in place is to fix Mantis issue 15811, which deals with a
+ situation where Asterisk will incorrectly interpret out of order
+ RFC2833 frames as duplicate DTMF digits. For instance, we would
+ receive a sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3:
+ DTMF 1 seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1
+ seqno 7: DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch
+ when we received the frame with seqno 5, we would interpret this
+ as a new DTMF 1. With this patch, we will check the seqno of the
+ incoming digit and not process the frame if the seqno is lower
+ than the last recorded seqno. Note that we do not record the
+ seqno of the dropped DTMF frame for future processing. While the
+ above situation is what was designed to be fixed, the patch is
+ written in such a way that the following would also be fixed too:
+ seqno 9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end)
+ seqno 13: DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno
+ 15: DTMF 2 (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In
+ this second situation, the beginning of the DTMF 2 arrives before
+ the final end frame of the DTMF 1. With the patch, seqno 12 is no
+ processed and thus we properly interpret the DTMF. Fix 2. The
+ second change in place is to fix an issue like the following:
+ seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) *packet
+ lost* seqno 4: DTMF 1 (end) *packet lost* seqno 5: DTMF 1 (end)
+ *packet lost* seqno 6: DTMF 2 When we receive seqno 6, we had
+ code in place that was supposed to properly end the previously
+ unended DTMF 1. The problem was that the code was essentially a
+ no-op. The code would set up an end frame for the DTMF 1 but
+ would immediately overwrite the frame with the begin for DTMF 2.
+ I changed process_dtmf_rfc2833() so that instead of returning a
+ single frame, it is given as an output parameter a list of
+ frames. Each frame that needs to be returned is appended to this
+ list. Fix 3. The final change is a minor one where an
+ AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco
+ DTMF or an RFC 3389 frame and no frame was returned, then we
+ would return &ast_null_frame. The problem is that earlier in the
+ function, we may have generated an AST_CONTROL_SRCCHANGE frame
+ and put it in the list of frames we wish to return. This frame
+ would be lost in such a case. The patch fixes this problem
+ ........
+
+2010-03-25 16:03 +0000 [r254453] Terry Wilson <twilson@digium.com>
+
+ * /, main/file.c: Merged revisions 254451 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r254451 | twilson | 2010-03-25 10:57:29 -0500 (Thu, 25 Mar 2010)
+ | 2 lines Handle new SRCCHANGE control message here too ........
+
+2010-03-25 15:27 +0000 [r254450] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/channel.c, channels/chan_sip.c, res/res_fax.c,
+ configs/sip.conf.sample, include/asterisk/frame.h,
+ channels/sip/include/sip.h: Improve handling of T.38 re-INVITEs
+ that arrive before a T.38-capable application is executing on a
+ channel. This patch addresses an issue found during working with
+ end-users using res_fax. If an incoming call is answered in the
+ dialplan, or jumps to the 'fax' extension due to reception of a
+ CNG tone (with faxdetect enabled), and then the remote endpoint
+ sends a T.38 re-INVITE, it is possible for the channel's T.38
+ state to be 'T38_STATE_NEGOTIATING' when the application starts
+ up. Unfortunately, even if the application wants to use T.38, it
+ can't respond to the peer's negotiation request, because the
+ AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent
+ originally has been lost, and the application needs the content
+ of that frame to be able to formulate a reply. This patch adds a
+ new 'request' type to AST_CONTROL_T38_PARAMETERS,
+ AST_T38_REQUEST_PARMS. If the application sends this request,
+ chan_sip will re-send the original control frame (with
+ AST_T38_REQUEST_NEGOTIATE as the request type), and the
+ application can respond as normal. If this occurs within the five
+ second timeout in chan_sip, the automatic cancellation of the
+ peer reinvite will be stopped, and the application will 'own' the
+ negotiation process from that point onwards. This also improves
+ the code path in chan_sip to allow sip_indicate(), when called
+ for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero
+ response, which should have been in place before since the
+ control frame *can* fail to be processed properly. It also
+ modifies ast_indicate() to return whatever result the channel
+ driver returned for this control frame, rather than converting
+ all non-zero results into '-1'. Finally, the new request type
+ intentionally returns a positive value, so that an application
+ that sends AST_T38_REQUEST_PARMS can know for certain whether the
+ channel driver accepted it and will be replying with a control
+ frame of its own, or whether it was ignored (if the
+ sip_indicate()/ast_indicate() path had properly supported failure
+ responses before, this would not be necessary). This patch also
+ modifies res_fax to take advantage of the new request. In
+ addition, this patch makes sip_t38_abort() actually lock the
+ private structure before doing its work... bad programmer, no
+ donut. This patch also enhances chan_sip's 'faxdetect' support to
+ allow triggering on T.38 re-INVITEs received as well as CNG tone
+ detection. Review: https://reviewboard.asterisk.org/r/556/
+
+2010-03-25 15:21 +0000 [r254446] Leif Madsen <lmadsen@digium.com>
+
+ * res/res_agi.c: handle_speechset has 4 arguments. Update code to
+ reflect that handle_speechset has 4 arguments. (closes issue
+ #17093) Reported by: gpatri Patches: res_agi.patch uploaded by
+ gpatri (license 1014) Tested by: pabelanger, mmichelson
+
+2010-03-25 10:09 +0000 [r254406] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_dahdi.c: remove unneeded explicit channel in dahdi
+ ioctls This patch removes some cases where the channel number for
+ an ioctl was passed as a member in a struct rather then through
+ the file descriptor. The gain setting functions passed around a
+ channel which is always 0, and thus this parameter is simply
+ dropped. Review: https://reviewboard.asterisk.org/r/584/
+
+2010-03-24 21:10 +0000 [r254362] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c: Fix potential invalid reads that could occur in pbx.c
+ Here is a cut and paste of my review request for this change:
+ This past weekend, Russell ran our current suite of unit tests
+ for Asterisk under valgrind. The PBX pattern match test caused
+ valgrind to spew forth two invalid read errors. This patch
+ contains two changes that shut valgrind up and do not cause any
+ new memory leaks. Change 1: In
+ ast_context_remove_extension_callerid2, valgrind reported an
+ invalid read in the for loop close to the function's end.
+ Specifically, one of the the strcmp calls in the loop control was
+ reading invalid memory. This was because the caller of
+ ast_context_remove_extension_callerid2 (__ast_context destroy in
+ this case) passed as a parameter a shallow copy of an ast_exten's
+ exten field. This same ast_exten was what was destroyed inside
+ the for loop, thus any iterations of the for loop beyond the
+ destruction of the ast_exten would result in invalid reads. My
+ fix for this is to make a copy of the ast_exten's exten field and
+ pass the copy to ast_context_remove_extension_callerid2. In
+ addition, I have also acted similarly with the ast_exten's
+ matchcid field. Since in this case a NULL is handled quite
+ differently than an empty string, I needed to be a bit more
+ careful with its handling. Change 2: In __ast_context_destroy, we
+ iterated over a hashtab and called
+ ast_context_remove_extension_callerid2 on each item.
+ Specifically, the hashtab over which we were iterating was an
+ ast_exten's peer_table. Inside of
+ ast_context_remove_extension_callerid2, we could possibly destroy
+ this ast_exten, which also caused the hashtab to be freed.
+ Attempting to call ast_hashtab_end_traversal on the hashtab
+ iterator caused an invalid read to occur when trying to read the
+ iterator->tab->do_locking field since iterator->tab had already
+ been freed. My handling of this problem is a bit less
+ straightforward. With each iteration over the hashtab's contents,
+ we set a variable called "end_traversal" based on the return of
+ ast_context_remove_extension_callerid2. If 0 is ever returned,
+ then we know that the extension was found and destroyed. Because
+ of this, we cannot call ast_hashtab_end_traversal because we will
+ be guaranteeing a read of invalid memory. In such a case, we
+ forego calling ast_hashtab_end_traversal and instead call
+ ast_free on the hashtab iterator. Review:
+ https://reviewboard.asterisk.org/r/585
+
+2010-03-24 18:13 +0000 [r254277-254321] Jeff Peeler <jpeeler@digium.com>
+
+ * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
+ Allow configuration of minsecs and nextaftercmd per mailbox.
+ Previously only configurable globally. A unit test has also been
+ written to provide protection against parse failures for
+ supported mailbox options. (closes issue #16864) Reported by:
+ kobaz Patches: voicemail2.patch uploaded by kobaz (license 834)
+ Review: https://reviewboard.asterisk.org/r/555/
+
+ * /, res/res_monitor.c: Merged revisions 254235 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r254235 | jpeeler | 2010-03-23 19:37:23 -0500 (Tue, 23 Mar 2010)
+ | 72 lines Ensure that monitor recordings are written to the
+ correct location (again) This is an extension to 248860. As such
+ the dialplan test has been extended: ; non absolute path, not
+ combined exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test)
+ exten => 5040, n, dial(sip/5001) ; absolute path, not combined
+ exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2) exten =>
+ 5041, n, dial(sip/5001) ; no path, not combined exten => 5042, 1,
+ monitor(wav,monitor_test3) exten => 5042, n, dial(sip/5001) ;
+ combined: changemonitor from non absolute to no path (leaves
+ tmp/jeff) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m)
+ exten => 5043, n, changemonitor(monitor_test5) exten => 5043, n,
+ dial(sip/5001) ; combined: changemonitor from no path to non
+ absolute path exten => 5044, 1, monitor(wav,monitor_test6,m)
+ exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this
+ wasn't possible before exten => 5044, n, dial(sip/5001) ; non
+ absolute path, combined exten => 5045, 1,
+ monitor(wav,tmp/jeff/monitor_test8,m) exten => 5045, n,
+ dial(sip/5001) ; absolute path, combined exten => 5046, 1,
+ monitor(wav,/tmp/jeff/monitor_test9,m) exten => 5046, n,
+ dial(sip/5001) ; no path, combined exten => 5047, 1,
+ monitor(wav,monitor_test10,m) exten => 5047, n, dial(sip/5001) ;
+ combined: changemonitor from non absolute to absolute (leaves
+ tmp/jeff) exten => 5048, 1,
+ monitor(wav,tmp/jeff/monitor_test11,m) exten => 5048, n,
+ changemonitor(/tmp/jeff/monitor_test12) exten => 5048, n,
+ dial(sip/5001) ; combined: changemonitor from absolute to non
+ absolute (leaves /tmp/jeff) exten => 5049, 1,
+ monitor(wav,/tmp/jeff/monitor_test13,m) exten => 5049, n,
+ changemonitor(tmp/jeff/monitor_test14) exten => 5049, n,
+ dial(sip/5001) ; combined: changemonitor from no path to absolute
+ exten => 5050, 1, monitor(wav,monitor_test15,m) exten => 5050, n,
+ changemonitor(/tmp/jeff/monitor_test16) exten => 5050, n,
+ dial(sip/5001) ; combined: changemonitor from absolute to no path
+ (leaves /tmp/jeff) exten => 5051, 1,
+ monitor(wav,/tmp/jeff/monitor_test17,m) exten => 5051, n,
+ changemonitor(monitor_test18) exten => 5051, n, dial(sip/5001) ;
+ not combined: changemonitor from non absolute to no path (leaves
+ tmp/jeff) exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19)
+ exten => 5052, n, changemonitor(monitor_test20) exten => 5052, n,
+ dial(sip/5001) ; not combined: changemonitor from no path to non
+ absolute exten => 5053, 1, monitor(wav,monitor_test21) exten =>
+ 5053, n, changemonitor(tmp/jeff/monitor_test22) exten => 5053, n,
+ dial(sip/5001) ; not combined: changemonitor from non absolute to
+ absolute (leaves tmp/jeff) exten => 5054, 1,
+ monitor(wav,tmp/jeff/monitor_test23) exten => 5054, n,
+ changemonitor(/tmp/jeff/monitor_test24) exten => 5054, n,
+ dial(sip/5001) ; not combined: changemonitor from absolute to non
+ absolute (leaves /tmp/jeff) exten => 5055, 1,
+ monitor(wav,/tmp/jeff/monitor_test24) exten => 5055, n,
+ changemonitor(tmp/jeff/monitor_test25) exten => 5055, n,
+ dial(sip/5001) ; not combined: changemonitor from no path to
+ absolute exten => 5056, 1, monitor(wav,monitor_test26) exten =>
+ 5056, n, changemonitor(/tmp/jeff/monitor_test27) exten => 5056,
+ n, dial(sip/5001) ; not combined: changemonitor from absolute to
+ no path (leaves /tmp/jeff) exten => 5057, 1,
+ monitor(wav,/tmp/jeff/monitor_test28) exten => 5057, n,
+ changemonitor(monitor_test29) exten => 5057, n, dial(sip/5001)
+ ........
+
+2010-03-23 22:48 +0000 [r254162] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * main/asterisk.c: make 'core show settings' should show all
+ settable directories (closes issue #17086) Reported by: tzafrir
+ Patches: asterisk_extra_settings_dirs.diff uploaded by tzafrir
+ (license 46)
+
+2010-03-23 22:35 +0000 [r254159] Russell Bryant <russell@digium.com>
+
+ * main/test.c: Put test output for a failure in a CDATA section in
+ the XML results.
+
+2010-03-23 21:17 +0000 [r254050] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c: Exit native bridging early for greater timing
+ accuracy with warnings This changes native bridging to break one
+ millisecond early so that the more accurate timeval calculations
+ done in the generic bridge can be performed using the bridge
+ config. Currently the time between exiting native bridging
+ slightly late can sometimes cause a large enough discrepancy for
+ warnings to be missed. For the record, 1.4 does not attempt to
+ native bridge at all when warnings are enabled. (closes issue
+ #15815) Reported by: adomjan Review:
+ https://reviewboard.asterisk.org/r/577/
+
+2010-03-23 20:52 +0000 [r254045] Sean Bright <sean@malleable.com>
+
+ * apps/app_queue.c: Remove unused structure member in app_queue.
+ (closes issue #15494) Reported by: makoto
+
+2010-03-23 19:19 +0000 [r254001] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * tests/Makefile: Change the name of the category 'TEST' to match
+ the name of the subdir
+
+2010-03-23 16:52 +0000 [r253958] Terry Wilson <twilson@digium.com>
+
+ * main/http.c: Don't act like an http write failed when it didn't
+ fwrite returns the number of items written, not the number of
+ bytes
+
+2010-03-23 14:22 +0000 [r253917] Kevin P. Fleming <kpfleming@digium.com>
+
+ * codecs/Makefile, include/asterisk/logger.h, main/Makefile,
+ Makefile.moddir_rules, pbx/Makefile, res/Makefile, CHANGES,
+ channels/Makefile, include/asterisk/options.h, main/cli.c: Change
+ per-file debug and verbose levels to be per-module, the way users
+ expect them to work. 'core set debug' and 'core set verbose' can
+ optionally change the level for a specific filename; however,
+ this is actually for a specific source file name, not the module
+ that source file is included in. With examples like chan_sip,
+ chan_iax2, chan_misdn and others consisting of multiple source
+ files, this will not lead to the behavior that users expect. If
+ they want to set the debug level for chan_sip, they want it set
+ for all of chan_sip, and not to have to also set it for
+ reqresp_parser and other files that comprise the chan_sip module.
+ This patch changes this functionality to be module-name based
+ instead of file-name based. To make this work, some Makefile
+ modifications were required to ensure that the AST_MODULE
+ definition is present in each object file produced for each
+ module as well. Review: https://reviewboard.asterisk.org/r/574/
+
+2010-03-22 20:32 +0000 [r253872] Mark Michelson <mmichelson@digium.com>
+
+ * main/asterisk.c: Initialize channels prior to loading "preload"
+ modules. We can have bad results when a module, upon being
+ loaded, attempts to reference the channels container if the
+ container hasn't yet been initialized. I saw this happen by
+ trying to preload pbx_config.so and having a hint defined which
+ referenced a non-existent SIP peer.
+
+2010-03-22 19:52 +0000 [r253800] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/features.c: Merged revisions 253799 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r253799 | mnicholson | 2010-03-22 14:50:00 -0500 (Mon, 22 Mar
+ 2010) | 4 lines Unconditionally copy the caller's account code to
+ the called party. (related to issue #16331) ........
+
+2010-03-22 19:05 +0000 [r253712-253758] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/dbsep.cgi: Update query should be an UPDATE, not
+ a SELECT.
+
+ * contrib/scripts/dbsep.cgi: Return the list for later
+ manipulation. This fixes an issue with the update procedure.
+ Debugging with mmichelson.
+
+ * contrib/scripts/dbsep.cgi, configs/dbsep.conf.sample: Accomodate
+ equal signs in DSNs and add documentation, based upon
+ mmichelson's feedback.
+
+2010-03-20 16:50 +0000 [r253536-253579] Russell Bryant <russell@digium.com>
+
+ * funcs/func_strings.c: Fix memory corruption found by unit tests.
+ ast_str_reset() was being called on a potentially uninitialized
+ pointer. Valgrind is my hero, once again.
+
+ * cel/cel_pgsql.c, main/tcptls.c, main/manager.c, main/features.c,
+ main/test.c, cdr/cdr_pgsql.c, main/stdtime/localtime.c,
+ main/cel.c: Resolve more compiler warnings on FreeBSD.
+
+ * apps/app_voicemail.c: Include sys/wait.h on FreeBSD to get the
+ WEXITSTATUS() macro.
+
+ * apps/app_dial.c, apps/app_followme.c: Resolve compiler warnings
+ on FreeBSD.
+
+ * pbx/pbx_dundi.c: Resolve a compiler warning on FreeBSD.
+
+ * channels/chan_dahdi.c: Use SHRT_MAX instead of MAXSHORT. These
+ changes fix build issues I had with this module on FreeBSD.
+
+2010-03-19 07:37 +0000 [r253490] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/astobj2.c: prevent segfault if bad magic number is
+ encountered. internal_ao2_ref uses INTERNAL_OBJ which mzy report
+ 'bad magic number', but internal_ao2_ref continues on, causing
+ segfault. Although AO2_MAGIC number is checked by INTERNAL_OBJ
+ before internal_ao2_ref is called, A02_MAGIC is being destroyed
+ (or a wrong pointer) by the time internal_ao2_ref uses
+ INTERNAL_OBJ. internal_ao2_ref now returns -1 if INTERNAL_OBJ
+ encouters a bad magic number. (issue #17037) Reported by:
+ alecdavis Patches: bug17037.diff.txt uploaded by alecdavis
+ (license 585) Tested by: alecdavis
+
+2010-03-18 18:23 +0000 [r253357-253378] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c: Update comment to reflect new timeout value.
+
+ * main/asterisk.c: Increase CLI command output timeout for asterisk
+ -rx to 60 seconds. (closes issue #17049) Reported by: russell
+ Tested by: russell Review:
+ https://reviewboard.asterisk.org/r/573/
+
+2010-03-18 17:52 +0000 [r253345] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_userevent.c: Change usage of pipe to comma in UserEvent
+ docs. Change the example usage of pipe as a separator to comma in
+ the UserEvent documentation. (closes issue #16961) Reported by:
+ jlpedrosa
+
+2010-03-18 15:59 +0000 [r253261] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * res/res_jabber.c: Prevent a crash when a buddy gets offline.
+ (closes issue #16760) Reported by: fiddur Patches: 248394.diff
+ uploaded by fiddur (license 678)i with modifications by me Tested
+ by: fiddur, phsultan
+
+2010-03-18 15:46 +0000 [r253256] Leif Madsen <lmadsen@digium.com>
+
+ * /, doc/tex/localchannel.tex: Update to new Local channel
+ documentation. Add same changes as commit to 1.4, but convert to
+ TeX. (issue #16963) Reported by: kobaz Patches:
+ localchannel-2.txt uploaded by kobaz (license 834)
+
+2010-03-18 15:45 +0000 [r253255] Tilghman Lesher <tlesher@digium.com>
+
+ * main/stdtime/localtime.c: Just in case of a race, send the signal
+ on interrupt.
+
+2010-03-17 19:06 +0000 [r253205] Leif Madsen <lmadsen@digium.com>
+
+ * main/test.c: main/test.c reports erroneous CLI message. (closes
+ issue #17051) Reported by: Nick_Lewis
+
+2010-03-17 14:16 +0000 [r253113] Tilghman Lesher <tlesher@digium.com>
+
+ * tests/test_gosub.c: Switch to using intptr_t, as suggested by
+ Kevin Fleming on the -dev list
+
+2010-03-17 00:40 +0000 [r253028-253032] Leif Madsen <lmadsen@digium.com>
+
+ * main/xmldoc.c: Fix a typo.
+
+ * configs/say.conf.sample: Merged revisions 253018 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r253018 | lmadsen | 2010-03-16 19:26:19 -0500 (Tue, 16
+ Mar 2010) | 6 lines Add french snipset to say.conf. Add the
+ french snipset to say.conf. (Closes issue #15799) ........
+
+2010-03-17 00:23 +0000 [r252976-253004] Tilghman Lesher <tlesher@digium.com>
+
+ * tests/test_gosub.c: Argh.
+
+ * configure, include/asterisk/autoconfig.h.in, tests/test_gosub.c,
+ configure.ac: Fix bamboo compile error by calculating an integer
+ with the same size as a pointer.
+
+ * tests/test_gosub.c (added), apps/app_stack.c: Mask out previous
+ arguments on each nested invocation of Gosub. (closes issue
+ #16758) Reported by: wdoekes Patches:
+ 20100316__issue16758.diff.txt uploaded by tilghman (license 14)
+ Review: https://reviewboard.asterisk.org/r/561/
+
+2010-03-16 19:36 +0000 [r252849] Russell Bryant <russell@digium.com>
+
+ * tests/test_time.c: Re-enable test_time on non-Linux.
+
+2010-03-16 19:36 +0000 [r252848] Sean Bright <sean@malleable.com>
+
+ * res/res_clialiases.c: Include an extra newline after "Aliased CLI
+ command" to get back the prompt. The other issue mentioned in
+ this bug will be more difficult to resolve since we have no idea
+ (right now) of knowing if the command that is aliased has been
+ installed yet. (issue #16978) Reported by: jw-asterisk Tested by:
+ seanbright
+
+2010-03-16 19:34 +0000 [r252846] Tilghman Lesher <tlesher@digium.com>
+
+ * tests/test_time.c, include/asterisk/localtime.h,
+ main/stdtime/localtime.c: Fix test_time on Mac OS X (and other
+ platforms without inotify) Reviewboard:
+ https://reviewboard.asterisk.org/r/554/
+
+2010-03-16 19:01 +0000 [r252767] Russell Bryant <russell@digium.com>
+
+ * utils/Makefile, /: Merged revisions 252766 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r252766 | russell | 2010-03-16 14:00:43 -0500 (Tue, 16 Mar 2010)
+ | 6 lines Don't treat warnings as errors for muted. muted
+ supports OS X, but uses functions marked as deprecated in 10.6.
+ However, the functions are still supported, so just ignore the
+ warnings for now and allow the build to proceed. ........
+
+2010-03-16 18:48 +0000 [r252762] Leif Madsen <lmadsen@digium.com>
+
+ * configs/extensions.ael.sample: Merged revisions 252761 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r252761 | lmadsen | 2010-03-16 13:46:20 -0500 (Tue, 16 Mar 2010)
+ | 7 lines Additional extensions.ael global variable fixes. Fixing
+ up a couple more overlapping global variable namespaces shared
+ with extensions.conf.sample. Also noticed a few of the lines that
+ were commented out didn't have the closing semi-colon so I added
+ that as well. (issue #17035) ........
+
+2010-03-16 18:40 +0000 [r252760] Tilghman Lesher <tlesher@digium.com>
+
+ * codecs/gsm/Makefile: OSARCH is not inherited to this directory
+
+2010-03-16 18:36 +0000 [r252759] Russell Bryant <russell@digium.com>
+
+ * tests/test_time.c: Disable this test on non-Linux for now.
+
+2010-03-15 22:48 +0000 [r252709] Kevin P. Fleming <kpfleming@digium.com>
+
+ * res/res_fax.c: Improve handling of values supplied to
+ FAXOPT(ecm). Previously, values that began with whitespace were
+ silently treated as 'no', and all non-'yes' values were also
+ treated as 'no'. Now the supplied value is specifically checked
+ for a 'yes' or 'no' (or equivalent) value, after skipping leading
+ whitespace. If the value is not valid, then a warning message is
+ generated.
+
+2010-03-15 22:14 +0000 [r252627] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Tell the RTP engine API about the initial
+ read and write format. Peer reviewed out-of-band by file.
+
+2010-03-15 21:55 +0000 [r252623] Sean Bright <sean@malleable.com>
+
+ * apps/app_meetme.c: Resolve a crash in SLATrunk when the specified
+ trunk doesn't exist. Reported by philipp64 in #asterisk-dev.
+
+2010-03-15 21:51 +0000 [r252619] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/org.asterisk.asterisk.plist, /: Merged revisions
+ 252617 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r252617 | tilghman | 2010-03-15 16:43:14 -0500 (Mon, 15 Mar 2010)
+ | 2 lines Uh, yeah. Umask. I'm stupid. ........
+
+2010-03-15 20:52 +0000 [r252534] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/extensions.ael.sample: Merged revisions 252533 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r252533 | lmadsen | 2010-03-15 15:48:56 -0500 (Mon, 15 Mar 2010)
+ | 7 lines Update extensions.ael file to not overlap
+ extensions.conf. Updated the extensions.ael file so the global
+ variables don't overlap those that we have in extensions.conf
+ (sample files). This way unexpected things won't happed hopefully
+ if both pbx_ael and res_config are loaded. (closes issue #17035)
+ Reported by: pprindeville ........
+
+2010-03-15 16:27 +0000 [r252362-252488] Tilghman Lesher <tlesher@digium.com>
+
+ * codecs/gsm/Makefile: Make the Makefile logic more explicit and
+ move the Snow Leopard logic down to where it's not executed on
+ non-Darwin systems. (closes issue #17028) Reported by: pabelanger
+ Patches: issue17028_20100315.patch uploaded by seanbright
+ (license 71) 20100315__issue17028.diff.txt uploaded by tilghman
+ (license 14) Tested by: tilghman, pabelanger
+
+ * channels/chan_sip.c: THIS IS NOT PYTHON. Indentation doesn't
+ matter, only braces do. (closes issue #17025) Reported by:
+ smurfix Patches: sip.patch uploaded by smurfix (license 547)
+
+ * /: Recorded merge of revisions 252366 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r252366 | tilghman | 2010-03-14 20:39:00 -0500 (Sun, 14 Mar 2010)
+ | 2 lines Typo ........
+
+ * Makefile, contrib/init.d/org.asterisk.asterisk.plist (added), /,
+ main/asterisk.c: Merged revisions 252361 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r252361 | tilghman | 2010-03-14 20:33:50 -0500 (Sun, 14 Mar 2010)
+ | 4 lines Launch Asterisk on Mac OS X with launchd. Reviewboard:
+ https://reviewboard.asterisk.org/r/551/ ........
+
+2010-03-14 17:43 +0000 [r252314] Sean Bright <sean@malleable.com>
+
+ * cdr/cdr_sqlite3_custom.c, cel/cel_sqlite3_custom.c: Fix building
+ CDR and CEL SQLite3 modules. They added a sqlite3_log() function
+ which was conflicting with our function names. (closes issue
+ #17017) Reported by: alephlg
+
+2010-03-14 14:42 +0000 [r252277] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
+ addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooCalls.h,
+ configs/chan_ooh323.conf.sample, addons/ooh323c/src/ooh245.h,
+ addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/ootypes.h,
+ addons/ooh323c/src/ooq931.c: generate roundtrip delay requests
+ and responses added response to roundtrip delay requests from
+ opposite side added roundtrip delay request sending to opposite
+ side after answer, added options for sending request (interval
+ between request and count of unreplied requests before forced
+ call hangup) (closes issue #16976) Reported by: vmikhelson
+ Patches: rtdr-1.6.0-2.patch uploaded by may213 (license 454)
+ Tested by: vmikhelson, may213
+
+2010-03-13 22:21 +0000 [r252229-252241] Russell Bryant <russell@digium.com>
+
+ * main/app.c: Resolve unit test failure that occurred on Mac OSX.
+ On Linux (glibc), regcomp() does not return an error for an empty
+ string. However, the version on OSX will return an error. The
+ test for channel group matching by regex now passes on the mac,
+ as well.
+
+ * tests/test_time.c: Resolve compiler warning by paying attention
+ to system() return value. This resolves the last compile failure
+ on bamboo.
+
+2010-03-12 23:18 +0000 [r252133] Tilghman Lesher <tlesher@digium.com>
+
+ * tests/test_time.c (added): Test script to verify that timezone
+ cache is properly removed on zonefile alteration.
+
+2010-03-12 22:04 +0000 [r252089] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, res/res_rtp_asterisk.c, addons/chan_ooh323.c,
+ main/rtp_engine.c, channels/chan_sip.c, channels/chan_skinny.c,
+ channels/chan_h323.c, configs/sip.conf.sample,
+ include/asterisk/frame.h, include/asterisk/rtp_engine.h,
+ channels/sip/include/sip.h, channels/chan_mgcp.c: Only change the
+ RTP ssrc when we see that it has changed This change basically
+ reverts the change reviewed in
+ https://reviewboard.asterisk.org/r/374/ and instead limits the
+ updating of the RTP synchronization source to only those times
+ when we detect that the other side of the conversation has
+ changed the ssrc. The problem is that SRCUPDATE control frames
+ are sent many times where we don't want a new ssrc, including
+ whenever Asterisk has to send DTMF in a normal bridge. This is
+ also not the first time that this mistake has been made. The
+ initial implementation of the ast_rtp_new_source function also
+ changed the ssrc--and then it was removed because of this same
+ issue. Then, we put it back in again to fix a different issue.
+ This patch attempts to only change the ssrc when we see that the
+ other side of the conversation has changed the ssrc. It also
+ renames some functions to make their purpose more clear. Review:
+ https://reviewboard.asterisk.org/r/540/
+
+2010-03-12 21:57 +0000 [r252088] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c: add missing mfcr2_skip_category setting
+
+2010-03-12 19:43 +0000 [r251989] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Don't override a user option with the
+ global option. (closes issue #16849) Reported by: ip-rob Patches:
+ 20100311__issue16849.diff.txt uploaded by tilghman (license 14)
+ Tested by: ip-rob
+
+2010-03-12 19:40 +0000 [r251946-251987] Richard Mudgett <rmudgett@digium.com>
+
+ * /: Merged revisions 251986 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r251986 | rmudgett | 2010-03-12 13:33:22 -0600 (Fri, 12 Mar 2010)
+ | 1 line Make chan_dahdi wakeup_sub() prototype not conditional.
+ ........
+
+ * channels/chan_dahdi.c: Doxegen this chan_dahdi lock.
+
+2010-03-11 21:07 +0000 [r251877-251884] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_exec.c: Because ExecIf needs to reprocess arguments,
+ it's best if we don't remove quotes during parsing. (closes issue
+ #16905) Reported by: ip-rob Patches:
+ 20100303__issue16905.diff.txt uploaded by tilghman (license 14)
+ Tested by: ip-rob
+
+ * tests/test_stringfields.c: Fix tests on 32-bit systems.
+
+ * apps/app_system.c: If the argument to the system application is
+ quoted, ensure we remove the quotes before trying to execute.
+ (closes issue #16842) Reported by: ip-rob Patches:
+ 20100310__issue16842.diff.txt uploaded by tilghman (license 14)
+ Tested by: ip-rob
+
+2010-03-11 18:07 +0000 [r251821] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c: Minor tweaks and
+ comment updates to chan_dahdi.
+
+2010-03-11 07:03 +0000 [r251779] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * apps/app_directory.c: Add supporting code for app-directory pause
+ option. Since 1.6.1 CLI help reports that option p(n) 'initial
+ pause' is available. Supporting code was never implemented.
+ (closes issue #16751) Reported by: alecdavis Patches:
+ directory_pause.trunk.diff.txt uploaded by alecdavis (license
+ 585) Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/481/
+
+2010-03-10 23:15 +0000 [r251736] Jeff Peeler <jpeeler@digium.com>
+
+ * tests/test_stringfields.c (added), main/utils.c: Add new unit
+ test for stringfields. (Copied from reviewboard) Tests the
+ following: 1. Basic allocation and setting of string fields. 2.
+ Shrinking a string field and re-expanding it. 3. Growing the last
+ allocation in a string field pool. 4. Setting a string to a large
+ value such that a new string field pool must be allocated. In
+ each part, we make sure that the string field is accurate (has
+ the correct value in it), make sure that the 2 bytes before the
+ string field has the correct capacity for the field, and for
+ tests 2-4, we make sure that the string field is where we expect
+ it to be in memory. Also tested: 5. Shrinking a string field and
+ partially re-expanding it. 6. Setting strings in such a way as to
+ create three separate string field pools and then removing the
+ middle pool. There is a bug fix in the init function, which
+ ensures the embedded_pool is set to NULL which is important for
+ stack allocated structures. Review:
+ https://reviewboard.asterisk.org/r/185/
+
+2010-03-10 20:54 +0000 [r251682] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_strings.c: Hmmm, apparently needed to be fixed in
+ trunk, too. (closes issue #16900) Reported by: bluecrow76
+ Patches: asterisk-1.6.2.4-func_strings.diff uploaded by
+ bluecrow76 (license 270)
+
+2010-03-10 20:53 +0000 [r251680] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_record.c: Be less ambiguous in Record() app docs. For
+ some reason the documentation for the 'k' application in trunk
+ and 1.6.2 is different than 1.6.0 and 1.6.1, so I'm setting them
+ all to match. The wording in 1.6.2 and trunk was ambiguous, so
+ you could interpret the wording the mean that recording would
+ continue upon hangup indefinitely, or you could interpret it to
+ mean that the recorded data would not be discarded upon hangup.
+ This change makes it clear we mean the latter, and not the
+ former. Came from a discussion in #asterisk on IRC.
+
+2010-03-10 20:51 +0000 [r251679] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c: Fix ParkAndAnnounce not respecting parking
+ options. The patch ensures that if a peer does not exist, parking
+ settings are read from the channel. A unit test has been written
+ to ensure proper operation for both standard parking and parking
+ using masquerades. (closes issue #16592) Reported by: mwyres
+ Patches: bug_16592.diff uploaded by snuffy (license 35) Review:
+ https://reviewboard.asterisk.org/r/539/
+
+2010-03-10 20:30 +0000 [r251677] Tilghman Lesher <tlesher@digium.com>
+
+ * tests/test_substitution.c, funcs/func_strings.c: It's amazing
+ what writing a test will find. (issue #16900) Reported by:
+ bluecrow76
+
+2010-03-10 18:25 +0000 [r251631] Jeff Peeler <jpeeler@digium.com>
+
+ * main/abstract_jb.c: Fix jitterbuffer logging not creating
+ logfiles. Three changes made here: 1) Do not fail if a previous
+ log does not exist (in fact, this is probably expected). 2)
+ Ensure that the file descriptor to write to gets assigned
+ properly. I am at a loss as to why assigning safe_fd outside the
+ if fixes this, but it makes the if statement slightly less
+ complicated anyway. 3) Move up the failure message so that the
+ errno of the failure is not overwritten by fclose. (closes issue
+ #16917) Reported by: Artem
+
+2010-03-10 16:55 +0000 [r251538-251585] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h, channels/sig_pri.c: Simplified
+ dahdi_request() channel selection failed reason/cause code. Also
+ avoid potential crash because cause could be NULL.
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
+ Reduce the amount of database access for
+ HAVE_PRI_SERVICE_MESSAGES. Rework HAVE_PRI_SERVICE_MESSAGES to
+ not use the active values directly from the database. Database
+ access is likely expensive. Database access now only happens on
+ initialization, destruction, and when the B channel is taken in
+ or out of service. This change is not related to call waiting but
+ it would cause the search for a call waiting interface to be very
+ expensive and slow down D channel message servicing.
+
+2010-03-09 20:30 +0000 [r251475] Tilghman Lesher <tlesher@digium.com>
+
+ * codecs/gsm/Makefile, Makefile.rules: Build system modifications
+ to ensure that Asterisk properly builds on Mac OS X 10.6. (closes
+ issue #16997) Reported by: jquinn Patches:
+ 20100309__issue16997__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: tilghman, russell
+
+2010-03-08 18:08 +0000 [r251310] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 251309 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r251309 | lmadsen | 2010-03-08 12:07:44 -0600 (Mon, 08 Mar 2010)
+ | 13 lines Fix Debian init script to not use -c. When using the
+ init script as-is currently, it could cause issues on Debian such
+ as high CPU usage. This fix has worked for several people so I'm
+ implementing the change. (closes issue #16784) Reported by:
+ pabelanger Tested by: pabelanger, mnick, davidw, mutineer612
+ (closes issue #16887) Reported by: jlpedrosa Tested by:
+ jlpedrosa, mutineer612 ........
+
+2010-03-08 05:15 +0000 [r251262-251263] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ main/stdtime/localtime.c: Remove portions that weren't meant to
+ be committed for the OS X compat fix
+
+ * funcs/func_pitchshift.c, configure,
+ include/asterisk/autoconfig.h.in, main/Makefile, configure.ac,
+ main/stdtime/localtime.c: Change needed to make Mac OS X 10.6
+ happy
+
+2010-03-07 14:53 +0000 [r251221-251222] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: Clean transmit_* for start/stop media
+ transmission Small patch changing skinny_set_rtp_peer to use
+ transmit_stopmediatransmission and to use new
+ transmit_startmediatransmission. Basic testing on 30VIP's by
+ wedhorn Basic testing on 7960 by me (closes issue #16956)
+ Reported by: wedhorn Patches: skinny-clean05b.diff uploaded by
+ wedhorn (license 30) Tested by: wedhorn,mvanbaak
+
+ * channels/chan_skinny.c: Cleanup transmit_callstate handling Broke
+ the various functions included in transmit_callstate to their own
+ functions. Transmit_callstate now just transmits callstate.
+ Generally left the functionality as it was, which highlight some
+ minor code issues (eg multiple transmit_callstate's). I did
+ however revise the hint code usage of the old transmit_callstate
+ as it it not appropriate to put a device on hook based on the
+ change of a hinted device. (closes issue #16939) Reported by:
+ wedhorn Patches: skinny-clean04.diff uploaded by wedhorn (license
+ 30) Tested by: mvanbaak,wedhorn
+
+2010-03-07 00:45 +0000 [r251181] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooq931.c: small log issue from bug 0016664
+
+2010-03-06 14:16 +0000 [r251137] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Fix a crash in SIP blind transfer handling
+ found by an automated external test. The first real test added to
+ the external test suite found a pretty nasty crash that occurred
+ in Asterisk trunk. The crash was due to a race condition between
+ the REFER handling and channel destruction in the channel thread.
+ After the transfer has been completed, we go back to the
+ transferrer channel and try to lock it so we can fire off a CEL
+ event. However, there was no guarantee that the channel was still
+ around at that point since it's racing against the channel
+ thread. Since ast_channel is a reference counted object, the fix
+ is simple. The code unlocks the transferrer channel before
+ finally completing the transfer with an async goto. At this point
+ the channel thread is going to start call tear down and the
+ channel will eventually be destroyed. To ensure that the channel
+ is valid when we want to fire off the CEL event, increase the
+ channel's reference count.
+
+2010-03-05 21:51 +0000 [r251038-251087] David Vossel <dvossel@digium.com>
+
+ * funcs/func_pitchshift.c: fixes xml error in func_pitchshift
+
+ * funcs/func_pitchshift.c (added), CHANGES: PITCH_SHIFT dialplan
+ function The PITCH_SHIFT function can be used on a channel to
+ independently modify the pitch of both rx and tx audio streams.
+ Now you can improve your conference calls by assigning a random
+ pitch effect to everyone entering a meetme room, or just make
+ your day more interesting by making your co-workers sound funny.
+ These are just some of the numerious practical uses for this
+ function. Enjoy! https://reviewboard.asterisk.org/r/526/
+
+2010-03-05 19:32 +0000 [r251022] Russell Bryant <russell@digium.com>
+
+ * build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, makeopts.in,
+ pbx/pbx_gtkconsole.c (removed): Remove pbx_gtkconsole and related
+ gtk1 checks. Review: https://reviewboard.asterisk.org/r/541/
+
+2010-03-05 19:10 +0000 [r250979] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_followme.c: Fix app_followme playing wrong sound files.
+ Fixes regression introduced in 140167 that uses the wrong
+ variable names. (closes issue #16930) Reported by: ianc Patches:
+ fix_reload_followme.diff uploaded by ianc (license 998)
+
+2010-03-05 05:03 +0000 [r250917] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Fix up some of chan_sip's usage of the RTP
+ engine API. The get_local_address() function for an RTP instance
+ was used when building an SDP, but the results were not honored.
+ The RTP engine activate() function was not being used once we
+ have determined that media will now flow.
+
+2010-03-05 04:37 +0000 [r250913] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Missing quote in ODBC query. (closes issue
+ #16953) Reported by: elguero Patches:
+ app_voicemail-odbc-syntax-fix.diff uploaded by elguero (license
+ 37)
+
+2010-03-05 02:07 +0000 [r250871] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/rtp_engine.h: Fix up the ast_rtp_property enum.
+ The mis-placement of the latest entry meant that when it was set,
+ it was writing one index past the end of the properties array in
+ the ast_rtp_instance (which happened to be the local_address
+ field).
+
+2010-03-05 01:05 +0000 [r250787] Jeff Peeler <jpeeler@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 250786 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r250786 | jpeeler | 2010-03-04 19:02:58 -0600 (Thu, 04
+ Mar 2010) | 9 lines Fix not being able to specify a URL in MOH
+ class directory. Don't attempt to chdir on a URL! (closes issue
+ #16875) Reported by: raarts Patches: moh-http.patch uploaded by
+ raarts (license 937) ........
+
+2010-03-04 20:12 +0000 [r250730] Mark Michelson <mmichelson@digium.com>
+
+ * funcs/func_channel.c: Adjust XML for func_channel to indicate
+ that rtpdest can take a "text" argument.
+
+2010-03-03 21:28 +0000 [r250609-250614] Leif Madsen <lmadsen@digium.com>
+
+ * /: Recorded merge of revisions 250613 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r250613 | lmadsen | 2010-03-03 16:28:02 -0500 (Wed, 03 Mar 2010)
+ | 11 lines Update existing Local channel documentation. A
+ complete re-write of the Local channel documentation has been
+ performed, with the existing information from localchannel.txt
+ and localchannel.tex merged in. (issue #16637) Reported by: kobaz
+ Patches: localchannel.tex uploaded by lmadsen (license 10)
+ localchannel.txt uploaded by lmadsen (license 10) Tested by:
+ lmadsen, jsmith, mmichelson ........
+
+ * doc/tex/localchannel.tex: Update existing Local channel
+ documentation. A complete re-write of the Local channel
+ documentation has been performed, with the existing information
+ from localchannel.txt and localchannel.tex merged in. (closes
+ issue #16637) Reported by: kobaz Patches: localchannel.tex
+ uploaded by lmadsen (license 10) localchannel.txt uploaded by
+ lmadsen (license 10) Tested by: lmadsen, jsmith, mmichelson
+
+2010-03-03 19:38 +0000 [r250565] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, channels/chan_dahdi.c, main/dial.c,
+ channels/chan_local.c, include/asterisk/channel.h,
+ apps/app_queue.c: Removed cdrflags from ast_channel structure.
+ Only chan_dahdi set a value in cdrflags. Everyone else just
+ copied it around the system. Noone cared about any value it may
+ have contained.
+
+2010-03-03 19:06 +0000 [r250481] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+ 250480 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r250480 | jpeeler | 2010-03-03 13:04:11 -0600 (Wed, 03 Mar 2010)
+ | 15 lines Make sure to clear red alarm after polarity reversal.
+ From the issue: The automatic overnight line tests (or manual
+ ones) used on UK (BT) lines causes a red alarm on a dahdi /
+ TDM400P connected channel. This is because the line uses voltage
+ tests (battery loss) and polarity reversal. The polarity reversal
+ causes chan_dahdi to initiate v23 CallerID processing but during
+ this the event DAHDI_EVENT_NOALARM is ignored so that the alarm
+ is never cleared. (closes issue #14163) Reported by: jedi98
+ Patches: chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license
+ 653) Tested by: mattbrown, Chainsaw, mikeeccleston ........
+
+2010-03-03 19:02 +0000 [r250395-250478] David Vossel <dvossel@digium.com>
+
+ * main/test.c: Changes 0ms to <1ms in cli END results during 'test
+ execute'
+
+ * /, channels/chan_iax2.c: Merged revisions 250394 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r250394 | dvossel | 2010-03-03 12:02:27 -0600 (Wed, 03
+ Mar 2010) | 16 lines fixes problem with duplicate TXREQ packets
+ When Asterisk receives an IAX2 TXREQ packet, try_transfer() will
+ call store_by_transfercallno() to link the chan_iax2_pvt struct
+ into iax_transfercallno_pvts. If a duplicate TXREQ packet is
+ received for the same call, the pvt struct will be linked into
+ iax_transfercallno_pvts multiple times. This patch fixes this.
+ Thanks rain for debugging this and providing a patch! (closes
+ issue #16904) Reported by: rain Patches:
+ iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested
+ by: rain, dvossel ........
+
+2010-03-03 17:37 +0000 [r250392] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, CHANGES:
+ Add new config option to control AMI alarm event reporting in
+ chan_dahdi. New config parameter "reportalarms" added in
+ chan_dahdi.conf which supports the following possible values:
+ "channels": report each channel alarms (current behavior, default
+ for backward compatibility) "spans": report an "SpanAlarm" event
+ when the span of any configured channel is alarmed "all": report
+ channel and span alarms (aggregated behavior) "none": do not
+ report any alarms (closes issue #16709) Reported by: nahuelgreco
+ Patches: chan_dahdi.c.reportalarms.patch uploaded by nahuelgreco
+ (license 162)
+
+2010-03-03 16:43 +0000 [r250303-250346] Tilghman Lesher <tlesher@digium.com>
+
+ * main/editline/configure: One more fix to editline
+
+ * main/editline/configure, main/editline/Makefile.in,
+ main/editline/sys.h, main/editline/configure.in: Eliminate
+ remaining libedit warnings (shown in bamboo)
+
+2010-03-03 15:39 +0000 [r250302] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_fax.c, apps/app_fax.c, CHANGES, res/res_fax_spandsp.c:
+ Updated CHANGES file to mention res_fax and res_fax_spandsp. Also
+ fixed MODULEINFO depends and conflicts for app_fax, res_fax, and
+ res_fax_spandsp.
+
+2010-03-03 00:18 +0000 [r250235-250246] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes signed to unsigned int comparision
+ issue for FaxMaxDatagram value.
+
+ * main/test.c: fixes assumption that test failed if it did not pass
+ when generating results
+
+ * tests/test_utils.c: base64 unit test
+
+2010-03-02 23:22 +0000 [r250190-250213] Matthew Nicholson <mnicholson@digium.com>
+
+ * configs/res_fax.conf.sample (added), include/asterisk/res_fax.h
+ (added): Merge missed files from res_fax/res_fax_spandsp merge.
+
+ * res/res_fax.c (added), res/res_fax.exports (added),
+ include/asterisk/frame.h, res/res_fax_spandsp.c (added): Merge
+ res_fax and res_fax_spandsp.
+
+2010-03-02 21:58 +0000 [r250141] David Vossel <dvossel@digium.com>
+
+ * apps/app_directed_pickup.c, CHANGES: adds 'p' option to
+ PickupChan The 'p' option allows the PickupChan app to pickup a
+ ringing phone by looking for the first match to a partial channel
+ name rather than requiring a full match. (closes issue #16613)
+ Reported by: syspert Patches: pickipbycallid.patch uploaded by
+ syspert (license 938) pickupbycallerid_v2.patch uploaded by
+ dvossel (license 671) Tested by: dvossel, syspert
+
+2010-03-02 21:09 +0000 [r249950-250051] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/imapstorage.tex: Update IMAP documentation. Update the
+ IMAP documentation to make it clear that storing voicemails in
+ the same folder as a large number of emails could potentially
+ cause significant slow downs when writing or retrieving
+ voicemails. (issue #16704) Reported by: TimeHider Tested by:
+ lmadsen, TimeHider
+
+ * /, configs/cdr.conf.sample: Merged revisions 250043 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02
+ Mar 2010) | 7 lines Update documentation to clarify purpose of
+ unanswered option. (closes issue #16267) Reported by: elsto
+ Patches: cdr.conf.sample.patch.txt uploaded by lmadsen (license
+ 10) Tested by: davidw, elsto ........
+
+ * /: Recorded merge of revisions 250041 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r250041 | lmadsen | 2010-03-02 15:45:37 -0500 (Tue, 02 Mar 2010)
+ | 4 lines Update documentation to not imply we support overriding
+ options. (issue #16855) Reported by: davidw ........
+
+ * doc/tex/configuration.tex: Update documentation to not imply we
+ support overriding options. (closes issue #16855) Reported by:
+ davidw
+
+ * apps/app_directory.c: Fix literal values wrapped in
+ documentation. (closes issue #16145) Reported by: tilghman
+
+2010-03-02 19:39 +0000 [r249947] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * apps/app_echo.c: revert ability to exit echo app caused a
+ regression, as only supported VOICE, not VIDEO etc. (issue
+ #16880)
+
+2010-03-02 19:24 +0000 [r249912-249925] Leif Madsen <lmadsen@digium.com>
+
+ * main/features.c: Add missing description of the PARKINGLOT
+ variable in XML documentation. (closes issue #16743) Reported by:
+ snuffy Patches: parkingdoc.diff uploaded by snuffy (license 35)
+
+ * pbx/pbx_dundi.c: Convert some DUNDI functions to XML
+ documentation. (closes issue #16798) Reported by: snuffy Patches:
+ xml_dundi.diff uploaded by snuffy (license 35)
+
+2010-03-02 19:08 +0000 [r249893] David Vossel <dvossel@digium.com>
+
+ * channels/chan_unistim.c, configs/chan_dahdi.conf.sample,
+ configs/console.conf.sample, channels/chan_local.c,
+ channels/chan_sip.c, configs/oss.conf.sample,
+ configs/usbradio.conf.sample, configs/misdn.conf.sample,
+ channels/chan_console.c, channels/chan_gtalk.c,
+ channels/chan_oss.c, channels/misdn_config.c,
+ include/asterisk/abstract_jb.h, configs/alsa.conf.sample,
+ channels/chan_jingle.c, channels/chan_usbradio.c,
+ channels/chan_dahdi.c, channels/chan_skinny.c,
+ configs/mgcp.conf.sample, main/abstract_jb.c,
+ channels/chan_h323.c, channels/chan_alsa.c,
+ configs/sip.conf.sample, channels/chan_mgcp.c: fixes adaptive
+ jitterbuffer configuration When configuring the adaptive
+ jitterbuffer, the target_extra value not only could not be set
+ from the configuration, but was not even being set to its proper
+ default. This value is required in order for the adaptive
+ jitterbuffer to work correctly. To resolve this a config option
+ has been added to expose this value to the conf files, and a
+ default value is provided when no config specific value is
+ present.
+
+2010-03-02 19:02 +0000 [r249892] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_osplookup.c, apps/app_confbridge.c, res/res_jabber.c:
+ Fix several XML documentation validate errors.
+
+2010-03-02 18:31 +0000 [r249889-249891] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c: fix build by checking result of symlink in
+ test_voicemail_vmsayname
+
+ * CHANGES, apps/app_voicemail.c: Add new application VMSayName for
+ use with voicemail. VMSayName that will play the recorded name of
+ the voicemail user if it exists, otherwise will play the mailbox
+ number. A unit test has been written to verify correct
+ functionality called test_voicemail_vmsayname. (closes issue
+ #14973) Reported by: ghjm Review:
+ https://reviewboard.asterisk.org/r/530/
+
+2010-03-02 07:38 +0000 [r249759-249801] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * apps/app_echo.c: fixes ability to exit echo app when called from
+ a ISDN channel, null frames prevent '#' exit. Now only echo back
+ VOICE and DTMF frames (issue #16880) Reported by: alecdavis
+ Patches: echo_exit.diff.txt uploaded by alecdavis (license 585)
+ Tested by: alecdavis
+
+ * channels/chan_dahdi.c: fix asterisk setting of pritimers from
+ chan_dahdi.conf regression since sig_pri split. (issue #16909)
+ Reported by: alecdavis Patches: pritimer.asterisk.diff.txt
+ uploaded by alecdavis (license 585) Tested by: alecdavis
+
+2010-03-01 19:36 +0000 [r249672] Sean Bright <sean@malleable.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 249671 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r249671 | seanbright | 2010-03-01 14:35:01 -0500 (Mon,
+ 01 Mar 2010) | 11 lines Fix crash in app_voicemail related to
+ message counting. We were passing a 'struct inprocess **' and
+ treating it like a 'struct inprocess *' causing a segfault.
+ (closes issue #16921) Reported by: whardier Patches:
+ 20100301_issue16921.patch uploaded by seanbright (license 71)
+ Tested by: whardier ........
+
+2010-03-01 19:33 +0000 [r249669-249670] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: Cleanup display_*message functions. This
+ patch splits transmit_displaymessage into
+ transmit_clear_display_message and transmit_display_message which
+ better aligns with the skinny protocol. The new
+ transmit_display_message is not used in the current code, but
+ will be and so it is commented. Moved handle_datetime from this
+ function to onhook and offhook functions (display now properly
+ cleared at the end of a call on 30VIP). Removed skinny debug
+ messages from inline code as there's an ast_verb in
+ transmit_clear_display_message. Also, removed commentary that it
+ was a clear display as it is now apparent from the function name.
+ Split transmit_displaypromptmessage into display and clear.
+ (closes issue #16878) Reported by: wedhorn Patches:
+ skinny-clean02.diff uploaded by wedhorn (license 30)
+ skinny-clean03.diff uploaded by wedhorn (license 30)
+
+ * channels/chan_skinny.c: fix endianes issues in chan_skinny
+ (closes issue #16826) Reported by: PipoCanaja Patches:
+ chan_skinny.c_bigendianPatch_20100218.diff uploaded by PipoCanaja
+ (license 994) Tested by: wedhorn
+
+2010-03-01 18:36 +0000 [r249623] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Constify a bit of app_voicemail, to make
+ ODBC and IMAP compile once again.
+
+2010-03-01 17:11 +0000 [r249538] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 249536 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r249536 | jpeeler | 2010-03-01 11:02:03 -0600 (Mon, 01
+ Mar 2010) | 11 lines Modify queued frames from local channels to
+ not set the other side to up In this case, attended transfers
+ were broken due to ast_feature_request_and_dial detecting the
+ channel being set to up before the answer frame could be read and
+ therefore failing to mark the channel as ready. This fix is a
+ regression fix for 244785, which should continue to work properly
+ as well. (closes issue #16816) Reported by: jamhed Tested by:
+ jamhed, corruptor ........
+
+2010-02-28 20:50 +0000 [r249491] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Fix unit test that Alec Davis broke.
+ (closes issue #16927) Reported by: alecdavis
+
+2010-02-28 16:36 +0000 [r249449] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * apps/app_voicemail.c: make unit test check for NULL folder, which
+ then defaults to INBOX previous test, gave false level of
+ assurance that code was healthy. (issue #16927) Reported by:
+ alecdavis Patches: based on app_voicemail_test.diff.txt uploaded
+ by alecdavis (license 585) Tested by: alecdavis
+
+2010-02-28 07:10 +0000 [r249405] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/app.h, apps/app_voicemail.c: Properly document
+ voicemail API documents. Also fix a crash reported via the -dev
+ list.
+
+2010-02-27 22:49 +0000 [r249320] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/sig_pri.c: overlap receiving: automatically send CALL
+ PROCEEDING when dialplan starts Following Q.931 5.2.4 When the
+ user has determined that sufficient call information has been
+ received the user shall stop T302 and send CALL PROCEEDING to the
+ network. Previously timeouts were possible if the dialplan took a
+ long time to issue any response back to the network. Verified
+ that our local TELCO also does the same. (issue #16789) Reported
+ by: alecdavis Patches: overlap_receiving_trunk.diff.txt uploaded
+ by alecdavis (license 585) Tested by: alecdavis
+
+2010-02-27 14:08 +0000 [r249235] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 249234 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r249234 | kpfleming | 2010-02-27 09:07:59 -0500 (Sat, 27
+ Feb 2010) | 1 line add a reference to the now-published IAX2 RFC
+ ........
+
+2010-02-26 18:41 +0000 [r249187] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Cleanups to fix bugs in the VM count API
+ functions. - Urgent voicemails were not attached, because the
+ attachment code looked in the wrong folder. - Urgent voicemails
+ were sometimes counted twice when displaying the count of new
+ messages. - Backends were inconsistent as to which voicemails
+ each API counted. - Unit tests added to verify behavior in the
+ future. (closes issue #15654) Reported by: tomo1657 Patches:
+ 20100225__issue15654.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman (closes issue #16448) Reported by: hevad
+ Review: https://reviewboard.asterisk.org/r/525/
+
+2010-02-26 18:41 +0000 [r249186] David Vossel <dvossel@digium.com>
+
+ * main/test.c: adds Time field to "test show results" cli command
+
+2010-02-26 17:13 +0000 [r249101-249105] Mark Michelson <mmichelson@digium.com>
+
+ * main/features.c: Send a manager event when the manager
+ BridgeAction command is used. (closes issue #16769) Reported by:
+ syspert Patches: bridgeaction.patch uploaded by syspert (license
+ 938)
+
+ * /, channels/chan_sip.c: Merged revisions 249100 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r249100 | mmichelson | 2010-02-26 11:04:29 -0600 (Fri, 26 Feb
+ 2010) | 8 lines For T.38 reINVITEs treat a 606 the same as a 488.
+ (closes issue #16792) Reported by: vrban Patches: t38_606.patch
+ uploaded by vrban (license 756) ........
+
+2010-02-26 08:45 +0000 [r249009-249058] Russell Bryant <russell@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c, cdr/cdr_syslog.c, cdr/cdr_sqlite.c,
+ cdr/cdr_adaptive_odbc.c, cdr/cdr_pgsql.c, cdr/cdr_odbc.c,
+ cdr/cdr_radius.c, cdr/cdr_custom.c, cdr/cdr_manager.c,
+ cdr/cdr_tds.c, cdr/cdr_csv.c: formatting tweaks and
+ constification
+
+ * main/cdr.c: Trim trailing whitespace (to help reduce diff against
+ cdr-q branch)
+
+ * include/asterisk/cdr.h: Trim trailing whitespace, convert lists
+ of defines to enums
+
+ * cdr/cdr_sqlite.c: trivial formatting tweak (working on reducing
+ diff against trunk for cdr-q)
+
+ * cdr/cdr_sqlite3_custom.c: remove include
+
+ * cdr/cdr_csv.c: constification, remove include
+
+ * cdr/cdr_tds.c: Remove unnecessary includes, formatting tweak
+
+ * cdr/cdr_pgsql.c: constification and remove unnecessary include
+
+2010-02-25 23:09 +0000 [r248952] Jeff Peeler <jpeeler@digium.com>
+
+ * /, res/res_monitor.c: Merged revisions 248860 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r248860 | jpeeler | 2010-02-25 15:22:06 -0600 (Thu, 25 Feb 2010)
+ | 18 lines Ensure that monitor recordings are written to the
+ correct location (again) This is an extension to 248757. As such
+ the dialplan test has been extended: exten => 5040, 1,
+ monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
+ dial(sip/5001) exten => 5041, 1,
+ monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
+ dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
+ exten => 5042, n, dial(sip/5001) exten => 5043, 1,
+ monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n,
+ changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001)
+ exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n,
+ changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by
+ design and emits a warning exten => 5044, n, dial(sip/5001)
+ ........
+
+2010-02-25 22:41 +0000 [r248946] Mark Michelson <mmichelson@digium.com>
+
+ * main/acl.c: Fix incorrect ACL behavior when CIDR notation of "/0"
+ is used. AST-2010-003
+
+2010-02-25 21:22 +0000 [r248861] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/asterisk.c: Merged revisions 248859 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r248859 | tilghman | 2010-02-25 15:21:05 -0600 (Thu, 25 Feb 2010)
+ | 15 lines Some platforms clear /var/run at boot, which makes
+ connecting a remote console... difficult. Previously, we only
+ created the default /var/run/asterisk directory at install time.
+ While we could create it in the init script, that would not work
+ for those who start asterisk manually from the command line. So
+ the safest thing to do is to create it as part of the Asterisk
+ boot process. This also changes the ownership of the directory,
+ because the pid and ctl files are created after we setuid/setgid.
+ (closes issue #16802) Reported by: Brian Patches:
+ 20100224__issue16802.diff.txt uploaded by tilghman (license 14)
+ Tested by: tzafrir ........
+
+2010-02-25 18:37 +0000 [r248793] Jeff Peeler <jpeeler@digium.com>
+
+ * /, res/res_monitor.c: Merged revisions 248757 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010)
+ | 15 lines Ensure that monitor recordings are written to the
+ correct location. Recordings should be placed in the monitor
+ directory when a non-absolute path is used. Exact dialplan used
+ for testing: exten => 5040, 1,
+ monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
+ dial(sip/5001) exten => 5041, 1,
+ monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
+ dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
+ exten => 5042, n, dial(sip/5001) ABE-2101 ........
+
+2010-02-24 22:44 +0000 [r248584-248667] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/Makefile: Also kill the .i files, or else the build
+ process will not recreate them, when we change flags. Fixes a
+ weird symbol problem mmichelson was having in a group branch, but
+ also applies to trunk.
+
+ * /, main/logger.c, include/asterisk/term.h, main/term.c: Merged
+ revisions 248582 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r248582 | tilghman | 2010-02-24 15:02:18 -0600 (Wed, 24 Feb 2010)
+ | 7 lines Remove color code sequences from verbose messages that
+ go to logfiles. (closes issue #16786) Reported by: dodo Patches:
+ logger2.patch uploaded by dodo (license 989) Tested by: tilghman
+ ........
+
+2010-02-24 06:39 +0000 [r248533-248534] Russell Bryant <russell@digium.com>
+
+ * funcs/func_strings.c: Remove unnecessary warning message, make a
+ couple of formatting tweaks
+
+ * tests/test_strings.c: Add ASTERISK_FILE_VERSION macro.
+
+2010-02-23 22:29 +0000 [r248489] Mark Michelson <mmichelson@digium.com>
+
+ * tests/test_strings.c (added): Unit test for ast_str API. Review:
+ https://reviewboard.asterisk.org/r/517
+
+2010-02-23 16:34 +0000 [r248397] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 248396 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010)
+ | 9 lines fixes invite with replaces deadlock (closes issue
+ #16862) Reported by: pwalker Patches: replaces_deadlock_1.4
+ uploaded by dvossel (license 671) Tested by: pwalker, dvossel
+ ........
+
+2010-02-22 20:19 +0000 [r248347] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Move the REF_DEBUG comment higher in the
+ include list. Uncommenting the REF_DEBUG definition where it was
+ in the source resulted in only a small part of the astobj2
+ references being logged to a file. Moving this up higher in the
+ include list causes all references to be logged as they should
+ be.
+
+2010-02-22 06:45 +0000 [r248225-248226] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/taskprocessor.h, main/taskprocessor.c: Minor
+ tweaks to comment blocks and includes. Fix the copyright lines,
+ tweak doxygen formatting, and remove some unnecessary includes.
+
+ * tests/test_devicestate.c: Tweak copyright and author lines.
+
+2010-02-21 12:09 +0000 [r248184] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: Cleanup transmit_* functions, part 1
+ Break transmit_tone into transmit_start_tone and
+ transmit_stop_tone as per the skinny protocol. (closes issue
+ #16874) Reported by: wedhorn Patches: skinny-clean01.diff
+ uploaded by wedhorn (license 30)
+
+2010-02-20 22:37 +0000 [r248108] Olle Johansson <oej@edvina.net>
+
+ * res/res_rtp_asterisk.c: Improve support for RTCP reports without
+ report blocks
+
+2010-02-19 18:38 +0000 [r248003] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c: mfcr2 issue 0016844 - Fix portability bit
+ fields and make mfcr2_immediate_accept work again, reported and
+ patched by korihor
+
+2010-02-19 17:40 +0000 [r247915] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: handle_request_invite revise comment, fix
+ coding guideline issues I'm working with this code right now
+ trying to analyze a deadlock. This change is just to clean up a
+ few things before I make a more complex patch.
+
+2010-02-19 17:33 +0000 [r247914] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 247910 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600
+ (Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
+ .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri,
+ 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing
+ consistent with other channel technologies. The processing of
+ DTMF tones on the receiving side of an ISDN channel is
+ inconsistent with the way it is handled in other channels,
+ especially DAHDI analog. This causes DTMF tones sent from an ISDN
+ phone to be doubled at the connected party. We are using the
+ following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes
+ Option one is necessary because the asterisk DSP DTMF detection
+ is better than mISDN's internal DSP. Not as many false positives.
+ Option two is necessary to transmit DTMF tones end to end when
+ mISDN channels are connected to SIP channels with out of band
+ DTMF for example. The symptom is that DTMF tones sent by an ISDN
+ phone are doubled on the way through asterisk when two mISDN
+ channels are connected with a Local channel in between or if it
+ is bridged to an analog channel. The doubling of DTMF tones is
+ because DTMF is passed inband to asterisk by the mISDN channel
+ and passed out of band once again after the release of the DTMF
+ tone. Passing it inband is wrong. Neither an analog channel nor
+ SIP channel passes DTMF inband if configured to inband DTMF.
+ Analog and SIP channels filter out the DTMF tones because they
+ use the voice frames returned by ast_dsp_process. But chan_misdn
+ passes the unfiltered input voice frames instead. To overcome one
+ aspect of the problem, the doubling of DTMF tones when two mISDN
+ channels are directly bridged, someone made an 'optimization',
+ where in that case the DTMF tone passed out-of-band to the peer
+ channel is not translated to an inband tone at the transmit side.
+ This optimization is bad because it does not work in general. For
+ example, analog channels or mISDN channels when bridged through
+ an intermediary local channel will generate DTMF tones from
+ out-of-band information. Also, of course, it must not be done
+ when there is no inband DTMF available. This patch fixes the
+ issue. Now chan_misdn will filter the received inband DTMF signal
+ the same as other channel types. Another change included: No need
+ to build an extra translation path because ast_process_dsp does
+ it if required. Patches: misdn-dtmf.patch JIRA ABE-2080
+ ................
+
+2010-02-18 23:13 +0000 [r247787-247841] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_speech.c: Revert an errant part of a previous cleanup, to
+ fix a memory corruption issue. (closes issue #16368) Reported by:
+ thirionjwf Patches: res_speech.c.patch uploaded by thirionjwf
+ (license 955)
+
+ * channels/chan_sip.c: If the peer record is from realtime, it
+ could be set to 0, due to MySQL not representing NULL well in
+ integer columns. NULL means the value is not specified for the
+ column, which normally means the driver uses whatever is the
+ default value. However, on MySQL, placing a NULL in either a
+ float or integer column results in a retrieval of the 0 value.
+ Hence, users get an errant error on load. This patch suppresses
+ that error and makes the value as if it was not there. Note that
+ this cannot be done in the realtime driver, because the lack of
+ difference between NULL and 0 can only be intepreted correctly by
+ the driver itself. If we did it in the realtime driver, then it
+ would be effectively impossible to set any realtime field to 0,
+ because it would act as if the field were unspecified and
+ possibly take on a different value. (closes issue #16683)
+ Reported by: wdoekes
+
+2010-02-18 21:23 +0000 [r247736-247770] David Vossel <dvossel@digium.com>
+
+ * bridges/bridge_softmix.c: fixes confbridge crash when no timing
+ module is loaded. (closes issue #16471) Reported by: kjotte
+ Patches: M16471.diff uploaded by junky (license 177) Tested by:
+ kjotte, junky
+
+ * apps/app_queue.c: fixes Queue with C option crash (closes issue
+ #16475) Reported by: okrief Patches: queue_crash.diff uploaded by
+ dvossel (license 671)
+
+2010-02-18 19:39 +0000 [r247652] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/features.c: Merged revisions 247651 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r247651 | mnicholson | 2010-02-18 13:38:09 -0600 (Thu, 18 Feb
+ 2010) | 6 lines Copy the calling party's account code to the
+ called party if they don't already have one. (closes issue
+ #16331) Reported by: bluefox Tested by: mnicholson ........
+
+2010-02-18 18:31 +0000 [r247609] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Fix placing ISDN calls on hold preventing native
+ bridging from being reexamined after a transfer. Consider the
+ following scenario: /-- B A == * == Network \-- C Party B calls
+ party A (EuroISDN BRI phone) Party A puts B on hold using the
+ HOLD/RETRIEVE messages. Party A calls party C. Party A puts C on
+ hold to talk with party B again. Party A transfers B to C by
+ hanging up. The call does not get the opportunity to get
+ re-transferred into the ISDN network by the native bridge because
+ native bridging is not being reexamined after the initial
+ transfer.
+
+2010-02-18 16:54 +0000 [r247503-247509] Leif Madsen <lmadsen@digium.com>
+
+ * /, README-SERIOUSLY.bestpractices.txt: Merged revisions 247508
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r247508 | lmadsen | 2010-02-18 11:53:44 -0500 (Thu, 18 Feb 2010)
+ | 1 line Add additional link to best practices document per
+ jsmith. ........
+
+ * /, README-SERIOUSLY.bestpractices.txt (added): Merged revisions
+ 247502 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r247502 | lmadsen | 2010-02-18 11:38:17 -0500 (Thu, 18 Feb 2010)
+ | 10 lines Add best practices documentation. (issue #16808)
+ Reported by: lmadsen (issue #16810) Reported by: Nick_Lewis
+ Tested by: lmadsen Review:
+ https://reviewboard.asterisk.org/r/507/ ........
+
+2010-02-18 16:34 +0000 [r247500] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * CHANGES, res/res_jabber.c: Add a new manager event for our
+ buddies status. The new JabberStatus event gives a concise view
+ of the status change to the AMI clients. Thanks fiddur! (closes
+ issue #16760) Reported by: fiddur Patches: 244498.2.diff uploaded
+ by fiddur (license 678) Tested by: fiddur, phsultan
+
+2010-02-18 04:20 +0000 [r247423] Russell Bryant <russell@digium.com>
+
+ * Makefile, /, sounds/Makefile: Merged revisions 247422 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r247422 | russell | 2010-02-17 22:19:01 -0600 (Wed, 17 Feb 2010)
+ | 10 lines Tweak argument handling for wget in the sounds
+ Makefile. 1) Fix the check to see if we are using wget to not be
+ full of fail. The configure script populates this variable with
+ the absolute path to wget if it is found, so it didn't work. 2)
+ Allow some extra arguments to be passed in for wget. This is just
+ a simple change to allow our Bamboo build script to tell wget to
+ be quiet and not fill up our logs with download status output.
+ ........
+
+2010-02-17 22:44 +0000 [r247335-247381] Mark Michelson <mmichelson@digium.com>
+
+ * main/test.c: Fix a couple of bugs in test tab completion. 1. Add
+ missing unlock of lists. 2. Swap order of arguments to
+ test_cat_cmp in complete_test_name.
+
+ * main/test.c: Tab completion for test categories and names for
+ "test show registered" and "test execute" CLI commands.
+
+ * main/strings.c, include/asterisk/strings.h: Fix two problems in
+ ast_str functions found while writing a unit test. 1. The
+ documentation for ast_str_set and ast_str_append state that the
+ max_len parameter may be -1 in order to limit the size of the
+ ast_str to its current allocated size. The problem was that the
+ max_len parameter in all cases was a size_t, which is unsigned.
+ Thus a -1 was interpreted as UINT_MAX instead of -1. Changing the
+ max_len parameter to be ssize_t fixed this issue. 2. Once issue 1
+ was fixed, there was an off-by-one error in the case where we
+ attempted to write a string larger than the current allotted size
+ to a string when -1 was passed as the max_len parameter. When
+ trying to write more than the allotted size, the ast_str's
+ __AST_STR_USED was set to 1 higher than it should have been.
+ Thanks to Tilghman for quickly spotting the offending line of
+ code. Oh, and the unit test that I referenced in the top line of
+ this commit will be added to reviewboard shortly. Sit tight...
+
+2010-02-17 19:51 +0000 [r247295] Jeff Peeler <jpeeler@digium.com>
+
+ * funcs/func_groupcount.c, tests/test_app.c (added), main/app.c,
+ CHANGES: Add support for GROUP_MATCH_COUNT regex matching on
+ category Current support for regex matching was previously only
+ available on the group. Also, error reporting for regex failures
+ has been added. In addition to this feature enhancement a unit
+ test has been written to check the regular expression logic to
+ ensure the count operation is working as expected. (closes issue
+ #16642) Reported by: kobaz Patches: groupmatch2.patch uploaded by
+ kobaz (license 834) Review:
+ https://reviewboard.asterisk.org/r/503/
+
+2010-02-17 19:23 +0000 [r247248-247282] David Vossel <dvossel@digium.com>
+
+ * tests/test_devicestate.c: modified device2extension_test's
+ category
+
+ * tests/test_devicestate.c (added): unit test for combined device
+ state mapping and device to exten state mapping Review:
+ https://reviewboard.asterisk.org/r/516/
+
+ * main/features.c, CHANGES, configs/features.conf.sample: addition
+ of dynamic parkinglots feature This feature allows for
+ parkinglots to be created dynamically within the dialplan. Thanks
+ to all who were involved with getting this patch written and
+ tested! (closes issue #15135) Reported by: IgorG Patches:
+ features.dynamic_park.v3.diff uploaded by IgorG (license 20)
+ 2009090400_dynamicpark.diff.txt uploaded by mvanbaak (license 7)
+ dynamic_parkinglot.diff uploaded by dvossel (license 671) Tested
+ by: eliel, IgorG, acunningham, mvanbaak, zktech Review:
+ https://reviewboard.asterisk.org/r/352/
+
+2010-02-17 16:24 +0000 [r247169] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 247168 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r247168 | mmichelson | 2010-02-17 10:24:17 -0600 (Wed, 17 Feb
+ 2010) | 3 lines Make sure that when autofill is disabled that
+ callers not in the front of the queue cannot place calls.
+ ........
+
+2010-02-17 07:01 +0000 [r247124-247125] Tilghman Lesher <tlesher@digium.com>
+
+ * main/loader.c: RTP documentation states that you can pass NULL as
+ the module, so make sure that's really the case.
+
+ * channels/sip/include/dialog.h (added), channels/chan_sip.c,
+ channels/sip/include/config_parser.h,
+ channels/sip/include/globals.h (added),
+ channels/sip/dialplan_functions.c (added), channels/Makefile,
+ channels/sip/include/sip_utils.h,
+ channels/sip/include/dialplan_functions.h (added): Make all of
+ the various rtpqos parameters in this branch available from the
+ CHANNEL function. Also includes a test for retrieving rtpqos
+ parameters, including a NULL RTP driver. Additionally, some
+ further separation of the SIP internal API into headers was
+ necessary. (closes issue #16652) Reported by: kkm Patches:
+ 20100204__issue16652.diff.txt uploaded by tilghman (license 14)
+ Review: https://reviewboard.asterisk.org/r/501/
+
+2010-02-16 23:44 +0000 [r247076] Mark Michelson <mmichelson@digium.com>
+
+ * main/strings.c: Add va_end calls to __ast_str_helper. According
+ to the man page for stdarg(3), "Each invocation of va_copy() must
+ be matched by a corresponding invocation of va_end() in the same
+ function." There were several cases in __ast_str_helper where
+ va_copy was not matched with a corresponding call to va_end.
+
+2010-02-16 22:58 +0000 [r247035] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c: generate
+ connected line info update from info in h.323 packets Tested by:
+ benngard
+
+2010-02-16 21:15 +0000 [r246985] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/strings.h: Add some clarifying documentation to
+ the ast_str_set and ast_str_append functions.
+
+2010-02-16 21:03 +0000 [r246980-246981] David Vossel <dvossel@digium.com>
+
+ * main/tcptls.c: swap openssl with OpenSSL in warning message.
+ (issue #16673)
+
+ * main/tcptls.c: warning message if openssl support is missing
+ while attempting tls connection (closes issue #16673) Reported
+ by: michaesc Patches: tls_error_msg.diff uploaded by dvossel
+ (license 671)
+
+2010-02-16 18:29 +0000 [r246942] Mark Michelson <mmichelson@digium.com>
+
+ * tests/test_pbx.c (added): Add unit test for dialplan pattern
+ matching. This test works by reading input from arrays to build a
+ sample dialplan. From there, patterns are attempted to be matched
+ against said dialplan, with the expected match given. We then
+ search in our example dialplan to see if we find a match and if
+ what we find matches what we expected it to match. (closes issue
+ #16809) Reported by: lmadsen Tested by: mmichelson Review:
+ https://reviewboard.asterisk.org/r/504/
+
+2010-02-16 17:07 +0000 [r246899] David Vossel <dvossel@digium.com>
+
+ * main/channel.c: fixes sample rate conversion issue with Monitor
+ application When using ast_seekstream with the read/write streams
+ of a monitor, the number of samples we are seeking must be of the
+ same rate as the stream or the jump calculation will be
+ incorrect. This patch adds logic to correctly convert the number
+ of samples to jump to the sample rate the read/write stream is
+ using. For example, if the call is G722 (16khz) and the
+ read/write stream is recording a 8khz wav, seeking 320 samples of
+ 16khz audio is not the same as seeking 320 samples of 8khz audio
+ when performing the ast_seekstream on the stream. ABE-2044
+
+2010-02-16 15:36 +0000 [r246710-246863] Tilghman Lesher <tlesher@digium.com>
+
+ * build_tools/cflags.xml, build_tools/cflags-devmode.xml: Revert
+ changes for now, pending discussion
+
+ * build_tools/cflags-devmode.xml: Add a few more targets for
+ DEBUG_THREADLOCALS
+
+ * build_tools/cflags.xml, channels/chan_usbradio.c,
+ build_tools/cflags-devmode.xml, main/strings.c,
+ apps/app_voicemail.c: Change the blanket rules to delete
+ .lastclean on all CFLAGS menuselect targets to be more
+ particular. This change builds upon the recent change to
+ menuselect to add 'touch_on_change' as an attribute of both
+ categories and members. This should allow only the most invasive
+ defines to cause a complete rebuild, while defines which only
+ affect a subset of modules will only cause a rebuild of that
+ smaller set.
+
+ * channels/chan_sip.c: Allow Timer B to be set on the peer, and
+ ensure SIP rules are followed (or warn) in comparison to Timer
+ T1. (closes issue #16643) Reported by: nahuelgreco Patches:
+ 20100204__issue16643.diff.txt uploaded by tilghman (license 14)
+ Tested by: oej
+
+ * Makefile, /: Merged revisions 246709 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r246709 | tilghman | 2010-02-15 17:42:33 -0600 (Mon, 15 Feb 2010)
+ | 5 lines Make the menuselect instructions correct by allowing
+ 'make menuselect' to actually solve dependency problems.
+ (Previously, it would fail out again with the same message about
+ running 'make menuselect', which was NOT at all helpful.)
+ ........
+
+2010-02-15 22:08 +0000 [r246669] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Restore triedtopribridge flag code removed
+ in -r211197. Ooops. Failed to note that we were inside a for loop
+ and pri_channel_bridge() needs to be executed only once.
+
+2010-02-15 21:37 +0000 [r246667] Tilghman Lesher <tlesher@digium.com>
+
+ * utils/utils.xml: Instead of just automatically filtering out in
+ the Makefile, give an indication of dependencies in menuselect.
+
+2010-02-15 15:45 +0000 [r246627] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c, channels/sip/reqresp_parser.c,
+ channels/sip/include/sip_utils.h,
+ channels/sip/include/reqresp_parser.h: chan_sip parse code
+ refactoring plus two new unit tests Code Refactoring Changes -
+ read_to_parts() moved to reqresp_parser.c and has been renamed as
+ get_name_and_number() - get_in_brackets() moved to
+ reqresp_parser.c - find_closing_quotes() added to sip_utils.h
+ Logic Changes - get_name_and_number() now uses parse_uri() and
+ get_calleridname() for parsing. Before this change only names
+ within quotes were found, when names not within quotes are
+ possible. New Unit Tests -sip_get_name_and_number_test
+ -sip_get_in_brackets_test (closes issue #16707) Reported by:
+ Nick_Lewis Patches: issue16706.diff uploaded by dvossel (license
+ 671) Review: https://reviewboard.asterisk.org/r/499/
+
+2010-02-12 23:32 +0000 [r246420-246546] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /: Merged revisions 246545 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r246545 | dvossel | 2010-02-12 17:30:17 -0600 (Fri, 12 Feb 2010)
+ | 16 lines lock channel during datastore removal On channel
+ destruction the channel's datastores are removed and destroyed.
+ Since there are public API calls to find and remove datastores on
+ a channel, a lock should be held whenever datastores are removed
+ and destroyed. This resolves a crash caused by a race condition
+ in app_chanspy.c. (closes issue #16678) Reported by:
+ tim_ringenbach Patches: datastore_destroy_race.diff uploaded by
+ tim ringenbach (license 540) Tested by: dvossel ........
+
+ * channels/chan_sip.c: fixes areas where port should be removed
+ from domain during parsing A patch was committed recently that
+ converted duplicate uri parsing code to use the parse_uri
+ function. There were two instances where this conversion did not
+ mimic previous behavior exactly because the port was not being
+ parsed off the end of the domain. In order to do this, a dummy
+ pointer argument needs to be passed into parse_uri so it will
+ know it must parse out the port from the domain. If a port output
+ paramenter is not present, the domain is returned with the port
+ still attached.
+
+2010-02-12 08:30 +0000 [r246382] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c, UPGRADE.txt, CHANGES: Updated doc for OSP
+ lookup application.
+
+2010-02-11 21:57 +0000 [r246299-246338] David Vossel <dvossel@digium.com>
+
+ * tests/test_heap.c, tests/test_event.c,
+ channels/sip/reqresp_parser.c, channels/sip/config_parser.c:
+ fixes some test description formatting inconsistencies so log
+ file looks nice
+
+ * tests/test_astobj2.c (added), main/astobj2.c: astobj2 unit test
+ and bug fix A bug was discovered during the creation of the
+ astobj2 unit test. When OBJ_MULTIPLE | OBJ_UNLINK is used, the
+ objects being returned had a ref count issue. This patch resolves
+ that. Review: https://reviewboard.asterisk.org/r/496/
+
+2010-02-10 23:19 +0000 [r246260] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/event.h, tests/test_event.c (added),
+ main/event.c: Add a test module for the event API, test_event.c.
+ This module includes a single test so far that creates events
+ using two different methods and does some verification on the
+ result to make sure the correct data can be retrieved from the
+ event that was created. One bug was found in the event API while
+ developing this test, which makes me happy. :-) Review:
+ https://reviewboard.asterisk.org/r/495/
+
+2010-02-10 23:13 +0000 [r246249] David Vossel <dvossel@digium.com>
+
+ * channels/sip/reqresp_parser.c,
+ channels/sip/include/reqresp_parser.h: additional parse_uri test
+ and documentation
+
+2010-02-10 21:55 +0000 [r246200-246208] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_pktccops.exports (added): res_pktccops needs to be able
+ to export a symbol for chan_mgcp (closes issue #16782) Reported
+ by: nahuelgreco Patches: res_pktccops.exports uploaded by
+ nahuelgreco (license 162)
+
+ * funcs/func_strings.c: Fussy compiler on another machine...
+
+ * funcs/func_strings.c: Fix weird issue with unit tests on
+ optimized build - turned out to be a signing issue.
+
+2010-02-10 17:49 +0000 [r246116] David Vossel <dvossel@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 246115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r246115 | dvossel | 2010-02-10 11:44:20 -0600 (Wed, 10 Feb 2010)
+ | 8 lines fixes random deadlock in app_queue with use_weight
+ during reload (closes issue #16677) Reported by: tim_ringenbach
+ Patches: app_queue_use_weight_deadlock.diff uploaded by tim
+ ringenbach (license 540) ........
+
+2010-02-10 16:47 +0000 [r246070] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_local.c: Change channel state on local channels for
+ busy,answer,ring. Previously local channels channel state never
+ changed. This became problematic when the state of the other side
+ of the local channel was lost, for example during a masquerade.
+ Changing the state of the local channel allows for the scenario
+ to be detected when the channel state is set to ringing, but the
+ peer isn't ringing. The specific problem scenario is described in
+ 164201. Although this was noted on one of the issues, here is the
+ tested dialplan verified to work: exten =>
+ 9700,1,Dial(Local/*9700@default&Local/0009700@default) exten =>
+ *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1)
+ exten => *9700,n,wait(3) ;3 works, 1 did not exten =>
+ *9700,n,Dial(SIP/5001) exten => 0009700,1,Wait(1) ;1 works, 3 did
+ not exten =>
+ 0009700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes
+ issue #14992) Reported by: davidw
+
+2010-02-10 16:01 +0000 [r245945-246030] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ res/res_agi.c: Solaris doesn't like outputting a NULL to a %s in
+ format strings. Detect all platforms that don't like that,
+ either, and ensure that when documentation is missing, we pass a
+ non-NULL pointer when outputting the corresponding documentation.
+ (closes issue #16689) Reported by: bklang Patches:
+ 20100209__issue16689__with_tests.diff.txt uploaded by tilghman
+ (license 14) Review: https://reviewboard.asterisk.org/r/497/
+
+ * funcs/func_strings.c: Enable warnings on atypical conditions for
+ the FILTER function (suggested by mmichelson on the -dev list).
+
+ * /, funcs/func_strings.c, configs/extensions.conf.sample: Merged
+ revisions 245944 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r245944 | tilghman | 2010-02-10 07:37:13 -0600 (Wed, 10 Feb 2010)
+ | 2 lines Include examples of FILTER usage in extension patterns
+ where a "." may be a risk. ........
+
+2010-02-09 23:32 +0000 [r245864] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/test.h, tests/test_sha1.c (removed),
+ include/asterisk/utils.h, tests/test_substitution.c,
+ tests/test_heap.c, tests/test_ast_format_str_reduce.c,
+ tests/test_skel.c, tests/test_utils.c, funcs/func_math.c,
+ channels/sip/reqresp_parser.c, main/test.c, tests/test_md5.c
+ (removed), channels/sip/config_parser.c, tests/test_sched.c:
+ Various updates to the unit test API. 1) It occurred to me that
+ the difference in usage between the error ast_str and the
+ ast_test_update_status() usage has turned out to be a bit
+ ambiguous in practice. In a lot of cases, the same message was
+ being sent to both. In other cases, it was only sent to one or
+ the other. My opinion now is that in every case, I think it makes
+ sense to do both; we should output it to the CLI as well as save
+ it off for logging purposes. This change results in most of the
+ changes in this diff, since it required changes to all existing
+ unit tests. It also allowed for some simplifications of unit test
+ API implementation code. 2) Update ast_test_status_update() to
+ include the file, function, and line number for the code
+ providing the update. 3) There are some formatting tweaks here
+ and there. Hopefully they aren't too distracting for code review
+ purposes. Reviewboard's diff viewer seems to do a pretty good job
+ of pointing out when something is a whitespace change. 4) I moved
+ the md5_test and sha1_test into the test_utils module. It seemed
+ like a better approach since these tests are so tiny. 5) I
+ changed the number of nodes used in heap_test_2 from 1 million to
+ 100 thousand. The only reason for this was to reduce the time it
+ took for this test to run. 6) Remove an unused function prototype
+ that was at the bottom of utils.h. 7) Simplify test_insert()
+ using the LIST_INSERT_SORTALPHA() macro. The one minor difference
+ in behavior is that it no longer checks for a test registered
+ with the same name. 8) Expand the code in test_alloc() to provide
+ specific error messages for each failure case, to clearly inform
+ developers if they forget to set the name, summary, description,
+ etc. 9) Tweak the output of the "test show registered" CLI
+ command. I swapped the name and category to have the category
+ first. It seemed more natural since that is the sort key. 10)
+ Don't output the status ast_str in the "test show results" CLI
+ command. This is going to tend to be pretty verbose, so just
+ leave that for the detailed test logs (test generate results).
+ Review: https://reviewboard.asterisk.org/r/493/
+
+2010-02-09 23:18 +0000 [r245793-245804] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: fixes a merging error for the iaxs and
+ iaxsl off by one fix
+
+ * /, channels/chan_iax2.c: Merged revisions 245792 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r245792 | dvossel | 2010-02-09 16:55:38 -0600 (Tue, 09
+ Feb 2010) | 12 lines Fixes iaxs and iaxsl size off by one issue.
+ 2^15 = 32768 which is the maximum allowed iax2 callnumber.
+ Creating the iaxs and iaxsl array of size 32768 means the maximum
+ callnumber is actually out of bounds. This causes a nasty crash.
+ (closes issue #15997) Reported by: exarv Patches: iax_fix.diff
+ uploaded by dvossel (license 671) ........
+
+2010-02-09 18:06 +0000 [r245729] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_fax.c: Ensure frames are only freed once. (closes issue
+ #16361) Reported by: vlad Patches: 20100208__issue16361.diff.txt
+ uploaded by tilghman (license 14) Tested by: kenny, bloodoff,
+ misaksen
+
+2010-02-09 17:40 +0000 [r245727] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: This commit removes an extra newline in T.38
+ generated SDP packets. This bug was caused by the fix introduced
+ in r243860. (closes issue #16766) Reported by: raivisr Patches:
+ t38-sdp-newline-fix1.diff uploaded by mnicholson (license 96)
+ Tested by: raivisr
+
+2010-02-09 16:24 +0000 [r245680] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_fax.c: Don't offer MMR or JBIG transcoding during T.38
+ negotiation. After further discussion with Steve Underwood, we
+ should not (yet) be offering to receive MMR or JBIG transcoded
+ streams from T.38 endpoints. A future spandsp release will
+ support those features, and then they can be enabled during
+ negotiation
+
+2010-02-08 23:43 +0000 [r245597-245624] Russell Bryant <russell@digium.com>
+
+ * main/event.c: Fix return value of get_ie_str() and
+ get_ie_str_hash() for non-existent IE. I found this bug while
+ developing a unit test for event allocation. Testing is awesome.
+
+ * tests/test_utils.c: UNREGISTER instead of REGISTER in
+ unload_module().
+
+ * main/pbx.c: Use memmove() instead of memcpy() for a case where
+ the buffers overlap. Once again, valgrind is freaking awesome.
+ That is all.
+
+ * channels/Makefile: Remove object files from the channels/sip/
+ directory on make clean.
+
+2010-02-08 22:31 +0000 [r245578] Tilghman Lesher <tlesher@digium.com>
+
+ * main/Makefile, channels/Makefile: Actually use _ASTLDFLAGS in the
+ main/ and channels/ Makefiles. They were previously passed
+ correctly, but they simply weren't used. This caused issues with
+ various platforms whose builds needed to pass special linker
+ flags via the configure script. (closes issue #16596) Reported
+ by: pprindeville Patches: asterisk-1.6-astldflags.patch uploaded
+ by pprindeville (license 347) Tested by: tilghman
+
+2010-02-08 20:41 +0000 [r245497] Jason Parker <jparker@digium.com>
+
+ * /, main/ast_expr2f.c, main/ast_expr2.fl: Merged revisions 245496
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r245496 | qwell | 2010-02-08 14:39:50 -0600 (Mon, 08 Feb 2010) |
+ 4 lines Remove reference of documentation in source directory.
+ People don't always build Asterisk from source (distro packages,
+ anybody?). ........
+
+2010-02-08 04:51 +0000 [r245268-245385] Russell Bryant <russell@digium.com>
+
+ * contrib/scripts/install_prereq: Add the libvpb-dev package as a
+ dependency.
+
+ * pbx/pbx_gtkconsole.c: Add a todo for pbx_gtkconsole for updating
+ to gtk2. This module needs to be converted to gtk2, or we will
+ eventually have to just remove it from the tree. gtk1 isn't even
+ packaged anymore in the distro I'm using. I suspect nobody uses
+ this and that nobody would notice if we removed it.
+
+ * contrib/scripts/install_prereq: Add more packages required for
+ building Asterisk modules.
+
+ * channels/chan_usbradio.c: Make chan_usbradio compile.
+
+ * tests/test_sha1.c (added): Add a SHA1 test module. Review:
+ https://reviewboard.asterisk.org/r/492/
+
+ * tests/test_md5.c: Remove unnecessary include, ast_md5_hash()
+ comes from utils.h.
+
+ * tests/test_md5.c (added): Add an MD5 test module. Review:
+ https://reviewboard.asterisk.org/r/491/
+
+ * tests/test_ast_format_str_reduce.c: Fix a couple of spelling
+ errors, and add format module dependencies.
+
+ * channels/sip/include/config_parser.h, channels/sip/include/sip.h,
+ channels/sip/include/sip_utils.h,
+ channels/sip/include/reqresp_parser.h: Tweak formatting and add
+ minor updates to some comments.
+
+ * main/test.c: Remove an extra space.
+
+2010-02-07 19:51 +0000 [r245230] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Remove parsing of constantssrc from
+ reload_config. This config option is already handled by the
+ function handle_common_options and it is unnecessary to parse the
+ value again.
+
+2010-02-06 14:43 +0000 [r245192] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: Remove useless sip
+ options related to hash table size. First off, these options
+ weren't actually doing anything. By the time the options were
+ parsed, the peer and dialog containers had already been allocated
+ with their default values. Second, hash table size is something
+ that doesn't really make sense to change in a config file. If a
+ user is that interested in changing the hashtable size, he can
+ modify the source itself. I have removed the parsing of the
+ hash_peer, hash_user, and hash_dialog options. I have removed the
+ hash_user_size variable altogether since it is not used at all. I
+ also changed hash_peer_size and hash_dialog_size to be constant,
+ and have changed the symbols to be in all caps as constants
+ typically are. I have also removed the entire section in
+ sip.conf.sample regarding configurable hashtable sizes.
+
+2010-02-05 21:21 +0000 [r245147] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/astobj2.h, main/astobj2.c: fixes astobj2
+ unlinking of multiple objects when OBJ_MULTIPLE was disabled When
+ OBJ_MULTIPLE was off but OBJ_UNLINK was on, all the items in a
+ bucket were being unlinked instead of just the first match. This
+ fixes that. Review: https://reviewboard.asterisk.org/r/490/
+
+2010-02-05 19:26 +0000 [r245090] Jeff Peeler <jpeeler@digium.com>
+
+ * /, LICENSE, contrib/firmware (removed): Merged revisions 245044
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r245044 | kpfleming | 2010-02-05 12:32:29 -0600 (Fri, 05 Feb
+ 2010) | 5 lines Remove contrib/firmware directory as it is empty
+ Remove explicit license for IAXy firmware as it is no longer
+ included in the tree ........
+
+2010-02-05 19:07 +0000 [r245046] Tilghman Lesher <tlesher@digium.com>
+
+ * tests/test_ast_format_str_reduce.c, main/file.c: Merge tests that
+ verify the same thing. (Oops.)
+
+2010-02-05 18:12 +0000 [r245006] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: adds total call numbers available to 'iax2
+ show callnumber usage' cli output
+
+2010-02-05 17:20 +0000 [r244945] Terry Wilson <twilson@digium.com>
+
+ * res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
+ res/res_calendar_caldav.c: Fix crash on 32-bit for users not
+ using https (closes issue #16778) Reported by: pitel Patches:
+ diff.txt uploaded by twilson (license 396) Tested by: twilson,
+ pitel
+
+2010-02-05 17:05 +0000 [r244927] Sean Bright <sean@malleable.com>
+
+ * /, main/asterisk.c: Merged revisions 244926 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r244926 | seanbright | 2010-02-05 12:03:35 -0500 (Fri, 05 Feb
+ 2010) | 1 line Update main copyright date. ........
+
+2010-02-05 16:59 +0000 [r244769-244924] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c, channels/sip/include/config_parser.h,
+ channels/sip/config_parser.c: fixes issue with sip registry not
+ having correct default expiry default expiry was not being set
+ correctly for a registry object. Thanks to ebroad for reporting
+ the issue and testing the patch.
+
+ * main/astobj2.c: fixes memory leak in astobj2 test
+ ao2_iterator_destroy was not being used on the iterator during
+ the test. This resulted in the container never actually being
+ destroyed.
+
+ * channels/chan_sip.c: parse_moved_contact tries to parse
+ contact_name twice parse_moved_contact attempts to remove a
+ quoted string twice, and the first try wasn't even being done
+ correctly.
+
+2010-02-04 22:43 +0000 [r244728-244768] Tilghman Lesher <tlesher@digium.com>
+
+ * main/file.c: Try to make ast_format_str_reduce fail...
+
+ * include/asterisk/manager.h: Oops
+
+ * include/asterisk/manager.h: Define a small set of constant return
+ values
+
+2010-02-04 15:36 +0000 [r244688] David Vossel <dvossel@digium.com>
+
+ * main/test.c: fix truncated format string in 'test show
+ registered' When using the 'test show registered' cli command the
+ 'Test Results' category was truncating the last few characters
+ making it look like 'Test Resul'. I also expanded other parts of
+ the format to better represent how long function names and
+ categories will likely be.
+
+2010-02-04 00:12 +0000 [r244647] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sip: Add ignore *.i files property to the new
+ channels/sip directory.
+
+2010-02-03 20:48 +0000 [r244598] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c, CHANGES: Add some additional option support for
+ non-default parking lots. The options are: parkedcallparking,
+ parkedcallhangup, parkedcallrecording, and parkedcalltransfers.
+ Previously these options were only available for the default
+ parking lot. (closes issue #16641) Reported by: bluecrow76
+ Patches: asterisk-1.6.2.1-features.c.diff uploaded by bluecrow76
+ (license 270)
+
+2010-02-03 20:33 +0000 [r244597] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c, channels/sip/include/config_parser.h
+ (added), channels/sip/reqresp_parser.c (added), channels/sip
+ (added), channels/Makefile, channels/sip/config_parser.c (added),
+ channels/sip/include (added), channels/sip/include/sip.h (added),
+ channels/sip/include/sip_utils.h (added),
+ channels/sip/include/reqresp_parser.h (added): -----Changes -----
+ New files - channels/sip/sip.h – A new header for shared #define,
+ enum, and struct definitions. - channels/sip/include/sip_utils.h
+ – sip util functions shared among the all the sip APIs -
+ channels/sip/include/config_parser.h – sip config-parser API -
+ channels/sip/config_parser.c – Contains sip.conf parsing helper
+ functions with unit tests. -
+ channels/sip/include/reqresp_parser.h – sip request response
+ parser API - channels/sip/reqresp_parser.c – Contains sip request
+ and response parsing helper functions with unit tests. New Unit
+ Tests - sip_parse_uri_test - sip_parse_host_test -
+ sip_parse_register_line_test Code Refactoring - All reusable
+ #define, enum, and struct definitions were moved out of
+ chan_sip.c into sip.h. During this process formatting changes
+ were made to comments in both sip.h and chan_sip.c in order to
+ better adhere to the coding guidelines. - The beginnings of three
+ new sip APIs, sip-utils.h, config-parser.h, reqresp-parser.h
+ using existing chan_sip.c functions. - parse_uri() and
+ get_calleridname() were moved from chan_sip.c to request-parser.c
+ along with unit tests for both functions. - sip_parse_host() and
+ sip_parse_register_line() were moved from chan_sip.c to
+ config-parser.c along with unit tests for both functions. Changes
+ to parse_uri() -removal of the options parameter. It was never
+ used and did not behave correctly. -additional check for
+ [?header] field. When this field was present, the transport type
+ was not being set correctly. ----- Overview ----- This patch is
+ introduced with the hope that unit tests for all our sip parsing
+ functions will be written soon. chan_sip is a huge file, and with
+ the addition of each unit test chan_sip is going to grow larger
+ and harder to maintain. I'm proposing we begin refactoring
+ chan_sip, starting with the parsing functions. With each parsing
+ function we move into a separate helper file, a unit test should
+ accompany it. I've attempted to lay down the ground work for this
+ change by creating two new parser helper files (config-parser.c
+ and reqresp-parser.c) and moving all shared structs, enums, and
+ defines from chan_sip.c into a shared sip.h file. We can't verify
+ everything in Asterisk using unit tests, but string parsing is
+ one area where unit tests make the most sense. By beginning to
+ restructure the code in this way, chan_sip not only becomes less
+ bloated, but Asterisk as a whole will become more stable. Review:
+ https://reviewboard.asterisk.org/r/477/
+
+2010-02-03 19:26 +0000 [r244547] Mark Michelson <mmichelson@digium.com>
+
+ * main/sched.c: Initialize counters in ast_sched_report so that
+ resulting data is not bogus.
+
+2010-02-03 18:34 +0000 [r244505] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c: The chanvar= setting should inherit the
+ entire list of variables, not just the first one. (closes issue
+ #16359) Reported by: raarts Patches: dahdi-setvars.diff uploaded
+ by raarts (license 937) Tested by: raarts
+
+2010-02-02 22:27 +0000 [r244443] David Vossel <dvossel@digium.com>
+
+ * main/udptl.c, channels/chan_sip.c, include/asterisk/udptl.h:
+ fixes crash during T.38 negotiation caused by invalid or missing
+ FaxMaxDatagram field AST-2010-001 (closes issue #16634) Reported
+ by: krn (closes issue #16724) Reported by: barthpbx (closes issue
+ #16517) Reported by: bklang (closes issue #16485) Reported by:
+ elsto
+
+2010-02-02 20:32 +0000 [r244071-244393] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_dial.c, CHANGES: Properly respect GOSUB_RESULT as to
+ what to do with the master channel. Previously, we would parse
+ GOSUB_RESULT, but not actually do anything with it. Also, allow
+ GOSUB_RETVAL to be inherited back across a peer/master channel.
+ (closes issue #16687) Reported by: bklang Patches:
+ app_dial-preserve-gosub_retval.patch uploaded by bklang (license
+ 919) (with modifications) (closes issue #16686) Reported by:
+ bklang Patches: app_dial-respect-gosub_result.patch uploaded by
+ bklang (license 919) (with modifications)
+
+ * funcs/func_math.c: Correct some off-by-one errors, especially
+ when expressions don't contain expected spaces. Also include the
+ tests provided by the reporter, as regression tests. (closes
+ issue #16667) Reported by: wdoekes Patches:
+ astsvn-func_match-off-by-one.diff uploaded by wdoekes (license
+ 717)
+
+ * /, apps/app_voicemail.c: Merged revisions 244242 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r244242 | tilghman | 2010-02-01 17:13:44 -0600 (Mon, 01
+ Feb 2010) | 11 lines Backup and restore original textfile, for
+ prosthesis (gerund of prepend). Also, fix menuselect such that
+ changing voicemail build options correctly causes rebuild.
+ (closes issue #16415) Reported by: tomo1657 Patches:
+ prepention.patch uploaded by tomo1657 (license 484) (with
+ modifications by me to backport to 1.4) ........
+
+ * main/channel.c, channels/chan_local.c, /: Merged revisions 244070
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r244070 | tilghman | 2010-02-01 11:46:31 -0600 (Mon, 01 Feb 2010)
+ | 16 lines Revert previous chan_local fix (r236981) and fix
+ instead by destroying expired frames in the queue. (closes issue
+ #16525) Reported by: kobaz Patches: 20100126__issue16525.diff.txt
+ uploaded by tilghman (license 14)
+ 20100129__issue16525__1.6.0.diff.txt uploaded by tilghman
+ (license 14) Tested by: kobaz, atis (closes issue #16581)
+ Reported by: ZX81 (closes issue #16681) Reported by: alexr1
+ ........
+
+2010-01-28 22:37 +0000 [r243986] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c: Optimization to manager events. When potentially
+ sending manager events, return immediately if there are no
+ sessions or hooks. Also, avoid locking the hooks list if it is
+ empty. (issue #16455) Reported by: atis Patches:
+ manager_hooks_trunk.patch uploaded by atis (license 242)
+
+2010-01-28 20:00 +0000 [r243943] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/iax2-parser.c: Informational message, not an error.
+
+2010-01-28 18:35 +0000 [r243780-243860] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Add a missing line terminator for T.38 SDP.
+
+ * /, channels/chan_sip.c: Merged revisions 243779 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r243779 | russell | 2010-01-28 09:03:17 -0600 (Thu, 28 Jan 2010)
+ | 2 lines Fix a bogus third argument to ast_copy_string().
+ ........
+
+2010-01-27 20:37 +0000 [r243551-243693] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 243691 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r243691 | jpeeler | 2010-01-27 14:35:56 -0600 (Wed, 27 Jan 2010)
+ | 5 lines Revert 243570, I should have looked at this closer.
+ Will reopen the issue, but am leaving the review closed as the
+ change was pointless. (issue #16488) ........
+
+ * CHANGES: expand code based appreviation of AST_CONFIG_DIR to
+ configuration directory
+
+ * /, apps/app_queue.c: Merged revisions 243570 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r243570 | jpeeler | 2010-01-27 12:47:34 -0600 (Wed, 27 Jan 2010)
+ | 9 lines Extend announcement URL used with Queue from 80 chars
+ to PATH_MAX. (closes issue #16488) Reported by: syspert Patches:
+ soundfilelen.pacth-2 uploaded by syspert (license 938) Review:
+ https://reviewboard.asterisk.org/r/475/ ........
+
+ * Makefile, CHANGES, include/asterisk/options.h, main/asterisk.c,
+ main/loader.c: Add new option to asterisk.conf (lockconfdir) to
+ protect conf dir during reloads (closes issue #16358) Reported
+ by: raarts Patches: lockconfdir.diff uploaded by raarts (license
+ 937) modified by me
+
+2010-01-27 18:08 +0000 [r243487] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 243486 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r243486 | mmichelson | 2010-01-27 12:06:43 -0600 (Wed, 27 Jan
+ 2010) | 3 lines Use a safe list traversal while checking for
+ duplicate vars in pbx_builtin_setvar_helper. ........
+
+2010-01-27 17:32 +0000 [r243482] Russell Bryant <russell@digium.com>
+
+ * funcs/func_channel.c, channels/chan_iax2.c: Fix the ability to
+ specify an OSP token for an outbound IAX2 call. When this patch
+ was originally submitted, the code allowed for the token to be
+ set via a channel variable. I decided that a cleaner approach
+ would be to integrate it into the CHANNEL() function.
+ Unfortunately, that is not a suitable approach. It's not possible
+ to get the value set on the channel soon enough using that
+ method. So, go back to the simple channel variable method.
+ (closes issue #16711) Reported by: homesick Patches: iax-svn.diff
+ uploaded by homesick (license 91)
+
+2010-01-26 23:56 +0000 [r243391] David Vossel <dvossel@digium.com>
+
+ * /, main/features.c: Merged revisions 243390 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r243390 | dvossel | 2010-01-26 17:55:49 -0600 (Tue, 26 Jan 2010)
+ | 9 lines fixes bug with channel receiving wrong privileges after
+ call parking (closes issue #16429) Reported by: Yasuhiro Konishi
+ Patches: features.c.diff uploaded by Yasuhiro Konishi (license
+ 947) Tested by: dvossel ........
+
+2010-01-26 20:49 +0000 [r243346] David Ruggles <thedavidfactor@gmail.com>
+
+ * apps/app_senddtmf.c: Code clean up in app_senddtmf Pushes code
+ clean up done in app_externalivr back into app_senddtmf Review:
+ https://reviewboard.asterisk.org/r/473/
+
+2010-01-26 18:20 +0000 [r243244-243266] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 243258 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r243258 | jpeeler | 2010-01-26 12:19:10 -0600 (Tue, 26 Jan 2010)
+ | 2 lines Remove unnecessary code in ast_read as issue 16058 has
+ been fully solved now. ........
+
+ * main/frame.c: Fix crash resulting from frames with invalid data
+ pointers. In ast_frdup the frame data union does not get set to
+ point to malloced memory if the datalen is zero, so make sure to
+ handle the same case in ast_frisolate appropriately. (closes
+ issue #16058) Reported by: atis Patches: bug16058-fix.patch
+ uploaded by jpeeler (license 325) Tested by: atis
+
+2010-01-26 17:40 +0000 [r243200-243242] David Vossel <dvossel@digium.com>
+
+ * main/test.c: modify 'test show registered' cli output format In
+ order to improve readability, the output from 'test show
+ registered' has been modified to truncate fields to fit within
+ the format output if they are over a certain length.
+
+ * include/asterisk/utils.h, channels/chan_sip.c, tests/test_utils.c
+ (added), main/test.c, main/utils.c: RFC compliant uri and
+ display-name encode/decode 1. URI Encoding This patch changes
+ ast_uri_encode()'s behavior when doreserved is enabled.
+ Previously when doreserved was enabled only a small set of
+ reserved characters were encoded. This set was comprised
+ primarily of the reserved characters defined in RFC3261 section
+ 25.1, but contained other characters as well. Rather than only
+ escaping the reserved set, doreserved now escapes all characters
+ not within the unreserved set as defined by RFC 3261 and RFC
+ 2396. Also, the 'doreserved' variable has been renamed to
+ 'do_special_char' in attempts to avoid confusion. When doreserve
+ is not enabled, the previous logic of only encoding the
+ characters <= 0X1F and > 0X7f remains, except for the '%'
+ character, which must always be encoded as it signifies a HEX
+ escaped character during the decode process. 2. URI Decoding:
+ Break up URI before decode. In chan_sip.c ast_uri_decode is
+ called on the entire URI instead of it's individual parts after
+ it is parsed. This is not good as ast_uri_decode can introduce
+ special characters back into the URI which can mess up parsing.
+ This patch resolves this by not decoding a URI until parsing is
+ completely done. There are many instances where we check to see
+ if pedantic checking is enabled before we decode a URI. In these
+ cases a new macro, SIP_PEDANTIC_DECODE, is used on the individual
+ parsed segments of the URI rather than constantly putting if
+ (pedantic) { decode() } checks everywhere in the code. In the
+ areas where ast_uri_decode is not dependent upon pedantic
+ checking this macro is not used, but decoding is still moved to
+ each individual part of the URI. The only behavior that should
+ change from this patch is the time at which decoding occurs.
+ Since I had to look over every place URI parsing occurs to create
+ this patch, I found several places where we use duplicate code
+ for parsing. To consolidate the code, those areas have updated to
+ use the parse_uri() function where possible. 3. SIP display-name
+ decoding according to RFC3261 section 25. To properly decode the
+ display-name portion of a FROM header, chan_sip's
+ get_calleridname() function required a complete re-write. More
+ information about this change can be found in the comments at the
+ beginning of this function. 4. Unit Tests. Unit tests for
+ ast_uri_encode, ast_uri_decode, and get_calleridname() have been
+ written. This involved the addition of the test_utils.c file for
+ testing the utils api. (closes issue #16299) Reported by: wdoekes
+ Patches: astsvn-16299-get_calleridname.diff uploaded by wdoekes
+ (license 717) get_calleridname_rewrite.diff uploaded by dvossel
+ (license 671) Tested by: wdoekes, dvossel, Nick_Lewis Review:
+ https://reviewboard.asterisk.org/r/469/
+
+2010-01-26 15:46 +0000 [r243118-243158] Russell Bryant <russell@digium.com>
+
+ * tests/test_substitution.c: Log the variable name being tested.
+
+ * tests/test_substitution.c: Update test_substitution to show
+ failures in the test log.
+
+ * funcs/func_aes.c: Update func_aes to its pre-ast_str_substitution
+ state. This change makes the AES tests in test_substitution.c
+ pass. We still need to work through what's going wrong in the
+ ast_str version.
+
+2010-01-26 01:56 +0000 [r242967-243077] Tilghman Lesher <tlesher@digium.com>
+
+ * tests/test_substitution.c: Fixing last errors in the conversion,
+ though it appears that the AES_* functions are still broken.
+
+ * tests/test_substitution.c: Using a dummy channel causes CDR()
+ testing to fail.
+
+ * tests/test_substitution.c: Wish I had gotten to the review before
+ this got submitted, because there's failures we need to address.
+
+ * /, main/Makefile, res/Makefile: Merged revisions 242969 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r242969 | tilghman | 2010-01-25 15:50:22 -0600 (Mon, 25 Jan 2010)
+ | 2 lines Err, and use the new menuselect define, too. ........
+
+ * build_tools/cflags.xml, /, build_tools/menuselect-deps.in,
+ configure, configure.ac: Merged revisions 242966 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r242966 | tilghman | 2010-01-25 15:36:33 -0600 (Mon, 25
+ Jan 2010) | 2 lines Only rebuild parsers by an option in
+ menuselect ........
+
+2010-01-25 21:32 +0000 [r242954-242965] Russell Bryant <russell@digium.com>
+
+ * tests/test_substitution.c, tests/test_heap.c,
+ tests/test_ast_format_str_reduce.c, tests/test_skel.c,
+ tests/test_sched.c: Make unit test modules depend on
+ TEST_FRAMEWORK instead of off by default.
+
+ * tests/test_substitution.c: Convert test_substitution module to
+ the unit test API. Review:
+ https://reviewboard.asterisk.org/r/474/
+
+2010-01-25 21:20 +0000 [r242933] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/oochannels.c,
+ addons/ooh323c/src/ooCalls.c: small corrections in call clearing
+
+2010-01-25 21:13 +0000 [r242904-242919] Olle Johansson <oej@edvina.net>
+
+ * main/pbx.c, main/manager.c, include/asterisk/pbx.h: Change api
+ for pbx_builtin_setvar to actually return error code if a
+ function can't be written to. This patch removes code that was
+ duplicated from pbx.c to manager.c in order to prevent API change
+ in released versions of Asterisk. There are propably also other
+ places that would benefit from reading the return code and react
+ if a function returns error codes on writing a value into it.
+
+ * main/manager.c, /: Merged revisions 242850 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r242850 | oej | 2010-01-25 21:03:38 +0100 (Mån, 25 Jan 2010) | 2
+ lines Report error when writing to functions returns error in AMI
+ setvar action ........
+
+2010-01-25 20:18 +0000 [r242857] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, main/Makefile, configure.ac, res/Makefile: Merged
+ revisions 242852 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r242852 | tilghman | 2010-01-25 14:15:45 -0600 (Mon, 25 Jan 2010)
+ | 2 lines Restore FreeBSD to able-to-compile-ish-mode ........
+
+2010-01-25 18:01 +0000 [r242812] Terry Wilson <twilson@digium.com>
+
+ * res/res_calendar.c: Fix INTERNAL_OBJ error on stop when
+ calendars.conf missing Initialize the calendars container before
+ calling load_config and return FAILURE on allocation failure.
+ Also, use the AST_MODULE_LOAD_* values for return values. Thanks
+ to rmudgett for pointing out the error and the need to use the
+ defined values for return
+
+2010-01-25 05:45 +0000 [r242719-242729] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/Makefile, res/Makefile: Merged revisions 242728 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r242728 | tilghman | 2010-01-24 23:42:22 -0600 (Sun, 24 Jan 2010)
+ | 2 lines Buildbot pointed out an error (thanks, buildbot!)
+ ........
+
+ * /, res/Makefile: Merged revisions 242723 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r242723 | tilghman | 2010-01-24 23:33:37 -0600 (Sun, 24 Jan 2010)
+ | 2 lines Oops, should have used CMD_PREFIX, not ECHO_PREFIX, for
+ the commands. ........
+
+ * /, main/Makefile: Merged revisions 242683 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r242683 | tilghman | 2010-01-24 23:13:28 -0600 (Sun, 24 Jan 2010)
+ | 2 lines Make the build of the Asterisk expression parser match
+ that of the AEL parser. ........
+
+2010-01-24 22:42 +0000 [r242645] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
+ addons/ooh323c/src/ooStackCmds.h,
+ addons/ooh323c/src/oochannels.c,
+ addons/ooh323c/src/ooCmdChannel.c,
+ addons/ooh323c/src/ooStackCmds.c: AST_CONTROL_CONNECTED_LINE
+ frame type processing added to setup DisplayIE field incorrect
+ q.931 message order filtered on incoming calls (first msg must be
+ setup, next must be not setup)
+
+2010-01-24 21:49 +0000 [r242607] Sean Bright <sean@malleable.com>
+
+ * res/res_phoneprov.c: Instead of crashing, allocate our header
+ ast_str before we try to use it. (closes issue #16680) Reported
+ by: lmadsen Patches: issue16680_20100122.patch uploaded by
+ seanbright (license 71) Tested by: lmadsen
+
+2010-01-24 06:40 +0000 [r242521] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ pbx/Makefile, res/Makefile, makeopts.in: Merged revisions 242520
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r242520 | tilghman | 2010-01-24 00:33:01 -0600 (Sun, 24 Jan 2010)
+ | 8 lines Only rebuild bison and flex source files on demand, if
+ bison and flex are detected by the configure script. Changed
+ after discussion on the -dev list about possible unnecessary
+ build failures, due to checkouts/untars causing these special
+ source files to possibly be newer than their resulting C files.
+ This should additionally ensure that nobody need learn about
+ extra Makefile arguments to ensure the proper files get rebuilt
+ when changes are made to these special source files. ........
+
+2010-01-22 21:45 +0000 [r242424] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/Makefile: Merged revisions 242423 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r242423 | tilghman | 2010-01-22 15:44:18 -0600 (Fri, 22 Jan 2010)
+ | 7 lines Rebuild from flex, bison sources when necessary. (issue
+ #14629) Reported by: Marquis Patches:
+ 20100121__issue14629.diff.txt uploaded by tilghman (license 14)
+ ........
+
+2010-01-22 16:20 +0000 [r242357] David Ruggles <thedavidfactor@gmail.com>
+
+ * apps/app_externalivr.c: Add send DTMF feature to ExternalIVR app
+ Implemented a new command 'D' that allows client IVRs to send
+ DTMF digits to the channel. (closes issue #16615) Reported by:
+ thedavidfactor Review: https://reviewboard.asterisk.org/r/465/
+
+2010-01-22 15:09 +0000 [r242317] Tilghman Lesher <tlesher@digium.com>
+
+ * tests/test_sched.c: The irony of not compile-testing a test
+ program before committing is killing me.
+
+2010-01-22 09:28 +0000 [r242227] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 242226 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r242226 | oej | 2010-01-22 10:19:30 +0100 (Fre, 22 Jan 2010) | 3
+ lines Initialize notify_types to NULL ........
+
+2010-01-22 04:57 +0000 [r242184-242186] Russell Bryant <russell@digium.com>
+
+ * main/test.c: Update the doxygenification of some comments.
+
+ * tests/test_sched.c: Convert scheduler API entry order test to the
+ test API. Review: https://reviewboard.asterisk.org/r/470/
+
+ * tests/test_skel.c: Add test API usage example to test_skel.c.
+ Review: https://reviewboard.asterisk.org/r/471/
+
+2010-01-21 22:37 +0000 [r242092] Mark Michelson <mmichelson@digium.com>
+
+ * main/acl.c: Add missing argument to ast_calloc calls.
+
+2010-01-21 21:05 +0000 [r242043] Olle Johansson <oej@edvina.net>
+
+ * main/acl.c: Make sure we initialize the ast_ha structure with
+ ast_calloc
+
+2010-01-21 15:27 +0000 [r241938] Sean Bright <sean@malleable.com>
+
+ * /, configure, configure.ac: Merged revisions 241932 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r241932 | seanbright | 2010-01-21 10:25:46 -0500 (Thu,
+ 21 Jan 2010) | 5 lines Fix configure check for PTHREAD_ONCE_INIT
+ when manually adding -Wall to CFLAGS. (closes issue #16666)
+ Reported by: romain_proformatique ........
+
+2010-01-21 15:14 +0000 [r241896] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_vpb.cc: Formats are inconsistent between even
+ 32-bit and 64-bit Linux. Use casts to ensure both compile.
+
+2010-01-21 14:10 +0000 [r241855-241856] Russell Bryant <russell@digium.com>
+
+ * main/test.c: Point to a useful reference on the XML output
+ format.
+
+ * main/test.c: Modify test results XML format to match the JUnit
+ format. When this code was developed, we came up with our own XML
+ format for the test output. I have since started looking at
+ integration with other tools, namely continuous integration
+ frameworks, and this format seems to be supported across a number
+ of applications. With these changes in place, I was able to get
+ Atlassian Bamboo to interpret the test results.
+
+2010-01-21 05:54 +0000 [r241766] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_math.c: Merged revisions 241765 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r241765 | tilghman | 2010-01-20 23:53:17 -0600 (Wed, 20 Jan 2010)
+ | 2 lines Guard against division by zero. ........
+
+2010-01-20 21:14 +0000 [r241627-241714] David Vossel <dvossel@digium.com>
+
+ * res/res_rtp_asterisk.c: rtp timestamp to timeval calculation fix
+ The rtp timestamp to timeval calculation was only accurate for
+ 8kHz audio. This patch corrects this. Review:
+ https://reviewboard.asterisk.org/r/468/ SWP-648
+
+ * Makefile, /: Merged revisions 241626 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r241626 | dvossel | 2010-01-20 14:00:04 -0600 (Wed, 20 Jan 2010)
+ | 6 lines fixes parsing error in Makefile. Some echo lines were
+ missing "; . Thanks to jparker for pointing out the problem.
+ ........
+
+2010-01-20 17:49 +0000 [r241581] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/cdr.c: Add Calling and Called Subaddress to CDR record
+ Requires 'callingsubaddr' and 'calledsubaddr' fields in backend
+ cdr. (closes issue #16600) Reported by: alecdavis Patches:
+ cdr_subaddr.diff.txt uploaded by alecdavis (license 585) Tested
+ by: alecdavis Review: https://reviewboard.asterisk.org/r/460/
+
+2010-01-20 13:01 +0000 [r241503] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_vpb.cc: Fix up compile breakage from
+ ast_tvdiff_ms() API change.
+
+2010-01-20 08:18 +0000 [r241416] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/pbx.c, channels/sig_pri.c: Update CDR variables as pbx
+ starts Allows CDR variables added in cdr.c:set_one_cid to become
+ visable during the call, by executing ast_cdr_update() early in
+ __ast_pbx run. Reverts sig_pri changes in trunk that are specific
+ to isdn technology only. (closes issue #16638) Reported by:
+ alecdavis Patches: cdr_update.diff3.txt uploaded by alecdavis
+ (license 585) Tested by: alecdavis
+
+2010-01-19 22:59 +0000 [r241366] Jeff Peeler <jpeeler@digium.com>
+
+ * main/pbx.c: Initialize data on the stack so that Park doesn't
+ interpret random arguments. passdata was only being set in
+ pbx_substitue_variables when arguments were passed. (closes issue
+ #16406) (closes issue #16586) Reported by: DLNoah Patches:
+ bug16586v2.patch uploaded by jpeeler (license 325) Tested by:
+ DLNoah
+
+2010-01-19 22:41 +0000 [r241364] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/janitor-projects.txt, apps/app_sendtext.c: Enable SendText to
+ send strings in encoded format. See
+ http://lists.digium.com/pipermail/asterisk-users/2010-January/243462.html
+
+2010-01-19 18:51 +0000 [r241314-241315] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_agent.c: small correction from 241314
+
+ * /, channels/chan_agent.c: Merged revisions 241227 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r241227 | jpeeler | 2010-01-19 11:22:18 -0600 (Tue, 19
+ Jan 2010) | 13 lines Fix deadlock in agent_read by removing call
+ to agent_logoff. One must always lock the agents list lock before
+ the agent private. agent_read locks the private immediately, so
+ locking the agents list lock is not an option (which is what
+ agent_logoff requires). Because agent_read already has access to
+ the agent private all that is necessary is to do the required
+ hanging up that agent_logoff performed. (closes issue #16321)
+ Reported by: valon24 Patches: bug16321.patch uploaded by jpeeler
+ (license 325) ........
+
+2010-01-19 17:42 +0000 [r241230] Jason Parker <jparker@digium.com>
+
+ * Makefile: Allow parallel make (-j) to work properly. After some
+ back and forth with the reporter, we came up with the necessary
+ changes. (closes issue #16489) Reported by: Chainsaw Patches:
+ asterisk-1.6.2.1-parallel-make-minimal.patch uploaded by Chainsaw
+ (license 723) Tested by: Chainsaw, qwell
+
+2010-01-19 00:28 +0000 [r241188] Tilghman Lesher <tlesher@digium.com>
+
+ * main/srv.c, res/res_agi.c, CHANGES, include/asterisk/srv.h:
+ Create iterative method for querying SRV results, and use that
+ for finding AGI servers. (closes issue #14775) Reported by:
+ _brent_ Patches: 20091215__issue14775.diff.txt uploaded by
+ tilghman (license 14) hagi-5.patch uploaded by brent (license
+ 388) Tested by: _brent_ Reviewboard:
+ https://reviewboard.asterisk.org/r/378/
+
+2010-01-19 00:24 +0000 [r241187] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/sig_pri.c: Update CDR variables before pbx starts
+ (overlap dial) Allows CDR variables added in cdr.c:set_one_cid to
+ become visable during the call. (issue #16638) Reported by:
+ alecdavis Patches: cdr_update.diff2.txt uploaded by alecdavis
+ (license 585) Tested by: alecdavis
+
+2010-01-18 22:31 +0000 [r241143] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, channels/chan_dahdi.c, channels/sig_analog.c,
+ main/features.c, pbx/pbx_dundi.c, main/enum.c,
+ include/asterisk/time.h, main/timing.c: Extend max call limit
+ duration from 24.8 days to 292+ million years. If the limit was
+ set past MAX_INT upon answering, the call was immediately hung up
+ due to overflow from the return of ast_tvdiff_ms (in
+ ast_check_hangup). The time calculation functions ast_tvdiff_sec
+ and ast_tvdiff_ms have been changed to return an int64_t to
+ prevent overflow. Also the reporter suggested adding a message
+ indicating the reason for the call hanging up. Given that the new
+ limit is so much higher, the message (which would only really be
+ useful in the overflow scenario) has been made a debug message
+ only. (closes issue #16006) Reported by: viraptor
+
+2010-01-18 22:03 +0000 [r241098] Jason Parker <jparker@digium.com>
+
+ * main/rtp_engine.c: Fix an RTP instance allocation failure on
+ Solaris. (closes issue #16543) Reported by: crjw Patches:
+ rtp_sin_family.patch uploaded by crjw (license 963) Tested by:
+ crjw, qwell
+
+2010-01-18 22:00 +0000 [r241097] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/sig_pri.c: Update CDR variables before pbx starts Allows
+ CDR variables added in cdr.c:set_one_cid to become visable during
+ the call. (closes issue #16638) Reported by: alecdavis Patches:
+ cdr_update.diff.txt uploaded by alecdavis (license 585)
+
+2010-01-18 19:57 +0000 [r241016] Sean Bright <sean@malleable.com>
+
+ * /, main/config.c: Merged revisions 241015 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r241015 | seanbright | 2010-01-18 14:54:19 -0500 (Mon, 18 Jan
+ 2010) | 12 lines Plug a memory leak when reading configs with
+ their comments. While reading through configuration files with
+ the intent of returning their full contents (comments
+ specifically) we allocated some memory and then forgot to free
+ it. This doesn't fix 16554 but clears up a leak I had in the lab.
+ (issue #16554) Reported by: mav3rick Patches:
+ issue16554_20100118.patch uploaded by seanbright (license 71)
+ Tested by: seanbright ........
+
+2010-01-18 19:26 +0000 [r241012] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_strings.c, CHANGES: Make HASHes inheritable across
+ channel creation.
+
+2010-01-18 18:00 +0000 [r240973-240974] David Ruggles <thedavidfactor@gmail.com>
+
+ * UPGRADE.txt: ExternalIVR information for UPGRADE.txt added a
+ paragraph about the fixes and changes to the ExternalIVR
+ application.
+
+ * doc/externalivr.txt: Updated ExternalIVR documentation Rewrote a
+ large portion of the existing documentation and added information
+ about the TCP/IP socket interface
+
+2010-01-18 17:45 +0000 [r240971] David Vossel <dvossel@digium.com>
+
+ * Makefile, CHANGES: transmit_silence_during_record replaced by
+ transmit_silence In asterisk.conf, transmit_silence_during_record
+ has been removed in favor of using only the transmit_silence
+ option. The transmit_silence_during_record option remains a valid
+ option in asterisk.conf, but has been removed from the sample
+ config and noted in CHANGES.
+
+2010-01-18 17:41 +0000 [r240969] David Ruggles <thedavidfactor@gmail.com>
+
+ * apps/app_externalivr.c: Add notification of interrupted file Add
+ file information to data element of T event so the file
+ information is sent to the client when it is interrupted.
+ Previously only notification of pending files that were dropped
+ was sent (closes issue #16147) Reported by: thedavidfactor Tested
+ by: thedavidfactor Review:
+ https://reviewboard.asterisk.org/r/449/
+
+2010-01-18 16:45 +0000 [r240842-240887] David Vossel <dvossel@digium.com>
+
+ * Makefile: updated transmit_silence option documentation in
+ asterisk.conf This patch updates the transmit_silence option to
+ better document why the option exists, and what it affects.
+ Thanks to russell for providing the verbage for this update.
+
+ * apps/app_queue.c: fixes spelling error. s/memeber/member
+
+2010-01-17 19:45 +0000 [r240717] Sean Bright <sean@malleable.com>
+
+ * main/pbx.c: Avoid a crash on Solaris when running 'core show
+ functions.' (closes issue #16309) Reported by: asgaroth
+
+2010-01-16 00:54 +0000 [r240667] Sean Bright <sean@malleable.com>
+
+ * res/res_musiconhold.c: Get MoH building on OpenSolaris.
+
+2010-01-15 23:50 +0000 [r240629] Tilghman Lesher <tlesher@digium.com>
+
+ * Makefile, main/asterisk.c: Err, oops, it was already the way I
+ intended.
+
+2010-01-15 23:09 +0000 [r240548-240552] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/doxygen/commits.h: Note where empty lines should
+ reside in commit messages.
+
+ * Makefile, /: Merged revisions 240547 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r240547 | russell | 2010-01-15 17:06:11 -0600 (Fri, 15 Jan 2010)
+ | 2 lines Fix a spelling error in the asterisk.conf sample.
+ ........
+
+2010-01-15 22:07 +0000 [r240505] Sean Bright <sean@malleable.com>
+
+ * res/res_timing_timerfd.c: Clarify error message in
+ res_timing_timerfd.
+
+2010-01-15 21:42 +0000 [r240421-240500] Tilghman Lesher <tlesher@digium.com>
+
+ * utils/astcanary.c: Oops, missed an include
+
+ * utils/astcanary.c, main/asterisk.c: The previous attempt at using
+ a pipe to guarantee astcanary shutdown did not work. We're
+ revisiting the previous patch, albeit with a method that
+ overcomes the prior criticism that it was not POSIX-compliant.
+ (closes issue #16602) Reported by: frawd Patches:
+ 20100114__issue16602.diff.txt uploaded by tilghman (license 14)
+ Tested by: frawd
+
+ * apps/app_directed_pickup.c, main/features.c,
+ include/asterisk/manager.h: Add pickup event to AMI. Also, fix
+ AMI documentation. (closes issue #16431) Reported by: syspert
+ Patches: 20100112__issue16431.diff.txt uploaded by tilghman
+ (license 14)
+
+2010-01-15 20:58 +0000 [r240420] Mark Michelson <mmichelson@digium.com>
+
+ * main/utils.c: Make sure to set owner_line, ownder_func, and
+ owner_file in ast_calloc_with_stringfields. Asterisk would crash
+ on startup if MALLOC_DEBUG were set in menuselect. This is
+ because the manager action UpdateConfig had to resize its string
+ field allocation to set the description. When the resize
+ occurred, ast_copy_string would crash because we were attempting
+ to copy a string from a NULL pointer. Setting the strings
+ initially makes the code much less crashy.
+
+2010-01-15 20:58 +0000 [r240415-240419] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Make sure that the limit is N, not N - 1.
+
+ * /, apps/app_voicemail.c: Merged revisions 240414 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r240414 | tilghman | 2010-01-15 14:52:27 -0600 (Fri, 15
+ Jan 2010) | 15 lines Disallow leaving more than maxmsg
+ voicemails. This is a possibility because our previous method
+ assumed that no messages are left in parallel, which is not a
+ safe assumption. Due to the vmu structure duplication, it was
+ necessary to track in-process messages via a separate structure.
+ If at some point, we switch vmu to an ao2-reference-counted
+ structure, which would eliminate the prior noted duplication of
+ structures, then we could incorporate this new in-process
+ structure directly into vmu. (closes issue #16271) Reported by:
+ sohosys Patches: 20100108__issue16271.diff.txt uploaded by
+ tilghman (license 14) 20100108__issue16271__trunk.diff.txt
+ uploaded by tilghman (license 14)
+ 20100108__issue16271__1.6.0.diff.txt uploaded by tilghman
+ (license 14) Tested by: jsutton ........
+
+2010-01-15 20:41 +0000 [r240411] Russell Bryant <russell@digium.com>
+
+ * main/event.c: Ensure payload type is properly checked when
+ comparing against cached events. (closes issue #16607) Reported
+ by: ddv2005 Patches: event.patch uploaded by ddv2005 (license
+ 769)
+
+2010-01-15 18:21 +0000 [r240368] Sean Bright <sean@malleable.com>
+
+ * main/pbx.c, main/manager.c, res/res_smdi.c, apps/app_meetme.c,
+ channels/chan_sip.c, cel/cel_tds.c, main/features.c,
+ res/res_phoneprov.c, cdr/cdr_tds.c, apps/app_jack.c: Convert a
+ few places to use ast_calloc_with_stringfields where applicable.
+
+2010-01-15 16:51 +0000 [r240329] Russell Bryant <russell@digium.com>
+
+ * configure: Update configure script for an OSP toolkit related
+ change.
+
+2010-01-15 16:28 +0000 [r240328] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configs/sip.conf.sample: Clarify RTP NAT handling a bit.
+
+2010-01-14 23:13 +0000 [r240226-240271] Sean Bright <sean@malleable.com>
+
+ * res/res_config_ldap.c: Plug a memory leak in res_config_ldap.
+ (closes issue #16257) Reported by: nito Patches:
+ issue16257_20100111.diff uploaded by seanbright (license 71)
+
+ * res/res_timing_timerfd.c: If we aren't running on a machine that
+ support CLOCK_MONOTONIC, don't load. Group developed and tested
+ by seanbright, Corydon76, Kobaz, and Amorsen.
+
+2010-01-14 18:03 +0000 [r240179] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c: Fix broken call pickup The problem was the
+ OUTGOING flag was not getting set properly on the channel,
+ resulting in pickup failing as ast_read thought the call was
+ inbound. Refer to 170393 for a more verbose description as this
+ is the same exact change. (closes issue #16539) Reported by:
+ syspert Patches: bug16539.patch uploaded by jpeeler (license 325)
+ Tested by: syspert
+
+2010-01-14 17:34 +0000 [r240129-240175] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Similarly, ensure that matchcid is duplicated
+ correctly when merging contexts.
+
+ * main/pbx.c: Ensure that the callerid is NULL when the parent is
+ effectively NULL. This applies only to pattern-match hints, which
+ create exact-match hints on the fly.
+
+2010-01-14 16:14 +0000 [r240078] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/udptl.c: This change fixes a few bugs in the way the far max
+ IFP was calculated that were introduced in r231692. (closes issue
+ #16497) Reported by: globalnetinc Patches:
+ udptl-max-ifp-fix1.diff uploaded by mnicholson (license 96)
+ Tested by: globalnetinc
+
+2010-01-14 14:38 +0000 [r240039] Leif Madsen <lmadsen@digium.com>
+
+ * doc/building_queues.txt (added): Add documentation about how to
+ build queues. Add a how-to set of documentation about building
+ queues with Asterisk. This documentation is based on Asterisk
+ 1.6.2 but should work on most versions with minor modifications.
+ (closes issue #16237) Reported by: lmadsen Patches: Building
+ Queues (FINAL).txt uploaded by lmadsen (license 10) Tested by:
+ pdhales, lmadsen, cmdrwalrus
+
+2010-01-13 23:22 +0000 [r239920-239997] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Oops, another tag error
+
+ * main/pbx.c: Oops, missed a closing tag
+
+ * main/pbx.c, include/asterisk/pbx.h: Add the TESTTIME() dialplan
+ function, which permits testing GotoIfTime. Specifically, by
+ setting TESTTIME() to a particular date and time, you can test
+ whether a dialplan correctly branches as was intended. This was
+ developed after recent questions on the -users list on how to
+ test their holiday dialplan logic. (closes issue #16464) Reported
+ by: tilghman Patches: 20100112__issue16464.diff.txt uploaded by
+ tilghman (license 14) Review:
+ https://reviewboard.asterisk.org/r/458/
+
+ * main/ast_expr2f.c, main/ast_expr2.fl: Flex uses fwrite
+ incorrectly, which breaks the build. Providing a workaround.
+
+2010-01-13 19:48 +0000 [r239839] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 239838 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r239838 | jpeeler | 2010-01-13 13:43:33 -0600 (Wed, 13 Jan 2010)
+ | 11 lines Fix regression for timed out parked call returning to
+ caller This issue seems to have been exposed by the fix in 160390
+ whereby using a masquerade prevented a crash. The new channel
+ used in the masquerade was not copying the macro information from
+ the old channel. (closes issue #15459) Reported by: djrodman
+ Patches: patch_15459.txt uploaded by mnick (license ) ........
+
+2010-01-13 19:31 +0000 [r239834] Leif Madsen <lmadsen@digium.com>
+
+ * configs/extensions.conf.sample: Add more examples to
+ extensions.conf showing how to use various functionality and
+ provide commonly useful features. (closes issue #16090) Reported
+ by: pprindeville Patches: extensions.conf-bugid16090.patch#3
+ uploaded by pprindeville (license 347) Tested by: tzafrir,
+ pprindeville, lmadsen
+
+2010-01-13 18:16 +0000 [r239797] Tilghman Lesher <tlesher@digium.com>
+
+ * main/Makefile, main/ast_expr2f.c, main/ast_expr2.fl: Code
+ previously added to ast_expr2f.c warranted a change in the source
+ file ast_expr2.fl. Also, made a Makefile change to ensure that
+ the expression parser C source files get regenerated correctly,
+ when we need that to happen.
+
+2010-01-13 16:31 +0000 [r239712] David Vossel <dvossel@digium.com>
+
+ * Makefile, main/channel.c, apps/app_waitforring.c,
+ apps/app_waitforsilence.c: add silence gen to wait apps
+ asterisk.conf's 'transmit_silence' option existed before this
+ patch, but was limited to only generating silence while recording
+ and sending DTMF. Now enabling the transmit_silence option
+ generates silence during wait times as well. To achieve this,
+ ast_safe_sleep has been modified to generate silence anytime no
+ other generators are present and transmit_silence is enabled.
+ Wait apps not using ast_safe_sleep now generate silence when
+ transmit_silence is enabled as well. (closes issue #16524)
+ Reported by: kobaz (closes issue #16523) Reported by: kobaz
+ Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/456/
+
+2010-01-13 10:45 +0000 [r239663-239665] Olle Johansson <oej@edvina.net>
+
+ * main/poll.c: MAX() moved to utils.h
+
+ * channels/chan_sip.c: SIP Show channelstats fix - use float
+ division to show proper stats (closes issue #15819) Reported by:
+ klaus3000 Patches: asterisk-sip-show-channelstats-trunk.txt
+ uploaded by klaus3000 (license 65) Tested by: klaus3000, oej This
+ patch is for trunk only and will be blocked in 1.6.2
+
+2010-01-13 07:02 +0000 [r239624-239625] TransNexus OSP Development <support@transnexus.com>
+
+ * doc/tex/channelvariables.tex: Updated channel variable list of
+ osplookup application.
+
+ * apps/app_osplookup.c: Updated XML doc for OSP.
+
+2010-01-12 19:58 +0000 [r239571] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Blank callerid and NULL callerid should not compare
+ equal. The second is the default state for matching CID in the
+ dialplan (no matching) while the first matches one particular
+ CallerID. This is a regression. (fixes AST-314, SWP-611)
+
+2010-01-12 18:55 +0000 [r239525] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/cdr.c: add Dialed Number Identifier (DNID) field to cdr
+ records. reviewboard link:
+ https://reviewboard.asterisk.org/r/455/ Reported by: alecdavis
+ Tested by: alecdavis Patch cdr_dnid.diff2.txt uploaded by
+ alecdavis (license 585)
+
+2010-01-12 18:22 +0000 [r239520] Leif Madsen <lmadsen@digium.com>
+
+ * configs/sip.conf.sample: Note that direct T.38 is not supported.
+ (closes issue #16411) Reported by: stanusr Patches:
+ __20091210-sip.conf.sample-documentation.txt uploaded by lmadsen
+ (license 10)
+
+2010-01-12 17:09 +0000 [r239473] Sean Bright <sean@malleable.com>
+
+ * res/res_config_ldap.c: Fix crash in res_config_ldap. We need to
+ allocate enough room for 2 pointers, not 2 characters. (closes
+ issue #16397) Reported by: bklang Patches: res_config_ldap.patch
+ uploaded by applsplatz (license 949) Tested by: applsplatz
+
+2010-01-12 16:14 +0000 [r239427] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes text support in sdp answer The code
+ that handled setting 'm=text' in the sdp was not executing in the
+ correct order. The check to see if text was needed came after the
+ check to add 'm=text' to the sdp, this resulted in 'm=text'
+ always being set to 0 because it looked like text was never
+ required. (closes issue #16457) Reported by: peterj Patches:
+ textportinsdp.diff uploaded by peterj (license 951)
+ issue16457.diff uploaded by dvossel (license 671) Tested by:
+ peterj
+
+2010-01-12 07:48 +0000 [r239389] Olle Johansson <oej@edvina.net>
+
+ * include/asterisk/astmm.h: Adding Tilghman's documentation from
+ asterisk-dev to the actual file.
+
+2010-01-12 03:21 +0000 [r239152-239308] Tilghman Lesher <tlesher@digium.com>
+
+ * /, contrib/scripts/safe_asterisk: Merged revisions 239307 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r239307 | tilghman | 2010-01-11 21:18:36 -0600 (Mon, 11 Jan 2010)
+ | 8 lines Portability and other fixes for the safe_asterisk
+ script (closes issue #16416) Reported by: bklang Patches:
+ safe_asterisk-compat-1.patch uploaded by bklang (license 919)
+ 20100106__issue16416__trunk.diff.txt uploaded by tilghman
+ (license 14) Tested by: bklang ........
+
+ * contrib/init.d/rc.mandriva.asterisk,
+ contrib/init.d/rc.debian.asterisk,
+ contrib/init.d/rc.redhat.asterisk,
+ contrib/init.d/rc.gentoo.asterisk,
+ contrib/init.d/rc.slackware.asterisk,
+ contrib/init.d/rc.archlinux.asterisk,
+ contrib/init.d/rc.suse.asterisk: Add LSB headers to init scripts.
+ (closes issue #14864) Reported by: lathama Patches:
+ lsb-init-info-debian.diff uploaded by pkempgen (license 169)
+
+ * res/res_pktccops.c: Socket level option is SOL_SOCKET, not
+ SO_SOCKET. (issue #16580)
+
+ * Makefile, contrib/init.d/rc.mandriva.asterisk,
+ contrib/init.d/rc.debian.asterisk,
+ contrib/init.d/rc.redhat.asterisk,
+ contrib/init.d/rc.suse.asterisk: Permit more options in the
+ Makefile as to startup options (closes issue #16454) Reported by:
+ syspert Patches: 20091228__issue16454__3.diff.txt uploaded by
+ tilghman (license 14) Tested by: syspert
+
+ * Makefile: Including bundle1.o breaks Tiger and Leopard (issue
+ #16449)
+
+ * addons/cdr_mysql.c, configs/cdr_mysql.conf.sample: Permit dates
+ and times to be stored in timezones other than the default
+ (typically, UTC) (closes issue #16401) Reported by: lordmortis
+
+2010-01-11 16:41 +0000 [r239111-239114] Sean Bright <sean@malleable.com>
+
+ * res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
+ res/res_calendar_caldav.c, res/res_clialiases.c: Pass NULL for
+ the ao2_callback function pointer instead of duplicating cb_true.
+
+ * main/astobj2.c: Fix ao2_callback when both OBJ_MULTIPLE and
+ OBJ_NODATA are passed. There is an issue which only affects trunk
+ and the new ao2_callback OBJ_MULTIPLE implementation. When both
+ OBJ_MULTIPLE and OBJ_NODATA are passed, only the first object is
+ visited, regardless of what is returned by the specified
+ callback. This causes a problem when we are clearing a container,
+ i.e.: ao2_callback(container, OBJ_UNLINK | OBJ_NODATA |
+ OBJ_MULTIPLE, NULL, NULL); Only unlinks the first object. This
+ patch resolves this. (closes issue #16564) Reported by: pj
+ Patches: issue16564_20100111.diff uploaded by seanbright (license
+ 71) Tested by: pj, seanbright Review:
+ https://reviewboard.asterisk.org/r/457/
+
+ * main/test.c: Fix spelling of 'category.'
+
+2010-01-10 19:37 +0000 [r239074] Tilghman Lesher <tlesher@digium.com>
+
+ * addons/chan_ooh323.c, main/frame.c, channels/chan_iax2.c:
+ According to POSIX, the capital L modifier applies only to
+ floating point types. Fixes a crash on Solaris. (closes issue
+ #16572) Reported by: crjw Patches: frame_changes.patch uploaded
+ by crjw (license 963) Plus several others found and fixed by me
+
+2010-01-10 17:53 +0000 [r239037] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooq931.h, addons/ooh323c/src/oochannels.c,
+ addons/ooh323c/src/ooq931.c: add docallbacks flag in q931decode
+ function because when we decode received q931 packet we must do
+ callbacks and when we print sended q931 packet we must not.
+
+2010-01-10 06:56 +0000 [r239000] Tilghman Lesher <tlesher@digium.com>
+
+ * Makefile, main/asterisk.c: It's been long enough -- make the
+ behavior introduced in 1.6 the default.
+
+2010-01-09 01:08 +0000 [r238916] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, /: Merged revisions 238915 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r238915 | tilghman | 2010-01-08 18:57:58 -0600 (Fri, 08 Jan 2010)
+ | 6 lines -1 is interpreted as an error, intead of the maximum
+ mask. (closes issue #16241) Reported by: vnovy Patches:
+ manager.c.patch uploaded by vnovy (license 922) ........
+
+2010-01-08 23:30 +0000 [r238835] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 238834 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r238834 | jpeeler | 2010-01-08 17:28:37 -0600 (Fri, 08 Jan 2010)
+ | 4 lines Stop a crash when no peer is passed to masq_park_call.
+ (distantly related to issue #16406) ........
+
+2010-01-08 22:54 +0000 [r238754-238795] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c: Add the class actually used in the
+ MusicOnHold start event. (closes issue #16499) Reported by:
+ syspert Patches: mohclass.patch uploaded by syspert (license 938)
+
+ * res/res_agi.c: Initialize variables that we attempt to free
+ later. (closes issue #16302) Reported by: yahsyn Patches:
+ 20091124__issue16302.diff.txt uploaded by tilghman (license 14)
+ Tested by: yahsyn
+
+2010-01-08 21:04 +0000 [r238716] Matthew Nicholson <mnicholson@digium.com>
+
+ * tests/test_ast_format_str_reduce.c (added): Added a test for
+ ast_format_reduce_str(). (related to issue #16560)
+
+2010-01-08 19:39 +0000 [r238635] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/audiohook.h, main/audiohook.c: fixes
+ AUDIOHOOK_INHERIT regression During the process of removing an
+ audiohook from one channel and attaching it to another the
+ audiohook's status is updated to DONE and then back to whatever
+ it was previously. Typically updating the status after setting it
+ to DONE is not a good idea because DONE can trigger unrecoverable
+ audiohook destruction events... because of this a conditional
+ check was added to audiohook_update_status to explicitly prevent
+ the audiohook from ever changing after being set to DONE. It was
+ this check that prevented audiohook inherit from work properly
+ though. Now ast_audiohook_move_by_source is treated as a special
+ exception, as the audiohook must be returned to its previous
+ status after attaching it to the new channel. This is only a safe
+ operation because the audiohook's lock is held the entire time,
+ otherwise this could cause trouble. (closes issue #16522)
+ Reported by: corruptor
+
+2010-01-08 19:32 +0000 [r238630] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/file.c: Merged revisions 238629 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r238629 | mnicholson | 2010-01-08 13:20:44 -0600 (Fri, 08 Jan
+ 2010) | 5 lines Properly calculate the remaining space in the
+ output string when reducing format strings. (closes issue #16560)
+ Reported by: goldwein ........
+
+2010-01-08 17:18 +0000 [r238583] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c: Stop trying to find a parking space after
+ traversing the parkinglot one time. (closes issue #16428)
+ Reported by: Yasuhiro Konishi
+
+2010-01-07 21:24 +0000 [r238527] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Fix using the wrong pointer type in
+ do_idle_thread().
+
+2010-01-07 20:42 +0000 [r238361-238492] David Vossel <dvossel@digium.com>
+
+ * main/channel.c: fixes ast_transfer stall until hangup if called
+ with a channel that doesn't support transfers ast_transfer sets
+ res to 0 if there is no technology transfer function, but then
+ tests for it to be negative before deciding to do an early exit.
+ As a result, it will will wait for an AST_CONTROL_TRANSFER
+ message that will never come. (closes issue #16424) Reported by:
+ davidw Patches: Issue_16424_trunk_234134.patch uploaded by davidw
+ (license 780)
+
+ * /, channels/chan_iax2.c: Merged revisions 238411 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07
+ Jan 2010) | 10 lines fixes crash in "scheduled_destroy" in
+ chan_iax A signed short was used to represent a callnumber. This
+ is makes it possible to attempt to access the iaxs array with a
+ negative index. (closes issue #16565) Reported by: jensvb
+ ........
+
+ * channels/chan_sip.c: Change in sip show channels display format
+ allowing more digits for CID (closes issue #16459) Reported by:
+ Rzadzins Patches: chan_sip_longer_cid.patch uploaded by Rzadzins
+ (license 953)
+
+ * apps/app_queue.c: cli 'queue show' formatting fix. queue name was
+ truncated over 12 characters (closes issue #16078) Reported by:
+ RoadKill Patches: quequename_limit.patch uploaded by ppyy
+ (license 906) Tested by: dvossel
+
+2010-01-07 09:14 +0000 [r238313] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * configs/sip.conf.sample: Document the usefulness of explicit
+ udp:// in the register string
+
+2010-01-06 21:45 +0000 [r238231] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_cdr.c: Merged revisions 238230 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r238230 | tilghman | 2010-01-06 15:41:55 -0600 (Wed, 06 Jan 2010)
+ | 4 lines Revise documentation on disposition values to the
+ actual values used. (closes issue #16289) Reported by: wdoekes
+ ........
+
+2010-01-06 20:37 +0000 [r238134-238181] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_meetme.c: Fix misreverting from 177158. (closes issue
+ #15725) Reported by: shanermn Patches: v1-15725.patch uploaded by
+ dimas (license 88) Tested by: shanermn
+
+ * main/features.c: Fix channel name comparison for bridge
+ application. The channel name comparison was not comparing the
+ whole string and therefore if one channel name was a substring of
+ the other, the bridge would fail. (closes issue #16528) Reported
+ by: telecos82 Patches: res_features_r236843.diff uploaded by
+ telecos82 (license 687)
+
+2010-01-06 16:36 +0000 [r238091] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/test.h: fixes test.c compile issue when
+ TEST_FRAMEWORK is not enabled The ast_test_status_update()
+ function is defined in test.h. When TEST_FRAMEWORK is not enabled
+ a macro is defined as a no-op place holder for this function. The
+ macro did not contain the correct number of arguments. This
+ caused a compile error. Much thanks to wdoekes for reporting the
+ issue and supplying the patch!
+
+2010-01-06 15:35 +0000 [r238014] Sean Bright <sean@malleable.com>
+
+ * addons/format_mp3.c: Fix reading samples from format_mp3 after
+ ast_seekstream/ast_tellstream. There is a bug when using
+ ast_seekstream/ast_tellstream with format_mp3 in that the file
+ read position is not reset before attempting to read samples. So
+ when we seek to determine the maximum size of the file (as in
+ res_agi's STREAM FILE) we weren't then resetting the file pointer
+ so that we could properly read samples. This patch addresses that
+ (in a similar manner to format_wav.c). (closes issue #15224)
+ Reported by: rbd Patches: 20091230_addons_1.4_issue15224.diff
+ uploaded by seanbright (license 71) Tested by: rbd, seanbright
+ Review: https://reviewboard.asterisk.org/r/453
+
+2010-01-06 15:19 +0000 [r238010] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_mp3.c: Merged revisions 238009 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r238009 | russell | 2010-01-06 09:18:22 -0600 (Wed, 06 Jan 2010)
+ | 7 lines Resolve a crash due to an ast_frame not being fully
+ initialized. (closes issue #16531) Reported by: john8675309
+ (closes SWP-615) ........
+
+2010-01-06 06:53 +0000 [r237968] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Whoa, duplicate setting (dead code).
+
+2010-01-05 23:08 +0000 [r237920] David Vossel <dvossel@digium.com>
+
+ * apps/app_queue.c: fixes holdtime playback issue in app_queue When
+ reporting hold time, the number of seconds should be mod 60.
+ Otherwise audio playback could be something like "2 minutes 123
+ seconds" rather than "2 minutes 3 seconds". Also, the "minute"
+ sound file is missing, so for the moment until that file can be
+ created the "minutes" file is used instead. (closes issue #16168)
+ Reported by: nickilo Patches: patch-unified-trunk-rev-222176
+ uploaded by nickilo (license ) Tested by: nickilo, wonderg
+
+2010-01-05 20:56 +0000 [r237882] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c: Mismerged a bit.
+
+2010-01-05 19:29 +0000 [r237839] David Vossel <dvossel@digium.com>
+
+ * main/pbx.c: fixes subscriptions being lost after 'module reload'
+ During a module reload if multiple extension configs are present,
+ such as both extensions.conf and extensions.ael, watchers for one
+ config's hints will be lost during the merging of the other
+ config. This happens because hint watchers are only preserved for
+ the current config being merged. The old context list is
+ destroyed after the merging takes place, meaning any watchers
+ that were not perserved will be removed. Now all hints are
+ preserved during merging regardless of what config file is being
+ merged. These hints are only restored if they are present within
+ the new context list. (closes issue #16093) Reported by: jlaroff
+
+2010-01-05 18:57 +0000 [r237804] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h, channels/sig_pri.c: Removed unused
+ parameters from analog_available() and sig_pri_available().
+
+2010-01-05 18:46 +0000 [r237802-237803] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, CHANGES: Add a missing part of the connected
+ line work into trunk. Part of the work done for connected line
+ was to add an optional argument to the 'f' option to allow for
+ the connected party information of the outgoing channel to be set
+ to the argument provided. This was overlooked during the merge of
+ the work to trunk and is being added back now. The CHANGES file
+ has also been updated to note this change.
+
+ * CHANGES: Spell "aficionado" like someone who isn't stupid.
+
+2010-01-05 17:26 +0000 [r237699-237749] Russell Bryant <russell@digium.com>
+
+ * main/utils.c: Fix build of utility apps that include utils.c.
+
+ * /, main/utils.c: Merged revisions 237697 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237697 | russell | 2010-01-05 11:13:28 -0600 (Tue, 05 Jan 2010)
+ | 7 lines Change a NOTICE log message to DEBUG where it belongs.
+ (closes issue #16479) Reported by: alexrecarey (closes SWP-577)
+ ........
+
+2010-01-05 16:08 +0000 [r237656] Michiel van Baak <michiel@vanbaak.info>
+
+ * apps/app_mixmonitor.c: Make CLI command 'mixmonitor start|stop
+ <channel> work again. (closes issue #16534) Reported by:
+ jlaguilar Fix as suggested by jlaguilar in the bugreport
+
+2010-01-04 21:48 +0000 [r237406-237574] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/say.c: Merged revisions 237573 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237573 | tilghman | 2010-01-04 15:45:46 -0600 (Mon, 04 Jan 2010)
+ | 6 lines Bounds checking for input string (closes issue #16407)
+ Reported by: qwell Patches: 20100104__issue16407.diff.txt
+ uploaded by tilghman (license 14) ........
+
+ * main/pbx.c, /: Merged revisions 237493 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237493 | tilghman | 2010-01-04 14:57:35 -0600 (Mon, 04 Jan 2010)
+ | 8 lines Regression in issue #15421 - Pattern matching (closes
+ issue #16482) Reported by: wdoekes Patches:
+ astsvn-16482-betterfix.diff uploaded by wdoekes (license 717)
+ 20091223__issue16482.diff.txt uploaded by tilghman (license 14)
+ Tested by: wdoekes, tilghman ........
+
+ * main/config.c: Oops, didn't compile (thanks, kpfleming)
+
+ * main/config.c: Further reduce the encoded blank values back to
+ blank in the realtime API. (closes issue #16533) Reported by:
+ sergee Patches: 200100104__issue16533.diff.txt uploaded by
+ tilghman (license 14) Tested by: sergee
+
+ * main/pbx.c, /, res/res_agi.c, include/asterisk/channel.h: Merged
+ revisions 237405 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010)
+ | 16 lines Add a flag to disable the Background behavior, for AGI
+ users. This is in a section of code that relates to two other
+ issues, namely issue #14011 and issue #14940), one of which was
+ the behavior of Background when called with a context argument
+ that matched the current context. This fix broke FreePBX,
+ however, in a post-Dial situation. Needless to say, this is an
+ extremely difficult collision of several different issues. While
+ the use of an exception flag is ugly, fixing all of the issues
+ linked is rather difficult (although if someone would like to
+ propose a better solution, we're happy to entertain that
+ suggestion). (closes issue #16434) Reported by: rickead2000
+ Patches: 20091217__issue16434.diff.txt uploaded by tilghman
+ (license 14) 20091222__issue16434__1.6.1.diff.txt uploaded by
+ tilghman (license 14) Tested by: rickead2000 ........
+
+2010-01-04 16:39 +0000 [r237327] David Vossel <dvossel@digium.com>
+
+ * apps/app_queue.c: app_queue segfaults if realtime field uniqueid
+ is NULL (closes issue #16385) Reported by: haakon Patches:
+ app_queue.c.patch uploaded by haakon (license 880)
+ app_queue.c.patch_v2 uploaded by dvossel (license 671) Tested by:
+ haakon
+
+2010-01-04 16:24 +0000 [r237323] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_agi.c: Fix timeout for AGI command speech recognize.
+ (closes issue #16297) Reported by: semond
+
+2010-01-04 16:20 +0000 [r237319] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 237318 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r237318 | tilghman | 2010-01-04 10:18:59 -0600 (Mon, 04
+ Jan 2010) | 3 lines It's also possible for the Local channel to
+ directly execute an Application. Reviewboard:
+ https://reviewboard.asterisk.org/r/452/ ........
+
+2010-01-04 07:55 +0000 [r237284] Olle Johansson <oej@edvina.net>
+
+ * res/res_pktccops.c, channels/chan_mgcp.c: - Disable res_pktccops
+ by default - Add dependency in chan_mgcp that was missing - Add a
+ small amount of doc to the source code
+
+2010-01-04 03:38 +0000 [r237250] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c: 1. Added reporting operator names in
+ AuthReq. 2. Added retrieving operator names from AuthRsp and
+ exporting them.
+
+2010-01-02 16:35 +0000 [r237213] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: global_contact_ha was renamed in trunk
+
+2010-01-02 09:54 +0000 [r237136] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 237135 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237135 | oej | 2010-01-02 10:52:30 +0100 (Lör, 02 Jan 2010) | 2
+ lines Release memory of the contact acl before unloading module
+ ........
+
+2009-12-30 23:51 +0000 [r237098] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooq931.c,
+ addons/ooh323c/src/ooCalls.c: small q931 processing and
+ signalling corrections don't decode UUIE from Q931StatusMessage
+ clean call without callIdentifier data don't start tcs/msd
+ exchange procedure after call proceeding received (closes issue
+ #16365) Reported by: benngard2 Tested by: may213, benngard2
+
+2009-12-30 22:30 +0000 [r237050] Jason Parker <jparker@digium.com>
+
+ * main/say.c, doc/lang/vietnamese.ods (added),
+ apps/app_voicemail.c: Add app_voicemail and say.c support for
+ Vietnamese. Also add an XXX comment that I'm baffled nobody has
+ ever complained about. We say "first message", and then we go
+ into language-specific stuff where we proceed to say..."first
+ message". (closes issue #15053) Reported by: dinhtrung Patches:
+ vietnamese.ods uploaded by dinhtrung (license 776)
+ app_voicemail.c.diff uploaded by dinhtrung (license 776) (closes
+ issue #15626) Reported by: dinhtrung Patches: say.c.diff uploaded
+ by dinhtrung (license 776)
+
+2009-12-30 21:59 +0000 [r236982] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 236981 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r236981 | tilghman | 2009-12-30 15:57:10 -0600 (Wed, 30
+ Dec 2009) | 9 lines Don't queue frames to channels that have no
+ means to process them. (closes issue #15609) Reported by: aragon
+ Patches: 20091230__issue16521__1.4__chan_local_only.diff.txt
+ uploaded by tilghman (license 14) Tested by: aragon Review:
+ https://reviewboard.asterisk.org/r/452/ ........
+
+2009-12-30 21:09 +0000 [r236893-236902] Jeff Peeler <jpeeler@digium.com>
+
+ * utils/ael_main.c: One more LOW_MEMORY compile fix.
+
+ * channels/chan_sip.c, main/cli.c: Fix compiling with LOW_MEMORY.
+ Modified handle_verbose to be LOW_MEMORY aware, removed old RTP
+ related code in chan_sip. (closes issue #16381) Reported by:
+ michael_iedema Patches: ast_complete_source_filename.patch
+ uploaded by michael iedema (license 942) modified by me
+
+2009-12-30 17:53 +0000 [r236802-236847] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_adaptive_odbc.c, cel/cel_adaptive_odbc.c: When the field
+ is blank, don't warn about the field being unable to be coerced,
+ just skip the column. (closes
+ http://lists.digium.com/pipermail/asterisk-dev/2009-December/041362.html)
+ Reported by Nic Colledge on the -dev list, fixed by me.
+
+ * channels/chan_sip.c: Shut down the SIP session timers more
+ gracefully, in order to prevent a possible crash. (closes issue
+ #16452) Reported by: corruptor Patches:
+ 20091221__issue16452.diff.txt uploaded by tilghman (license 14)
+ Tested by: corruptor
+
+2009-12-29 10:59 +0000 [r236756] TransNexus OSP Development <support@transnexus.com>
+
+ * configs/osp.conf.sample, apps/app_osplookup.c, configure.ac: 1.
+ Updated for OSP Toolkit 3.6.0. 2. Added service type ported
+ number query. 3. Formated code.
+
+2009-12-28 22:09 +0000 [r236713] Jason Parker <jparker@digium.com>
+
+ * main/ast_expr2.y, main/ast_expr2.c: Allow "REMAINDER" to function
+ properly in expressions. (closes issue #16427) Reported by:
+ wdoekes Patches: ast16-reminder-remainder.patch uploaded by
+ wdoekes (license 717) Tested by: wdoekes
+
+2009-12-28 17:37 +0000 [r236667] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Use recommended option, not deprecated
+ option. (closes issue #16515) Reported by: ManChicken
+
+2009-12-28 15:22 +0000 [r236510-236613] Sean Bright <sean@malleable.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/threadstorage.h: Merged revisions 236585 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236585 | seanbright | 2009-12-28 10:12:08 -0500 (Mon, 28 Dec
+ 2009) | 7 lines Try a test compile to see if PTHREAD_ONCE_INIT
+ requires extra braces. There was conditional code (based on build
+ platform) to optioinally wrap PTHREAD_ONCE_INIT in braces that
+ was removed since it is fixed in newer versions of
+ Solaris/OpenSolaris, but I am still running into it on Solaris 10
+ x86 so add a configure-time check for it. ........
+
+ * /, apps/app_meetme.c: Merged revisions 236509 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec
+ 2009) | 12 lines Avoid a crash with large numbers of MeetMe
+ conferences. Similar to changes made to Queue(), when we have
+ large numbers of conferences in meetme.conf (1000s) and we use
+ alloca()/strdupa(), we can blow out the stack and crash, so
+ instead just use a single fixed buffer. (closes issue #16509)
+ Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded
+ by seanbright (license 71) Tested by: seanbright ........
+
+2009-12-27 18:20 +0000 [r236434] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 236433 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236433 | tilghman | 2009-12-27 12:19:38 -0600 (Sun, 27 Dec 2009)
+ | 2 lines Turn on colors in the daemon, since there's many
+ requests for it on Ubuntu. ........
+
+2009-12-26 15:27 +0000 [r236358] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, sounds/Makefile: Merged revisions 236357 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236357 | kpfleming | 2009-12-26 09:26:17 -0600 (Sat, 26 Dec
+ 2009) | 1 line update to latest releases with zero uid/gid
+ ........
+
+2009-12-23 19:17 +0000 [r236304-236312] David Vossel <dvossel@digium.com>
+
+ * CHANGES: Update CHANGES to reflect new QUEUE_MEMBER option,
+ "ready"
+
+ * apps/app_queue.c: QUEUE_MEMBER(..., ready) counts only ready
+ agents, not free agents wrapping up The QUEUE_MEMBER dialplan
+ function can return total members, logged-in members and "free"
+ members count. A member is counted as "free" immediately after
+ his call ends, even though its wrap-up time, if specified in
+ queues.conf, has not yet expired, and the queue will not actually
+ route a call to it. This Patch introduces a new "ready" option
+ that only counts free agents no longer in the wrap up time
+ period. (closes issue #16240) Reported by: kkm Patches:
+ appqueue-memberfun-readyoption-trunk.diff uploaded by kkm
+ (license 888) Tested by: kkm, dvossel
+
+ * CHANGES, apps/app_queue.c: update CHANGES to reflect new 'R'
+ app_queue option plus a minor optimization to the feature patch
+ (issue #16384)
+
+ * apps/app_queue.c: new parameter 'R' to the Queue application The
+ 'R' argument stops moh and indicates ringing once the agent is
+ ringing. This allows the person in the queue to know their call
+ is potentially about to be answered. (closes issue #16384)
+ Reported by: haakon Patches: new_app_queue.c.patch uploaded by
+ haakon (license 880) Tested by: haakon, loloski, dvossel
+
+2009-12-23 18:25 +0000 [r236183-236300] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_stack.c: AGI may be invoked from outside the dialplan
+ (closes issue #16510) Reported by: atis Patches:
+ 20091223__issue16510.diff.txt uploaded by tilghman (license 14)
+ Tested by: atis
+
+ * /, res/res_agi.c: Merged revisions 236184 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236184 | tilghman | 2009-12-22 20:55:24 -0600 (Tue, 22 Dec 2009)
+ | 4 lines If EXEC only gets a single argument, don't crash when
+ the second is used. (closes issue #16504) Reported by: bklang
+ ........
+
+ * include/asterisk/test.h: Allow test_heap.c to compile when
+ AST_DEVMODE is true, but TEST_FRAMEWORK is false
+
+ * apps/app_voicemail.c: Actually use tmp for something (brings
+ trunk back into sync with 1.6 branches).
+
+2009-12-22 21:53 +0000 [r236027-236144] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: fixes iax "can't compress subclass
+ 4294967295" error (closes issue #16456) Reported by: dvossel
+ Tested by: dvossel
+
+ * /, channels/chan_sip.c: Merged revisions 236062 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236062 | dvossel | 2009-12-22 10:58:19 -0600 (Tue, 22 Dec 2009)
+ | 11 lines fixes issue with p->method incorrectly set to ACK It
+ is possible for a second ACK to come in for a retransmitted
+ message. If an ack does not match an unacked message in our
+ queue, restore the previous p->method as this ACK is completely
+ ignored. (closes issue #16295) Reported by: omolenkamp Patches:
+ issue16295_v2.diff uploaded by dvossel (license 671) ........
+
+ * CHANGES: update CHANGES to reflect the addition of the test
+ framework
+
+ * include/asterisk/test.h (added), build_tools/cflags-devmode.xml,
+ tests/test_heap.c, main/test.c (added),
+ include/asterisk/_private.h, main/asterisk.c: Unit Test Framework
+ API The Unit Test Framework is a new API that manages
+ registration and execution of unit tests in Asterisk with the
+ purpose of verifying the operation of C functions. The Framework
+ consists of a single test manager accompanied by a list of
+ registered test functions defined within the code. A test is
+ defined, registered, and unregistered from the framework using a
+ set of macros which allow the test code to only be compiled
+ within asterisk when the TEST_FRAMEWORK flag is enabled in
+ menuselect. This allows the test code to exist in the same file
+ as the C functions it intends to verify. Registered tests may be
+ viewed and executed via a set of new CLI commands. CLI commands
+ are also present for generating and exporting test results into
+ xml and txt formats. For more information and use cases please
+ refer to the documentation provided at the beginning of the
+ test.h file. Review: https://reviewboard.asterisk.org/r/447/
+
+2009-12-21 19:54 +0000 [r235941] Jeff Peeler <jpeeler@digium.com>
+
+ * /, res/res_monitor.c: Merged revisions 235940 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r235940 | jpeeler | 2009-12-21 13:43:41 -0600 (Mon, 21 Dec 2009)
+ | 13 lines Change Monitor to not assume file to write to does not
+ contain pathing. 227944 changed the fname_base argument to always
+ append the configured monitor path. This change was necessary to
+ properly compare files for uniqueness. If a full path is given
+ though, nothing needs to be appended and that is handled
+ correctly now. (closes issue #16377) (closes issue #16376)
+ Reported by: bcnit Patches: res_monitor.c-issue16376-1.patch
+ uploaded by dant (license 670) ........
+
+2009-12-21 18:51 +0000 [r235904] Kevin P. Fleming <kpfleming@digium.com>
+
+ * contrib/upstart/asterisk.upstart-0.3.9, include/asterisk/cel.h,
+ main/say.c, include/asterisk/channel.h,
+ include/asterisk/manager.h, channels/sig_pri.c,
+ include/asterisk/logger.h, include/asterisk/http.h,
+ include/asterisk/callerid.h, include/asterisk/syslog.h,
+ channels/chan_dahdi.c, include/asterisk/app.h,
+ include/asterisk/doxyref.h, include/asterisk/event.h,
+ channels/sig_analog.c, channels/chan_misdn.c,
+ contrib/upstart/asterisk.user.conf,
+ include/asterisk/rtp_engine.h,
+ include/asterisk/security_events.h,
+ include/asterisk/stringfields.h: Change all refererences to 1.6.3
+ to be 1.8, since that will be the next feature release
+
+2009-12-21 17:00 +0000 [r235822] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/features.c: Merged revisions 235821 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r235821 | tilghman | 2009-12-21 10:45:03 -0600 (Mon, 21 Dec 2009)
+ | 8 lines Send parking lot announcement to the channel which
+ parked the call, not the park-ee. (closes issue #16234) Reported
+ by: yeshuawatso Patches: 20091210__issue16234.diff.txt uploaded
+ by tilghman (license 14) 20091221__issue16234__1.4.diff.txt
+ uploaded by tilghman (license 14) Tested by: yeshuawatso ........
+
+2009-12-20 08:22 +0000 [r235740-235774] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/dsp.c: restarts busydetector (if enabled) when DTMF is
+ received after call is bridged. (closes issue 0016389) Reported
+ by: alecdavis Tested by: alecdavis Patch
+ dtmf_busydetector.diff2.txt uploaded by alecdavis (license 585)
+
+ * apps/app_dial.c, CHANGES: app_dial optional parameter to option
+ 'r' to allow play indication from indications.conf (closes issue
+ #14504) Reported by: alecdavis Tested by: alecdavis,jsmith Patch
+ app_dial.play_ring_indications.diff7.txt uploaded by alecdavis
+ (license 585)
+
+2009-12-18 22:51 +0000 [r235660] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /, include/asterisk/cdr.h: Merged revisions
+ 235635 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009)
+ | 48 lines Correct CDR dispositions for BUSY/FAILED This patch is
+ simple in that it reorders the disposition defines so that the
+ fix for issue 12946 works properly (the default CDR disposition
+ was changed to AST_CDR_NOANSWER). Also, the
+ AST_CDR_FLAG_ORIGINATED flag was set in ast_call to ensure all
+ CDR records are written. The side effects of CDR changes are
+ scary, so I'm documenting the test cases performed to attempt to
+ catch any regressions. The following tests were all performed
+ using 1.4 rev 195881 vs head (235571) + patch: A calls B C calls
+ B (busy) Hangup C Hangup A (Both SIP and features) A calls B A
+ blind transfers to C Hangup C (Both SIP and features) A calls B A
+ attended transfers to C Hangup C A calls B A attended transfers
+ to C (SIP) C blind transfers to A (features) Hangup A All of the
+ test scenario CDRs matched. The following tests were performed
+ just with the patch to ensure proper operation (with
+ unanswered=yes): exten =>s,1,Answer exten =>s,n,ResetCDR(w) exten
+ =>s,n,ResetCDR(w) exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w)
+ (closes issue #16180) Reported by: aatef Patches: bug16180.patch
+ uploaded by jpeeler (license 325) ........
+
+2009-12-18 22:40 +0000 [r235573-235656] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 235652 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r235652 | tilghman | 2009-12-18 16:39:30 -0600 (Fri, 18
+ Dec 2009) | 2 lines Revise verbiage, per #asterisk-dev discussion
+ ........
+
+ * /, configure, configure.ac: Merged revisions 235572 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r235572 | tilghman | 2009-12-18 15:18:16 -0600 (Fri, 18
+ Dec 2009) | 2 lines Point to the typical missing package, not the
+ cryptic "termcap support". ........
+
+2009-12-17 23:21 +0000 [r235521] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Remove some old code for going to the 'fax'
+ extension when a T.38 switchover occurs. This would have already
+ happened when we detected the CNG tone so this was basically a
+ noop.
+
+2009-12-17 17:19 +0000 [r235422] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 235421 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r235421 | tilghman | 2009-12-17 11:17:51 -0600 (Thu, 17 Dec 2009)
+ | 8 lines Use context from which Macro is executed, not macro
+ context, if applicable. Also, ensure that the extension COULD
+ match, not just that it won't match more. (closes issue #16113)
+ Reported by: OrNix Patches: 20091216__issue16113.diff.txt
+ uploaded by tilghman (license 14) Tested by: OrNix ........
+
+2009-12-17 00:52 +0000 [r235342-235382] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c: Fix call forwarding
+ for analog phones. (closes issue #16440) Reported by: mmichelson
+
+ * configs/jabber.conf.sample, include/asterisk/jabber.h, CHANGES,
+ res/res_jabber.c: Add auth_policy option to jabber.conf for auto
+ user registration. The option is global and currently the
+ acceptable values as noted in the sample config are accept or
+ deny. (closes issue #15228) Reported by: lp0
+
+2009-12-16 05:24 +0000 [r235298] Jared Smith <jaredsmith@jaredsmith.net>
+
+ * /, configs/sip.conf.sample: Merged revisions 235181 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r235181 | jsmith | 2009-12-15 15:07:55 -0600 (Tue, 15
+ Dec 2009) | 4 lines Add a line showing that we can use CIDR
+ notation. patch by jsmith, after discussion with jtodd ........
+
+2009-12-16 00:31 +0000 [r235265] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c, CHANGES: Enhance AMI redirect to allow channels
+ to be redirected to different places. New parameters
+ ExtraContext, ExtraExtension, and ExtraPriority have been added
+ to redirect the second channel to a different location.
+ Previously, it was only possible to redirect both channels to the
+ same place. (closes issue #15853) Reported by: haakon Patches:
+ trunk-manager.c.patch uploaded by haakon (license 880) Tested by:
+ jpeeler
+
+2009-12-15 23:51 +0000 [r235229] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/strings.h: Is it Friday yet?
+
+2009-12-15 23:41 +0000 [r235226] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c: Change match criteria existence in
+ ast_channel_cmp_cb to use ast_strlen_zero. (closes issue #16161)
+ Reported by: may213 Patches: core-show-channel.patch uploaded by
+ may213 (license 454)
+
+2009-12-15 18:43 +0000 [r235132] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: reverse minor sip registration regression A
+ registration regression caused by a code tweak in (issue #14331)
+ and a bug fix in (issue #15539) caused some sip registration
+ config entries to be constructed incorrectly. Origially issue
+ #14331 contained the code tweak as well as a bug fix, but since
+ the issue was reported as a tweak the bug fix portion was moved
+ into issue #15539. Both the tweak and the bug fix contained minor
+ incorrect logic that resulted in some SIP registrations to fail.
+ (issue #14331) (issue #15539)
+
+2009-12-15 15:33 +0000 [r235053] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 235052 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r235052 | tilghman | 2009-12-15 09:29:24 -0600 (Tue, 15 Dec 2009)
+ | 4 lines Mandatory argument checking (closes issue #16446)
+ Reported by: nicchap ........
+
+2009-12-15 14:35 +0000 [r235010] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_fax.c: spandsp does in fact support V.17 modulation at
+ 14.4 kilobits per second, so we should generate T38MaxBitRate of
+ 14400 (even though that doesn't really affect the FAX
+ transmission much at all)
+
+2009-12-15 07:18 +0000 [r234855-234976] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * apps/app_directory.c: Support option 'n', as applications like
+ Playback, Background etc. Suggested on asterisk-dev as trivial
+ application change. Reported by: alecdavis Tested by: alecdavis
+
+ * main/dsp.c: Whitespace.
+
+ * main/dsp.c: restarts busydetector (if enabled) when DTMF is
+ received. (closes issue #16389) Reported by: alecdavis Tested by:
+ alecdavis Patch dtmf_busydetector.diff.txt uploaded by alecdavis
+ (license 585)
+
+ * apps/app_directory.c: fixes escape to extensions 'o' and 'a', for
+ digits '0' and '*' (closes issue #16437) Reported by: alecdavis
+ Tested by: alecdavis Patch extension_o_a_fix.diff.txt uploaded by
+ alecdavis (license 585)
+
+ * apps/app_directory.c: ast_stream_and_wait(chan,dir-usingkeypad)
+ didn't capture the dialled DTMF. (closes issue #16409) Reported
+ by: alecdavis Tested by: alecdavis Patch bug_16409.diff.txt
+ uploaded by alecdavis (license 585)
+
+2009-12-14 23:16 +0000 [r234820] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
+ Allow greetings-only mailboxes for Voicemail. (closes issue
+ #15132) Reported by: floletarmo Patches: voicemail_changes.patch
+ uploaded by floletarmo (license 784) (with some additional
+ changes by me)
+
+2009-12-14 21:32 +0000 [r234776] Jason Parker <jparker@digium.com>
+
+ * apps/app_readexten.c: Allow tonelist as argument to ReadExten.
+ ReadExten already supported playing a tonezone from
+ indications.conf. It now has the ability to use a tonelist like
+ 440+480/2000|0/4000 (closes issue #15185) Reported by: jcovert
+ Patches: app_readexten.c.patch uploaded by jcovert (license 551)
+ Tested by: qwell Patch modified by me, to maintain backwards
+ compatibility.
+
+2009-12-14 21:13 +0000 [r234700] Tilghman Lesher <tlesher@digium.com>
+
+ * /, build_tools/make_version_c, build_tools/make_version_h: Merged
+ revisions 234699 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r234699 | tilghman | 2009-12-14 15:09:56 -0600 (Mon, 14 Dec 2009)
+ | 5 lines Deal with the situation where .flavor exists but
+ .version does not. Also make the script slightly more portable,
+ in keeping with autoconf syntax. (closes issue #14737) Reported
+ by: davidw ........
+
+2009-12-14 17:19 +0000 [r234631] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/imapstorage.tex, /: Update IMAP build documentation.
+ Update the IMAP build documentation to show how to build on
+ 64-bit platforms. (issue #16433) Reported by: shrift Tested by:
+ lmadsen
+
+2009-12-14 16:08 +0000 [r234572] Sean Bright <sean@malleable.com>
+
+ * main/timing.c: The default rate for 'timing test' is actually
+ 50/sec, not 100/sec as advertised.
+
+2009-12-14 10:46 +0000 [r234526] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 234492 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r234492 | oej | 2009-12-14 11:16:00 +0100 (Mån, 14 Dec 2009) | 8
+ lines Stop sending 183's after call hangup. There where still
+ cases where the 183 keep-alive mechanism would not stop sending
+ 183's even though the Asterisk server had sent a final reply to
+ the invite. EDVX-28 ........
+
+2009-12-13 09:41 +0000 [r234458] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Trim leading/trailing spaces from the filename, to
+ deal with common user error.
+
+2009-12-11 23:17 +0000 [r234380] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 234379 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r234379 | jpeeler | 2009-12-11 16:37:21 -0600 (Fri, 11 Dec 2009)
+ | 11 lines Fix talking detection status after conference user is
+ muted. This patch ensures that when a conference user is muted
+ that the accompanying AMI Meetme talking off event is sent. Also,
+ the meetme list output is updated to show the muted user as
+ unmonitored. (closes issue #16247) Reported by: dimas Patches:
+ v3-16247.patch uploaded by dimas (license 88) ........
+
+2009-12-10 21:01 +0000 [r234256] Jason Parker <jparker@digium.com>
+
+ * Makefile, /: Merged revisions 234255 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r234255 | qwell | 2009-12-10 14:58:09 -0600 (Thu, 10 Dec 2009) |
+ 9 lines Fix unselecting of menuselect options via GLOBAL_MAKEOPTS
+ and USER_MAKEOPTS. (closes issue #16296) Reported by: abelbeck
+ Patches: issue16296-20091210.diff uploaded by qwell (license 4)
+ (abelbeck described a fix, which I expanded upon) Tested by:
+ abelbeck, qwell, lmadsen ........
+
+2009-12-10 18:56 +0000 [r234210] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c: Missed a case that emits a WARNING where
+ none is warranted.
+
+2009-12-10 17:31 +0000 [r234173] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_meetme.c, apps/app_page.c, main/app.c, CHANGES: Add
+ audio announcement option to app_page As described in the CHANGES
+ file: * MeetMe has a new option 'G' to play an announcement
+ before joining a conference. * Page has a new option 'A(x)' which
+ will playback an announcement simultaneously to all paged phones
+ (and optionally excluding the caller's one using the new option
+ 'n') before the call is bridged. To add the new option to meetme,
+ the conference flag options had to be extended to 64 bits.
+ (closes issue #14365) Reported by: dferrer Patches:
+ page_announce.patch uploaded by dferrer (license 525) modified by
+ me Review: https://reviewboard.asterisk.org/r/188/
+
+2009-12-10 16:24 +0000 [r234129] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 234095 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r234095 | tilghman | 2009-12-10 10:08:20 -0600 (Thu, 10 Dec 2009)
+ | 9 lines When we receive no response at all to our INVITE, allow
+ the channel to be destroyed. (closes issue #15627) Reported by:
+ falves11 Patches: 20091209__issue15627__1.6.0.diff.txt uploaded
+ by tilghman (license 14) 20091209__issue15627__1.4.diff.txt
+ uploaded by tilghman (license 14) Tested by: falves11 Review:
+ https://reviewboard.asterisk.org/r/446/ (closes issue #15716)
+ Reported by: dant (closes issue #16270) Reported by: corruptor
+ (closes issue #15356) Reported by: falves11 (issue #16382)
+ Reported by: lftsy ........
+
+2009-12-09 23:35 +0000 [r233967-234055] Russell Bryant <russell@digium.com>
+
+ * UPGRADE.txt, CHANGES: Move an entry from CHANGES to UPGRADE.txt.
+
+ * UPGRADE.txt, CHANGES: Move an entry from CHANGES that should be
+ in UPGRADE.txt.
+
+ * CHANGES: Provide a real description of LOCAL_PEEK().
+
+ * CHANGES: Remove a feature from CHANGES that was listed twice for
+ 1.6.2.
+
+ * CHANGES: Fix up the faxdetect entry in CHANGES. This feature was
+ listed as a 1.6.2 feature, even though it's in all 1.6.X
+ versions. The description of the feature was also no longer
+ accurate.
+
+ * CHANGES: Remove an entry from CHANGES that is already in
+ UPGRADE.txt (where it should be).
+
+2009-12-08 18:40 +0000 [r233718-233732] Tilghman Lesher <tlesher@digium.com>
+
+ * addons/res_config_mysql.c: Typo pointed out on #asterisk-dev (by
+ atis_work)
+
+ * res/res_musiconhold.c: Find another ref leak and change how we
+ manage module references. (closes issue #16388, closes issue
+ #16279, closes issue #16390) Reported by: parisioa Patches:
+ 20091208__issue16388.diff.txt uploaded by tilghman (license 14)
+ Tested by: parisioa, tilghman Review:
+ https://reviewboard.asterisk.org/r/442/
+
+2009-12-08 18:00 +0000 [r233692] Russell Bryant <russell@digium.com>
+
+ * formats/format_sln.c, formats/format_wav.c,
+ formats/format_ogg_vorbis.c, formats/format_sln16.c,
+ formats/format_wav_gsm.c, formats/format_siren7.c,
+ formats/format_ilbc.c, formats/format_vox.c,
+ formats/format_pcm.c, formats/format_h263.c,
+ formats/format_g723.c, formats/format_h264.c,
+ formats/format_g726.c, formats/format_siren14.c,
+ formats/format_jpeg.c, formats/format_gsm.c,
+ formats/format_g729.c: Set a module load priority for format
+ modules. A recent change to app_voicemail made it such that the
+ module now assumes that all format modules are available while
+ processing voicemail configuration. However, when autoloading
+ modules, it was possible that app_voicemail was loaded before the
+ format modules. Since format modules don't depend on anything,
+ set a module load priority on them to ensure that they get loaded
+ first when autoloading. This fix applies to trunk, 1.6.1, and
+ 1.6.2. The fix for 1.4 and 1.6.0 will require a different
+ approach since the module load priority functionality is not
+ present in the module API. (issue #16412) Reported by: jiddings
+
+2009-12-07 23:28 +0000 [r233611] David Vossel <dvossel@digium.com>
+
+ * main/utils.c: fixes incorrect logic in ast_uri_encode issue
+ #16299
+
+2009-12-07 23:10 +0000 [r233577] Atis Lezdins <atis@iq-labs.net>
+
+ * contrib/valgrind.supp: Fix compatibility with valgrind 3.3 and
+ older. (noticed in issue #16388) Reported by: parisioa Patches:
+ valgrind.supp uloaded by atis (license 242) Tested by: atis,
+ parisioa
+
+2009-12-07 19:48 +0000 [r233545] David Ruggles <thedavidfactor@gmail.com>
+
+ * apps/app_externalivr.c: Fix TCP Client interface Fix a couple of
+ very minor bugs that prevent the socket client from working. The
+ wrong set of properties were used in one place and the size of
+ the address variable isn't set if the host name is an ip address.
+ Also includes a fix for a bug that was introduced previously.
+ (closes issue #16121) Reported by: thedavidfactor Tested by:
+ thedavidfactor Review: https://reviewboard.asterisk.org/r/439/
+
+2009-12-07 18:08 +0000 [r233472] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 233471 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r233471 | dvossel | 2009-12-07 12:07:38 -0600 (Mon, 07 Dec 2009)
+ | 9 lines fixes missing Contact header angle brackets (closes
+ issue #16298) Reported by: mgernoth Patches:
+ reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested
+ by: dvossel ........
+
+2009-12-07 17:59 +0000 [r233468] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/jabber.h, CHANGES, res/res_jabber.c: Add
+ applications JabberJoin, JabberLeave, JabberSendGroup for XMPP
+ groupchat (closes issue #14352) Reported by: fiddur Patches:
+ trunk-14352-2.diff uploaded by phsultan (license 73) Tested by:
+ fiddur
+
+2009-12-07 16:14 +0000 [r233394] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Do not reject SDP packets describing only
+ non audio streams. (closes issue #16387) Reported by: zalex1953
+ Patches: media-level-c-fix1.diff uploaded by mnicholson (license
+ 96) Tested by: mnicholson, zalex1953
+
+2009-12-06 07:01 +0000 [r233358] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/compat.h, main/strcompat.c, main/app.c: Move
+ implementation of closefrom(3) from app.c to strcompat.c
+
+2009-12-04 21:54 +0000 [r233280] David Vossel <dvossel@digium.com>
+
+ * configs/iax.conf.sample, /: Merged revisions 233279 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r233279 | dvossel | 2009-12-04 15:54:01 -0600 (Fri, 04
+ Dec 2009) | 7 lines clarify requirecalltoken option in
+ iax.sample.conf (closes issue #16223) Reported by: bklang
+ Patches: clarify-iax-requirecalltoken.patch uploaded by bklang
+ (license 919) ........
+
+2009-12-04 21:06 +0000 [r233239] Tilghman Lesher <tlesher@digium.com>
+
+ * main/translate.c: Using the builtin function breaks OpenBSD 4.2
+ (closes issue #16395) Reported by: jtodd
+
+2009-12-04 20:21 +0000 [r233121-233235] David Vossel <dvossel@digium.com>
+
+ * CHANGES: update CHANGES file for .m3u support in Mp3Player
+ application
+
+ * apps/app_mp3.c: .m3u support for Mp3Player app (closes issue
+ #14823) Reported by: macli Patches: app_mp3.diff1 uploaded by
+ macli (license ) Tested by: macli, dvossel
+
+ * CHANGES: update CHANGES for new queue option,
+ penaltymemberslimit.
+
+ * apps/app_queue.c: changes penaltymemberslimit to use scanf for
+ config value parsing
+
+ * configs/queues.conf.sample, apps/app_queue.c: new queue option,
+ penaltymemberslimit, disregards penalty on too few queue members
+ when enabled (closes issue #14559) Reported by: fiddur Patches:
+ trunk-199584-1.diff uploaded by fiddur (license 678) Tested by:
+ fiddur, dvossel
+
+ * /, apps/app_voicemail.c: Merged revisions 233116 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r233116 | dvossel | 2009-12-04 11:21:34 -0600 (Fri, 04
+ Dec 2009) | 6 lines document and rename strip_control() in
+ app_voicemail (closes issue #16291) Reported by: wdoekes ........
+
+2009-12-04 17:18 +0000 [r233100] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 233092 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r233092 | russell | 2009-12-04 11:12:47 -0600 (Fri, 04 Dec 2009)
+ | 7 lines Only do frame payload check for HOLD frames. This code
+ was added for helping to debug the source of invalid HOLD frames.
+ However, a side effect of this is that it will incorrectly report
+ errors for frames that have an integer payload. Make the check
+ for this block specific to the HOLD frame case. ........
+
+2009-12-04 17:15 +0000 [r233093] Matthias Nick <mnick@digium.com>
+
+ * pbx/pbx_config.c: Parse global variables or expressions in hint
+ extensions Parse global variables or expressions in hint
+ extensions. Like: exten => 400,hint,DAHDI/i2/${GLOBAL(var)}
+ (closes issue #16166) Reported by: rmudgett Tested by: mnick,
+ rmudgett
+
+2009-12-04 16:55 +0000 [r233059-233089] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: Let's unlock the lines list after the
+ AST_LIST_TRAVERSE instead of inside it.
+
+ * channels/chan_skinny.c: Only assign line and device in
+ handle_transfer_button when we have a subchannel. (closes issue
+ #16040) Reported by: ebroad
+
+2009-12-04 16:08 +0000 [r233050] Tilghman Lesher <tlesher@digium.com>
+
+ * addons/res_config_mysql.c: Update the mysql driver to always
+ return NULL columns, as this is needed for the realtime API to
+ work correctly. (closes issue #16138) Reported by: sohosys
+ Patches: 20091029__issue16138.diff.txt uploaded by tilghman
+ (license 14) Tested by: sohosys
+
+2009-12-04 15:38 +0000 [r233046] Matthias Nick <mnick@digium.com>
+
+ * /, main/dsp.c: Merged revisions 233014 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r233014 | mnick | 2009-12-04 09:17:03 -0600 (Fri, 04 Dec 2009) |
+ 11 lines Warning message gets displayed only once Added
+ additional field 'int display_inband_dtmf_warning', which when
+ set to '1' displays the warning ('Inband DTMF is not supported on
+ codec %s. Use RFC2833'), and when set to '0' doesn't display the
+ warning. Otherwise you would get hundreds of warnings every
+ second. (closes issue #15769) Reported by: falves11 Patches:
+ patch_15769_14.txt uploaded by mnick (license 874) Tested by:
+ mnick, falves11 ........
+
+2009-12-04 05:26 +0000 [r232854-232982] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_pktccops.c: Buildbot complained
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ res/res_pktccops.c: OS X does not define MSG_NOSIGNAL, but it
+ does have a socket option SO_NOSIGPIPE. (closes issue #16178)
+ Reported by: oej
+
+ * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Add
+ pagerdateformat, to allow shorter dates for SMS messages. (closes
+ issue #16263) Reported by: andrew Patches: pagerdate.patch
+ uploaded by andrew (license 240) (with a slight modification by
+ me)
+
+ * /, apps/app_voicemail.c: Merged revisions 232820 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r232820 | tilghman | 2009-12-03 14:10:19 -0600 (Thu, 03
+ Dec 2009) | 8 lines Deprecate "cz" in favor of "cs". Also, change
+ the use of language codes so that language registers as a prefix,
+ rather than an exact match. (closes issue #16272) Reported by:
+ patrol-cz Patches: 20091203__issue16272.diff.txt uploaded by
+ tilghman (license 14) ........
+
+2009-12-03 20:26 +0000 [r232853] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
+ addons/ooh323c/src/ooh245.c: jitterbuffer setup correction
+ correction of double pointer references from previous rev
+
+2009-12-03 08:47 +0000 [r232738-232771] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c: Replaced two deprecated functions of OSP
+ Toolkit.
+
+ * apps/app_osplookup.c: Added custom info support.
+
+2009-12-03 00:38 +0000 [r232700] Jeff Peeler <jpeeler@digium.com>
+
+ * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
+ Extend voicemail to allow IMAP folders to be specified per
+ mailbox. Previously only possible per context, new option called
+ imapfolder. (closes issue #14298) Reported by: jablko Patches:
+ patch-200906202 uploaded by jablko (license 675)
+
+2009-12-03 00:09 +0000 [r232660-232661] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c: Remove debugging line
+
+ * include/asterisk/astobj2.h, res/res_musiconhold.c: Fix multiple
+ issues with musiconhold, which led to classes not getting
+ destroyed properly. * Classes are now tracked past removal from
+ the core container, and module removal is actively prevented
+ until all references are freed. * A hanging reference stored in
+ the channel has been removed. This could have caused a mismatch
+ and the music state not properly cleared, if two or more reloads
+ occurred between MOH being stopped and MOH being restarted. * In
+ certain circumstances, duplicate classes were possible. * A race
+ existed at reload time between a process being killed and the
+ thread responsible for reading from the related pipe respawning
+ that process. * Several reference counts have also been
+ corrected. At least one could have caused deleted classes to
+ stick around forever, consuming resources. This originally
+ manifested as MOH external processes that were not killed at
+ reload time. (closes issue #16279, closes issue #16207) Reported
+ by: parisioa, dcabot Patches: 20091202__issue16279__2.diff.txt
+ uploaded by tilghman (license 14) Tested by: parisioa, tilghman
+
+2009-12-02 23:27 +0000 [r232657] David Vossel <dvossel@digium.com>
+
+ * UPGRADE.txt, CHANGES: update CHANGES and UPGRADE.txt for early
+ media behavior change between 1.6.1 and 1.6.2 (closes issue
+ #16212) Reported by: miki
+
+2009-12-02 22:17 +0000 [r232587] David Ruggles <thedavidfactor@gmail.com>
+
+ * apps/app_externalivr.c: Prevent double closing of FDs by EIVR
+ This caused a problem when asterisk was under heavy load and
+ running both AGI and EIVR applications. EIVR would close an FD at
+ which point it would be considered freed and be used by a new AGI
+ instance the second close would then close the FD now in use by
+ AGI. (closes issue #16305) Reported by: diLLec Tested by:
+ thedavidfactor, diLLec Review:
+ https://reviewboard.asterisk.org/r/436/
+
+2009-12-02 22:02 +0000 [r232582] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c, /: Merged revisions 232581 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232581 | jpeeler | 2009-12-02 15:57:42 -0600 (Wed, 02 Dec 2009)
+ | 7 lines Send ack (response/message) after receiving manager
+ action userevent (closes issue #16264) Reported by: dimas
+ Patches: event-ack.patch uploaded by dimas (license 88) ........
+
+2009-12-02 21:37 +0000 [r232580] Matthew Nicholson <mnicholson@digium.com>
+
+ * addons/chan_mobile.c: Fix support for multiline SMS messages in
+ chan_mobile. (closes issue #16278) Reported by: Artem Patches:
+ multiline-sms-fix2.diff uploaded by mnicholson (license 96)
+ Tested by: Artem
+
+2009-12-02 21:32 +0000 [r232576] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c: Make manager response to "Action: events" finish
+ with empty line (closes issue #16275) Reported by: vnovy Patches:
+ manager.c.diff uploaded by vnovy (license 922)
+
+2009-12-02 21:13 +0000 [r232544] Matthew Nicholson <mnicholson@digium.com>
+
+ * addons/chan_mobile.c: Do something with the service indicator so
+ that asterisk does not attempt to use a chan_mobile endpoint that
+ does not have service. (closes issue #16132) Reported by: nikkk
+ Patches: service-indicator2.diff uploaded by mnicholson (license
+ 96) Tested by: nikkk
+
+2009-12-02 20:10 +0000 [r232442-232510] Joshua Colp <jcolp@digium.com>
+
+ * CHANGES, main/asterisk.c, doc/asterisk.sgml: Add an 'X' option to
+ the asterisk application which enables #exec for configuration
+ files. This option can be used to enable #exec support in the
+ asterisk.conf configuration file. (closes issue #16260) Reported
+ by: atis Patches: exec_includes.patch uploaded by atis (license
+ 242)
+
+ * apps/app_record.c, CHANGES: Add an option to Record which enables
+ a mode where any DTMF digit will terminate recording. (closes
+ issue #15436) Reported by: Vince Patches: app_record.diff
+ uploaded by Vince (license 823) Tested by: dbrooks
+
+2009-12-02 17:18 +0000 [r232365] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Do not change the exten string field or
+ rebuild the contact header on an inbound sip_pvt if the outbound
+ call is redirected.
+
+2009-12-02 17:06 +0000 [r232356] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_amd.c: Merged revisions 232355 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232355 | file | 2009-12-02 13:04:52 -0400 (Wed, 02 Dec 2009) | 5
+ lines Fix a bug where if you hung up very quickly after calling
+ AMD it would overwrite the AMDSTATUS of HANGUP with TOOLONG.
+ (closes issue #16239) Reported by: CGMChris ........
+
+2009-12-02 17:00 +0000 [r232351] David Vossel <dvossel@digium.com>
+
+ * /, main/acl.c: Merged revisions 232350 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232350 | dvossel | 2009-12-02 10:59:18 -0600 (Wed, 02 Dec 2009)
+ | 6 lines ast_outaddrfor doesn't do htons() on port, looks odd in
+ strace. (closes issue #16290) Reported by: wdoekes ........
+
+2009-12-02 16:40 +0000 [r232345] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Add support for handling the 415 Unsupported
+ media type response like we do for a 488 Not acceptable here
+ response. (closes issue #16186) Reported by: atis Patches:
+ sip_t38_response_415.patch uploaded by atis (license 242)
+
+2009-12-02 15:42 +0000 [r232269] David Vossel <dvossel@digium.com>
+
+ * funcs/func_groupcount.c, /: Merged revisions 232268 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r232268 | dvossel | 2009-12-02 09:41:36 -0600 (Wed, 02
+ Dec 2009) | 9 lines fixes segfault in func_groupcount closes
+ issue #16337) Reported by: Parantido Patches: issue_16337.diff
+ uploaded by dvossel (license 671) Tested by: Parantido, dvossel
+ ........
+
+2009-12-02 14:54 +0000 [r232230] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix a bug where a scheduled item ID would
+ get retained on registrations in a certain scenario causing code
+ to execute during reload that should not. (issue AST-263)
+
+2009-12-02 03:26 +0000 [r232164] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in,
+ include/asterisk/compat.h, main/strcompat.c, configure.ac: So
+ apparently, some platforms don't have ffsll(3). The manpage lies;
+ it says that the function is in POSIX, but that's only for
+ ffs(3), not ffsll(3).
+
+2009-12-02 00:45 +0000 [r232091] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 232090 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01
+ Dec 2009) | 10 lines Do not modify the gain settings on data
+ calls. (The digital flag actually represents a data call.)
+ (closes issue #15972) Reported by: udosw Patches:
+ transcap_digital_fix.diff.txt uploaded by alecdavis (license 585)
+ Tested by: alecdavis ........
+
+2009-12-01 23:56 +0000 [r232008-232017] Russell Bryant <russell@digium.com>
+
+ * main/translate.c: Use __builtin_ffsll() from gcc instead of
+ ffssll() to fix a FreeBSD build error.
+
+ * funcs/func_lock.c: Fix a build error on FreeBSD.
+
+ * /, main/file.c: Merged revisions 232007 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232007 | russell | 2009-12-01 17:25:36 -0600 (Tue, 01 Dec 2009)
+ | 2 lines Fix a warning pointed out by buildbot. ........
+
+2009-12-01 21:54 +0000 [r231927] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 231911 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231911 | jpeeler | 2009-12-01 15:29:31 -0600 (Tue, 01 Dec 2009)
+ | 12 lines Fix crash with invalid frame data The crash was
+ happening as a result of a frame containing an invalid data
+ pointer, but was set with data length of zero. The few times the
+ issue was reproduced it _seemed_ that the frame was queued
+ properly, that is the data pointer was set to NULL. I never could
+ reproduce the crash so as a last resort the crash has been fixed,
+ but a check in __ast_read has been added to give as much
+ information about the source of problematic frames in the future.
+ (closes issue #16058) Reported by: atis ........
+
+2009-12-01 21:20 +0000 [r231867] David Vossel <dvossel@digium.com>
+
+ * main/pbx.c, /: Merged revisions 231853 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231853 | dvossel | 2009-12-01 15:14:31 -0600 (Tue, 01 Dec 2009)
+ | 3 lines WaitExten m option with no parameters generates frame
+ with zero datalen but non-null data ptr ........
+
+2009-12-01 20:27 +0000 [r231814-231850] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_rtp_asterisk.c, channels/chan_unistim.c,
+ main/rtp_engine.c, addons/chan_ooh323.c, channels/chan_sip.c,
+ res/res_adsi.c, addons/chan_ooh323.h,
+ include/asterisk/callerid.h, channels/chan_phone.c,
+ channels/chan_dahdi.c, channels/chan_skinny.c, main/callerid.c,
+ channels/chan_h323.c, addons/ooh323cDriver.c,
+ include/asterisk/rtp_engine.h, addons/ooh323cDriver.h: More
+ 32->64 bit codec conversions. In the process of swapping ULAW to
+ a place in the extended codec space, we found several unhandled
+ cases, where a 32-bit integer was still being used to handle a
+ codec field. Most of these have been fixed with this commit,
+ although there is at least one case (codec_dahdi) which depends
+ upon outside headers to be altered before a conversion can be
+ made. (Fixes AST-278, SWP-459)
+
+ * include/asterisk/mod_format.h: Formats need to be able to
+ represent all 64 codec bits.
+
+2009-12-01 15:47 +0000 [r231741] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/file.c: Merged revisions 231740 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231740 | mnicholson | 2009-12-01 09:34:57 -0600 (Tue, 01 Dec
+ 2009) | 2 lines Ignore unknown formats in ast_format_str_reduce()
+ and return an error if no know formats are found. ........
+
+2009-11-30 21:47 +0000 [r231692] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, channels/chan_sip.c, include/asterisk/udptl.h:
+ Another round of UDPTL stack fixes/improvements: 1) Allow users
+ of UDPTL stack to associate a character-string tag with a UDPTL
+ session, so that log/error/debug messages generated by the UDPTL
+ stack can be 'connected' to the endpoint that caused them to be
+ generated. 2) Improve comments (and process) of calculating the
+ far end's maximum IFP size when redundancy mode is in use for
+ error correction. 3) When an IFP larger than the calculated 'far
+ max IFP' size is presented for writing, truncate it rather than
+ putting in the buffer and allowing the buffer to overflow; this
+ will cause the ends to retrain to a lower bit rate that produces
+ IFPs of an appropriate size if possible, and if not possible, the
+ FAX transfer will fail completely. In these cases, it is due to
+ the one endpoint supplying a T38FaxMaxDatagram value that is
+ improperly calculated and is too low to be of use; we have
+ configuration options available to override this behavior. 4)
+ Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no
+ longer needed.
+
+2009-11-30 21:31 +0000 [r231616-231688] Matthew Nicholson <mnicholson@digium.com>
+
+ * include/asterisk/file.h, /, main/file.c, main/app.c,
+ apps/app_voicemail.c: Merged revisions 231614 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov
+ 2009) | 8 lines Remove duplicate entries from voicemail format
+ lists. This prevents app_voicemail from entering an infinite loop
+ when the same format is specified twice in the format list.
+ (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson
+ Review: https://reviewboard.asterisk.org/r/429/ ........
+
+ * include/asterisk/file.h, /, main/app.c, apps/app_voicemail.c:
+ Reverted 231616
+
+ * include/asterisk/file.h, /, main/app.c, apps/app_voicemail.c:
+ Merged revisions 231614 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov
+ 2009) | 8 lines Remove duplicate entries from voicemail format
+ lists. This prevents app_voicemail from entering an infinite loop
+ when the same format is specified twice in the format list.
+ (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson
+ Review: https://reviewboard.asterisk.org/r/429/ ........
+
+2009-11-30 20:44 +0000 [r231602] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: When receiving SDP that matches the version
+ of the last one do not treat it as a fatal error. (closes issue
+ #16238) Reported by: seandarcy
+
+2009-11-30 18:55 +0000 [r231491-231556] David Vossel <dvossel@digium.com>
+
+ * apps/app_queue.c: app_queue crashes randomly, often during
+ call-transfers This patch adds a ref to the queue_ent object's
+ parent call_queue in queue_exec() so the call_queue won't be
+ destroyed while the the queue_ent still holds a pointer to it.
+ (closes issue 0015686) Tested by: dvossel, aragon
+
+ * res/res_rtp_asterisk.c, /: Merged revisions 231441 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r231441 | dvossel | 2009-11-30 11:14:08 -0600 (Mon, 30
+ Nov 2009) | 11 lines fixes crash caused by RTP comfort noise
+ payload greater than 24 bytes AST-2009-010 (closes issue #16242)
+ Reported by: amorsen Patches: issue16242.diff uploaded by oej
+ (license 306) Tested by: amorsen, oej, dvossel ........
+
+2009-11-30 16:53 +0000 [r231439] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.dynamics (added), Makefile.rules: Export dynamic
+ (weak-linked) symbols correctly. (closes issue #15193) Reported
+ by: eliel Patches: 20091111__issue15193.diff.txt uploaded by
+ tilghman (license 14)
+
+2009-11-30 16:29 +0000 [r231436] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix a bug where an immediate masquerade
+ would cause a queued unhold frame to get lost. Now we just
+ indicate unhold directly after the masquerade is complete. (issue
+ ABE-2011)
+
+2009-11-27 08:47 +0000 [r231401] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c: 1. Modified exported variable names. 2.
+ Added destination port support. 3. Added new protocols. 4. Added
+ QoS.
+
+2009-11-26 02:09 +0000 [r231299-231369] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/CODING-GUIDELINES, main/asterisk.c: Reorder option flags.
+ Change guidelines so that example code is consistent with
+ guidelines
+
+ * main/channel.c, /: Merged revisions 231298 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231298 | tilghman | 2009-11-25 16:31:57 -0600 (Wed, 25 Nov 2009)
+ | 2 lines After a frame duplication failure, unlock the channel
+ before returning. ........
+
+2009-11-25 15:42 +0000 [r231189] Matthew Nicholson <mnicholson@digium.com>
+
+ * pbx/pbx_lua.c: Load pbx_lua with global symbols to allow linking
+ with other lua libraries. Found by Maxim Litnitskiy.
+
+2009-11-24 20:31 +0000 [r231134] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c: Found a few places where queue refcounts were
+ counted incorrectly. Also add debug statements. (closes issue
+ #15982, closes issue #15984) Reported by: atis Patches:
+ 20091111__issue15982.diff.txt uploaded by tilghman (license 14)
+ Tested by: atis
+
+2009-11-24 18:50 +0000 [r231058-231095] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c: Fix erroneous hangup extension execution
+ ast_spawn_extension behaves differently from 1.4 in that hangups
+ and extensions that do not exist do not return an error, whereas
+ in 1.6 it does. This is now taken into account so that the
+ AST_FLAG_BRIDGE_HANGUP_RUN flag gets set properly. (closes issue
+ #16106) Reported by: ajohnson Tested by: ajohnson
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
+ Fix problem on digital channels due to digital flag not getting
+ set Changed areas in sig_pri to set the digital flag using a
+ callback that will also set the corresponding flag in chan_dahdi.
+ Modified dahdi_request slightly so that if a bearer is marked as
+ digital, that information is available when creating the new
+ channel. (closes issue #16151) Reported by: alecdavis Patch based
+ on bug_16151.diff.txt uploaded by alecdavis (license 585)
+
+2009-11-24 13:52 +0000 [r231025] Matthew Nicholson <mnicholson@digium.com>
+
+ * CHANGES: Updated CHANGES file to describe the new 'd' option to
+ app_followme added in r230964 (related to issue #14155) Reported
+ by: junky
+
+2009-11-24 04:58 +0000 [r230994] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/app.h, funcs/func_strings.c, CHANGES: Add
+ REPLACE & PASSTHRU functions, overhaul of func_strings, fix API
+ docs for the ast_get_encoded_* functions. * Add REPLACE function,
+ which searches a given variable for a set of characters and
+ replaces each with a given character. * Add PASSTHRU function,
+ which passes a literal string back, like a NoOp for functions.
+ Intent is to be able to specify a literal string to another
+ function that takes a variable name as an argument. * Let the
+ array manipulation functions work with dialplan functions, in
+ addition to variables. This allows the array manipulation
+ functions to modify ASTDB and ODBC backends, assuming the
+ func_odbc configuration has both read and write functions.
+ (closes issue #15223) Reported by: ajohnson Patches:
+ 20091112__issue15223.diff.txt uploaded by tilghman (license 14)
+ Tested by: lmadsen, tilghman
+
+2009-11-23 22:37 +0000 [r230964] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_followme.c: Add an option to app_followme to disable the
+ "please hold" announcement. (closes issue #14155) Reported by:
+ junky Patches: M14555-trunk.diff uploaded by junky (license 177)
+ (modified) Tested by: junky
+
+2009-11-23 15:45 +0000 [r230881] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: Change fax
+ detection in chan_sip so it behaves as one would expect.
+ Internally the way T.38 is negotiated has changed and the option
+ no longer reflects a behavior that is valid. It will now look for
+ a CNG tone on received calls and if present send the call to the
+ 'fax' extension. It is then up to the application or channel to
+ request the switch over to T.38.
+
+2009-11-23 15:34 +0000 [r230773-230877] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 230839 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r230839 | kpfleming | 2009-11-23 09:09:24 -0600 (Mon, 23 Nov
+ 2009) | 1 line Correct fix for issue #16268... the reporter's
+ original patch was very close to correct. ........
+
+ * /, channels/chan_sip.c: Merged revisions 230772 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r230772 | kpfleming | 2009-11-23 08:13:56 -0600 (Mon, 23 Nov
+ 2009) | 5 lines Ensure that SDP parsing does not ignore the last
+ line of the SDP. (closes issue #16268) Reported by: sgimeno
+ ........
+
+2009-11-20 22:35 +0000 [r230726] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: fixes iax2 show cache locking error, thanks
+ alecdavis! (closes issue #16094) Reported by: alecdavis Patches:
+ bug16094.diff.txt uploaded by alecdavis (license 585) Tested by:
+ alecdavis, dvossel
+
+2009-11-20 21:47 +0000 [r230697] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/unaligned.h: Revert code in error and include
+ the gcc suggested workaround for the original problem, while gcc
+ investigates.
+
+2009-11-20 21:01 +0000 [r230628] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/features.c: Merged revisions 230627 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r230627 | mnicholson | 2009-11-20 14:53:06 -0600 (Fri, 20 Nov
+ 2009) | 8 lines Copy the peer CDR's userfield to the bridge CDR
+ if it exists. This is necessary for the recordagentcalls option
+ in chan_agent to store the recorded file name in the bridge CDR.
+ (closes issue #14590) Reported by: msetim Patches:
+ queue_agent_userfield.patch uploaded by Laureano (license 265)
+ Tested by: Laureano, mnicholson ........
+
+2009-11-20 17:28 +0000 [r230584] David Ruggles <thedavidfactor@gmail.com>
+
+ * doc/externalivr.txt, apps/app_externalivr.c: Fix/Implement error
+ events for non-existing files also include a better cmd define
+ for S command Review: https://reviewboard.asterisk.org/r/430/
+
+2009-11-20 17:26 +0000 [r230509-230583] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/audiohook.h, main/audiohook.c: audiohook signal
+ trigger on every status change (issue #14618) Review:
+ https://reviewboard.asterisk.org/r/434/
+
+ * /, apps/app_mixmonitor.c: Merged revisions 230508 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r230508 | dvossel | 2009-11-19 15:22:46 -0600 (Thu, 19
+ Nov 2009) | 10 lines fixes MixMonitor thread not exiting when
+ StopMixMonitor is used (closes issue #16152) Reported by: AlexMS
+ Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license
+ 671) Tested by: dvossel, AlexMS Review:
+ https://reviewboard.asterisk.org/r/424/ ........
+
+2009-11-19 14:53 +0000 [r230438] David Ruggles <thedavidfactor@gmail.com>
+
+ * apps/app_externalivr.c: Basic cleanup of ExternalIVR: cleaned up
+ argument parsing; implemented good coding practices where
+ applicable; replaced most notice level logging with verbose
+ logging; replaced warning messages that terminated with error
+ messages; fixed memory leak identified by russellb
+
+2009-11-16 16:40 +0000 [r230343-230381] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_fax.c: Fix another buglet in T.38 session teardown at
+ the end of FAX sessions.
+
+ * apps/app_fax.c: Ensure that only one end of a T.38 session
+ initiates teardown at completion.
+
+2009-11-16 01:49 +0000 [r230314] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c: 1. Added SIP Diversion support. 2. Fixed
+ compile warning for UUID.
+
+2009-11-15 17:23 +0000 [r230247] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 230246 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r230246 | kpfleming | 2009-11-15 11:19:06 -0600 (Sun, 15
+ Nov 2009) | 6 lines Correct mistaken option name in error
+ message. The configuration option for allowing hosts to make
+ non-token-based calls is 'calltokenoptional', not
+ 'calltokenignore'. (reported on asterisk-users) ........
+
+2009-11-15 07:53 +0000 [r230217] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/channel.h: Increase maximum length of language
+ buffers (closes issue #16217) Reported by: dsessions
+
+2009-11-13 22:00 +0000 [r230145] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 230144 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r230144 | file | 2009-11-13 16:00:19 -0600 (Fri, 13 Nov 2009) | 8
+ lines Respect the maddr parameter in the Via header. (closes
+ issue #14446) Reported by: frawd Patches: via_maddr.patch
+ uploaded by frawd (license 610) Tested by: frawd ........
+
+2009-11-13 20:42 +0000 [r230111] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_dial.c, channels/chan_sip.c, apps/app_meetme.c,
+ apps/app_fax.c, configs/manager.conf.sample,
+ res/res_musiconhold.c, include/asterisk/manager.h,
+ channels/chan_iax2.c, apps/app_queue.c, CHANGES,
+ res/res_monitor.c, main/cdr.c, main/channel.c, main/manager.c,
+ main/features.c, apps/app_minivm.c, apps/app_chanspy.c,
+ apps/app_voicemail.c: Display a list of channel variables in each
+ channel-oriented event. (Closes AST-33) Reviewboard:
+ https://reviewboard.asterisk.org/r/368/
+
+2009-11-13 19:44 +0000 [r229912-230039] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 230038 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r230038 | file | 2009-11-13 13:44:07 -0600 (Fri, 13 Nov
+ 2009) | 9 lines Fix a crash caused by two threads thinking they
+ should both free the chan_local private structure when only one
+ should. (closes issue #15314) Reported by: sroberts Patches:
+ Issue15314_Move_Nulling_owner.patch uploaded by davidw (license
+ 780) Tested by: davidw, lottc ........
+
+ * UPGRADE.txt, apps/app_chanisavail.c, CHANGES: Store the cause
+ code that is returned when trying to create a channel in
+ ChanIsAvail in the AVAILCAUSECODE dialplan variable instead of
+ overwriting the device state in AVAILSTATUS. (closes issue
+ #14426) Reported by: macli
+
+ * /: Merged revisions 229965 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229965 | file | 2009-11-13 11:19:59 -0600 (Fri, 13 Nov 2009) | 6
+ lines Document a limitation in the AVAILSTATUS variable from
+ ChanIsAvail and provide a workaround for it that does not change
+ existing behavior. (closes issue #14426) Reported by: macli
+ ........
+
+ * channels/chan_sip.c: Fix T.38 negotiation regression introduced
+ with the SDP parser changes.
+
+2009-11-13 10:53 +0000 [r229819-229871] Olle Johansson <oej@edvina.net>
+
+ * main/loader.c: Fixing trunk in a way so that it compiles again.
+ Thanks, Philippe :-)
+
+ * addons/cdr_mysql.c: If CDR logging is disabled, it's considered a
+ FAILURE
+
+ * configs/modules.conf.sample, CHANGES, main/asterisk.c,
+ main/loader.c: Add the capability to require a module to be
+ loaded, or else Asterisk exits. Review:
+ https://reviewboard.asterisk.org/r/426/
+
+2009-11-13 03:16 +0000 [r229788] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c: Added full number portability parameter
+ support.
+
+2009-11-12 23:43 +0000 [r229750-229754] Jason Parker <jparker@digium.com>
+
+ * configs/alsa.conf.sample: Update sample config for ALSA mute and
+ noaudiocapture
+
+ * channels/chan_alsa.c: Add mute functionality. Add config option
+ to not try to open capture device. Adds "console {mute|unmute}"
+ CLI command. Adds mute and noaudiocapture config options (will
+ update sample configs shortly). (closes issue #14673) Reported
+ by: Nick_Lewis Patches: chan_alsa.c-oneway3.patch uploaded by
+ Nick Lewis (license 657) Tested by: qwell
+
+ * channels/chan_oss.c: Fix mute toggling on OSS channels.
+
+2009-11-12 16:44 +0000 [r229670] David Vossel <dvossel@digium.com>
+
+ * funcs/func_audiohookinherit.c, /: Merged revisions 229669 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229669 | dvossel | 2009-11-12 10:41:49 -0600 (Thu, 12 Nov 2009)
+ | 6 lines fixes merging error, datastore was being freed in the
+ wrong function. (closes issue #16219) Reported by: aragon
+ ........
+
+2009-11-12 13:54 +0000 [r229639] Leif Madsen <lmadsen@digium.com>
+
+ * configs/sip.conf.sample: Update sip.conf.sample. Just updating a
+ spelling error and some capitalization in a documentation update
+ that Olle added. May the Swenglish be with you.
+
+2009-11-12 10:24 +0000 [r229606-229607] Olle Johansson <oej@edvina.net>
+
+ * configs/sip.conf.sample: Clarification
+
+ * configs/sip.conf.sample: Clarify some security issues early in
+ the sample configuration
+
+2009-11-11 20:47 +0000 [r229568] David Ruggles <thedavidfactor@gmail.com>
+
+ * doc/externalivr.txt: Remove non-functional feature from
+ ExternalIVR documentation Remove non-functional socket
+ implementation of ExternalIVR from documentation (closes issue
+ #16225) Reported by: thedavidfactor Patches:
+ externalivr.txt.20091111.1542.patch uploaded by thedavidfactor
+ (license 903)
+
+2009-11-11 19:48 +0000 [r229460-229499] David Brooks <dbrooks@digium.com>
+
+ * main/pbx.c, /: Merged revisions 229498 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229498 | dbrooks | 2009-11-11 13:46:19 -0600 (Wed, 11 Nov 2009)
+ | 8 lines Solaris doesn't like NULL going to ast_log Solaris will
+ crash if NULL is passed to ast_log. This simple patch simply uses
+ S_OR to get around this. (closes issue #15392) Reported by:
+ yrashk ........
+
+ * apps/app_softhangup.c: Flags not initialized in app_softhangup.c,
+ causing undefined behavior Trivial patch [kobaz] to initialize an
+ ast_flags = {0} (closes issue #16129) Reported by: kobaz
+
+2009-11-11 14:30 +0000 [r229431] Leif Madsen <lmadsen@digium.com>
+
+ * CHANGES: Update CHANGES file. Updating the CHANGES file after
+ noticing an email on the asterisk-dev mailing list from Russell.
+ (issue #15874)
+
+2009-11-10 22:14 +0000 [r229361] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 229360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229360 | tilghman | 2009-11-10 16:09:16 -0600 (Tue, 10 Nov 2009)
+ | 12 lines If two pattern classes start with the same digit and
+ have the same number of characters, they will compare equal. The
+ example given in the issue report is that of [234] and [246],
+ which have these characteristics, yet they are clearly not
+ equivalent. The code still uses these two characteristics, yet
+ when the two scores compare equal, an additional check will be
+ done to compare all characters within the class to verify
+ equality. (closes issue #15421) Reported by: jsmith Patches:
+ 20091109__issue15421__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: jsmith, thedavidfactor ........
+
+2009-11-10 22:01 +0000 [r229356] David Ruggles <thedavidfactor@gmail.com>
+
+ * doc/externalivr.txt: Merged revisions 229355 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229355 | diruggles | 2009-11-10 16:45:15 -0500 (Tue, 10 Nov
+ 2009) | 9 lines Fix ExternalIVR Documentation Remove
+ documentation for event that doesn't function (closes issue
+ #16220) Reported by: thedavidfactor Patches:
+ externalivr.txt.20091110.1622.patch uploaded by thedavidfactor
+ (license 903) ........
+
+2009-11-10 21:22 +0000 [r229351] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_stack.c: When GOSUB is invoked within an AGI, it may not
+ exit correctly. (closes issue #16216) Reported by: atis Patches:
+ 20091110__atis_work.diff.txt uploaded by tilghman (license 14)
+ Tested by: atis
+
+2009-11-10 20:06 +0000 [r229282] Joshua Colp <jcolp@digium.com>
+
+ * /, codecs/codec_g726.c: Merged revisions 229281 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229281 | file | 2009-11-10 16:03:14 -0400 (Tue, 10 Nov 2009) | 8
+ lines Remove broken support for direct transcoding between G.726
+ RFC3551 and G.726 AAL2. On some systems the translation core
+ would actually consider g726aal2 -> g726 -> signed linear to be a
+ quicker path then g726aal2 -> signed linear which exposed this
+ problem. (closes issue #15504) Reported by: globalnetinc ........
+
+2009-11-10 17:33 +0000 [r229228] David Ruggles <thedavidfactor@gmail.com>
+
+ * /, doc/externalivr.txt: Merged revisions 229191 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229191 | diruggles | 2009-11-10 12:23:59 -0500 (Tue, 10 Nov
+ 2009) | 11 lines Document ExternalIVR event tag collision
+ ExternalIVR uses the D tag for two different event types. This
+ documents that behavior and how to differentiate between the two
+ cases. Also includes a minor spelling fix and clarification
+ (closes issue #16211) Reported by: thedavidfactor Patches:
+ externalivr.txt.20091109.1507.patch uploaded by thedavidfactor
+ (license 903) ........
+
+2009-11-10 17:16 +0000 [r229168] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 229167 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r229167 | dvossel | 2009-11-10 11:15:57 -0600 (Tue, 10
+ Nov 2009) | 9 lines don't crash on log message in solaris
+ AST-2009-006 (closes issue #16206) Reported by: bklang Tested by:
+ bklang ........
+
+2009-11-10 15:53 +0000 [r229102] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Reverted revision 201717. (closes issue
+ 0016175) Reported by: paul-tg
+
+2009-11-10 15:27 +0000 [r229093] David Vossel <dvossel@digium.com>
+
+ * res/res_config_pgsql.c: fixes pgsql double free of threadstorage
+ A thread storage variable was being freed incorrectly, which
+ resulted in a double free if two queries were made in the same
+ thread. (closes issue #16011) Reported by: cristiandimache
+ Patches: issue16011.diff uploaded by dvossel (license 671)
+
+2009-11-10 11:16 +0000 [r229050] Gavin Henry <ghenry@suretecsystems.com>
+
+ * contrib/scripts/asterisk.ldap-schema: Schema file additions *
+ Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox
+ objectClasses to allow standalone dialplan, account and mailbox
+ entries (STRUCTURAL) * Added new Fields: - AstAccountLanguage,
+ AstAccountTransport, AstAccountPromiscRedir, -
+ AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
+ - AstAccountVideoSupport, AstAccountIgnoreSDPVersion * Removed
+ redundant IPaddr (there's already IPAddress) - Gives more
+ configuration Flags for SIP-Users available (tested) - Allows to
+ create Asterisk Attributes in defined Asterisk ObjectClasses
+ without extensibleObject (which really should be the last
+ resort); gives also additional possibilities for LDAP-filter
+ (closes issue #15874) Reported by: Medozas Patches:
+ asterisk.ldap-schema.patch uploaded by Medozas (license 41)
+ Tested by: Medozas, suretec
+
+2009-11-09 22:50 +0000 [r229015] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c: Don't crash when bridge->tech_pvt == NULL
+ This is a similar solution to what is in place for chan_agent
+ (closes issue #16003) Reported by: atis Tested by: twilson
+
+2009-11-09 17:17 +0000 [r228979] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/iax2-parser.c: Don't try to convert a 64-bit integer,
+ where only a 32-bit integer is stored. (closes issue #16194)
+ Reported by: habile
+
+2009-11-09 16:28 +0000 [r228947] Matthew Nicholson <mnicholson@digium.com>
+
+ * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add the
+ 'relative-periodic-announce' option to app_queue to allow for
+ calculating the time of announcments from the end of the previous
+ announcment rather than from the beginning. (closes issue #15260)
+ Reported by: tonils
+
+2009-11-09 15:38 +0000 [r228897] Leif Madsen <lmadsen@digium.com>
+
+ * main/channel.c, /: Merged revisions 228896 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228896 | lmadsen | 2009-11-09 09:37:43 -0600 (Mon, 09 Nov 2009)
+ | 6 lines Update WARNING message. Update a WARNING message to
+ give a suggested fix when encountered. (closes issue #16198)
+ Reported by: atis Tested by: atis ........
+
+2009-11-09 14:37 +0000 [r228858] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 228827 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r228827 | mnicholson | 2009-11-09 08:16:03 -0600 (Mon,
+ 09 Nov 2009) | 8 lines Perform limited bounds checking when
+ destroying ast_mutex_t structures to make sure we don't try to
+ use negative indices. (closes issue #15588) Reported by: zerohalo
+ Patches: 20090820__issue15588.diff.txt uploaded by tilghman
+ (license 14) Tested by: zerohalo ........
+
+2009-11-09 07:37 +0000 [r228798] Tilghman Lesher <tlesher@digium.com>
+
+ * addons/cdr_mysql.c, main/event.c, channels/chan_console.c,
+ res/res_pktccops.c, main/loader.c: Fix various problems detected
+ with Valgrind. * chan_console accessed pvts after deallocation. *
+ cdr_mysql stored a pointer that was freed by realloc() * The
+ module loader did not check usecount on shutdown, which led to
+ chan_iax2 reading a timer that was already unloaded. * The event
+ subsystem sometimes creates an event with no IEs. Due to a corner
+ condition, the code would read beyond the memory boundary. *
+ res_pktccops did not correctly check whether its monitor thread
+ was started. (closes issue #16062) Reported by: alexanderheinz
+ Patches: 20091109__issue16062.diff.txt uploaded by tilghman
+ (license 14) Tested by: tilghman
+
+2009-11-07 17:02 +0000 [r228766] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * contrib/init.d/rc.debian.asterisk: Add LSB headers to the Debian
+ init.d script See also issue #14864 .
+
+2009-11-06 22:35 +0000 [r228693] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /: Merged revisions 228692 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228692 | dvossel | 2009-11-06 16:33:27 -0600 (Fri, 06 Nov 2009)
+ | 9 lines fixes audiohook write crash occuring in chan_spy
+ whisper mode. After writing to the audiohook list in ast_write(),
+ frames were being freed incorrectly. Under certain conditions
+ this resulted in a double free crash. (closes issue #16133)
+ Reported by: wetwired (closes issue #16045) Reported by:
+ bluecrow76 Patches: issue16045.diff uploaded by dvossel (license
+ 671) Tested by: bluecrow76, dvossel, habile ........
+
+2009-11-06 22:32 +0000 [r228691] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, CHANGES, channels/sig_pri.c: Created
+ standard location to add options to chan_dahdi for ISDN dialing.
+ Dial(DAHDI/g1[/extension[/options]]) Current options:
+ K(<keypad_digits>) R Reverse charging indication (Collect calls)
+ The earlier Dial(DAHDI/g1[/K<keypad_digits>][/extension] format
+ was variable and did not allow for the easy addition of more
+ options. The earlier 'C' prefix character for reverse charge
+ indiation would conflict with the a-d DTMF digits if ISDN uses
+ them.
+
+2009-11-06 22:07 +0000 [r228661] David Brooks <dbrooks@digium.com>
+
+ * tests/test_amihooks.c: ami_testhooks.c automatically registers
+ hook ami_testhooks.c was registering for AMI events upon module
+ load. Moved the registration to its own CLI command. Added CLI
+ command for unregistering the hook. Changed some of the wording,
+ removed unnecessary arguments/parameters. Reported by: rmudgett
+
+2009-11-06 22:02 +0000 [r228658-228659] Mark Michelson <mmichelson@digium.com>
+
+ * addons/chan_ooh323.c: Make compilation of chan_ooh323 disabled by
+ default. All addons modules should be disabled by default,
+ requiring the user to turn them on if desired. After all, these
+ are addons we're talking about here.
+
+ * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooh245.c: Get
+ chan_ooh323 to compile with gcc 4.2. For some reason, the code
+ compiles just fine with later versions of GCC, but this one
+ requires some weird double casting in order to get rid of all
+ warnings. Whatever.
+
+2009-11-06 19:53 +0000 [r228621] Richard Mudgett <rmudgett@digium.com>
+
+ * main/frame.c: Fix compiler warning gcc 4.2.4 found
+
+2009-11-06 19:47 +0000 [r228620] Matthew Nicholson <mnicholson@digium.com>
+
+ * funcs/func_base64.c, /, main/utils.c: Merged revisions 228378 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228378 | mnicholson | 2009-11-06 10:26:59 -0600 (Fri, 06 Nov
+ 2009) | 8 lines Properly handle '=' while decoding base64
+ messages and null terminate strings returned from BASE64_DECODE.
+ (closes issue #15271) Reported by: chappell Patches:
+ base64_fix.patch uploaded by chappell (license 8) Tested by:
+ kobaz ........
+
+2009-11-06 19:38 +0000 [r228616] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_nbs.c, addons/chan_mobile.c: Missed these two
+ channel drivers on the codec_bits merge
+
+2009-11-06 18:37 +0000 [r228499-228548] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 228547 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228547 | file | 2009-11-06 14:32:58 -0400 (Fri, 06 Nov 2009) | 4
+ lines Don't overwrite caller ID name on a trunk with the
+ configured fullname when using users.conf (issue ABE-1989)
+ ........
+
+ * doc/tex/localchannel.tex: Fix the localchannel.tex file.
+
+2009-11-06 17:22 +0000 [r228420-228441] David Vossel <dvossel@digium.com>
+
+ * codecs/codec_ilbc.c: Fixes merging issue from 1.4, frame data is
+ held in data.ptr in trunk
+
+ * /, codecs/codec_ilbc.c: Merged revisions 228418 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228418 | dvossel | 2009-11-06 11:07:13 -0600 (Fri, 06 Nov 2009)
+ | 13 lines fixes segfault in iLBC For reasons not yet known, it
+ appears possible for an ast_frame to have a datalen greater than
+ zero while the actual data is NULL during Packet Loss
+ Concealment. Most codecs don't support PLC so this doesn't affect
+ them. This patch catches the malformed frame and prevents the
+ crash from occuring. Additional efforts to determine why it is
+ possible for a frame to look like this are still being
+ investigated. (issue #16979) ........
+
+2009-11-06 16:42 +0000 [r228410] Joshua Colp <jcolp@digium.com>
+
+ * /, main/abstract_jb.c: Merged revisions 228409 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228409 | file | 2009-11-06 12:41:20 -0400 (Fri, 06 Nov 2009) | 7
+ lines Fix a bug caused by a partially invalid frame (from the
+ jitterbuffer) passing through the Asterisk core. (closes issue
+ #15560) Reported by: jvandal (closes issue #15709) Reported by:
+ covici ........
+
+2009-11-06 15:42 +0000 [r228268-228339] David Vossel <dvossel@digium.com>
+
+ * /, main/astfd.c: Merged revisions 228338 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228338 | dvossel | 2009-11-06 09:41:41 -0600 (Fri, 06 Nov 2009)
+ | 5 lines fixes crash in astfd.c (closes issue #15981) Reported
+ by: slavon ........
+
+ * funcs/func_audiohookinherit.c: fixes memory leak in
+ func_audiohookinherit.c (closes issue #15394) Reported by: boroda
+ Patches: bug15394_memoryleak_diff2.txt uploaded by dbrooks
+ (license 790) Tested by: dbrooks, boroda
+
+2009-11-05 22:59 +0000 [r228233] Mark Michelson <mmichelson@digium.com>
+
+ * funcs/func_cdr.c: Fix XML in func_cdr.c
+
+2009-11-05 22:12 +0000 [r228191-228196] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c: Yet another error message in the dialplan
+ (thanks, rmudgett/russellb)
+
+ * apps/app_meetme.c: MEETME_INFO should not return a literal error
+ message to the dialplan. (closes issue #15450) Reported by:
+ JimVanM Patches: meetmeinfopatch.diff.txt uploaded by dbrooks
+ (license 790) Tested by: JimVanM
+
+2009-11-05 21:23 +0000 [r228189] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_chanspy.c: Fix the fix for chanspy option o In 224178, I
+ assumed the uploaded patch was correct as it had received
+ positive feedback. The flags were being checked in the incorrect
+ location. Upon testing the fix this time it was also found that
+ the flags from the dialplan weren't being copied to the
+ chanspy_translation_helper. (closes issue #16167) Reported by:
+ marhbere
+
+2009-11-05 19:34 +0000 [r228145] David Brooks <dbrooks@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 228078 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05
+ Nov 2009) | 9 lines chan_misdn Asterisk 1.4.27-rc2 crash Crash
+ related to chan_misdn connection. Patch submitted by
+ gknispel_proformatique, tested by francesco_r. "I have many crash
+ since i have upgraded to Asterisk 1.4.27-rc2. Attached a full
+ bt." This patch zeros out an ast_frame. (closes issue #16041)
+ Reported by: francesco_r ........
+
+2009-11-05 19:16 +0000 [r228080] Jason Parker <jparker@digium.com>
+
+ * channels/chan_vpb.cc, /: Merged revisions 228079 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r228079 | qwell | 2009-11-05 13:14:25 -0600 (Thu, 05 Nov
+ 2009) | 8 lines Fix crash on VPB exception when no hardware is
+ present. (closes issue #14970) Reported by: tzafrir Patches:
+ vpb_exception.diff uploaded by tzafrir (license 46) Tested by:
+ markwaters ........
+
+2009-11-05 17:26 +0000 [r228015-228049] Tilghman Lesher <tlesher@digium.com>
+
+ * main/frame.c: Rework codecs command to comply with the 64-bit
+ scheme
+
+ * apps/app_externalivr.c: Don't crash if no arguments are passed.
+ (closes issue #16119) Reported by: thedavidfactor
+
+2009-11-04 23:50 +0000 [r227914-227945] Jeff Peeler <jpeeler@digium.com>
+
+ * /, res/res_monitor.c: Merged revisions 227944 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227944 | jpeeler | 2009-11-04 17:47:08 -0600 (Wed, 04 Nov 2009)
+ | 14 lines Fix incorrect filename comparsion after monitor file
+ change The logic to detect if a requested file is indeed a
+ different file from the current file was incorrect. The main
+ issue being confusion of the use of filename_base which was
+ previously set without pathing information and then compared to
+ another full path. Robust file comparison logic has been added to
+ properly check if two files are the same even if symlinks are
+ used. (closes issue #15313) Reported by: caspy Patches:
+ 20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license
+ 325) but mostly tilghman's work ........
+
+ * addons/chan_ooh323.c: Update chan_ooh323 to support the expanded
+ codec bitfield from 227580.
+
+2009-11-04 22:10 +0000 [r227898] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/oochannels.h,
+ addons/ooh323c/src/ooCmdChannel.h, addons/chan_ooh323.c,
+ addons/ooh323c/src/printHandler.h, addons/ooh323c/src/ooq931.h,
+ addons/ooh323c/src/ootrace.h, addons/chan_ooh323.h,
+ addons/ooh323c/src/ooasn1.h, addons/ooh323c/src/ootypes.h,
+ addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooStackCmds.c,
+ addons/ooh323c/src/errmgmt.c, addons/ooh323c/src/ooTimer.c,
+ addons/ooh323c/src/ooLogChan.h,
+ addons/ooh323c/src/ooCapability.c,
+ addons/ooh323c/src/ooStackCmds.h, addons/ooh323c/src/dlist.c,
+ addons/ooh323c/src/eventHandler.c,
+ addons/ooh323c/src/ooCapability.h,
+ addons/ooh323c/src/eventHandler.h, addons/Makefile,
+ addons/ooh323cDriver.c, addons/ooh323c/src/ooDateTime.c,
+ addons/ooh323c/src/rtctype.c, addons/ooh323cDriver.h,
+ addons/ooh323c/src/ooCalls.c, addons/ooh323c/src/encode.c,
+ addons/ooh323c/src/ooUtils.c, addons/ooh323c/src/ooGkClient.c,
+ addons/ooh323c/src/ooDateTime.h, addons/ooh323c/src/ooCalls.h,
+ addons/ooh323c/src/ooh323ep.c, addons/ooh323c/src/ooGkClient.h,
+ addons/ooh323c/src/ooports.c, addons/ooh323c/src/ooh323ep.h,
+ addons/ooh323c/src/memheap.c, addons/ooh323c/src/ooh323.c,
+ addons/ooh323c/src/h323/H323-MESSAGESDec.c,
+ addons/ooh323c/src/ooh245.c, addons/ooh323c/src/memheap.h,
+ addons/ooh323c/src/ooh323.h, addons/ooh323c/src/decode.c,
+ addons/ooh323c/src/context.c, addons/ooh323c/src/perutil.c,
+ addons/ooh323c/src/h323/MULTIMEDIA-SYSTEM-CONTROLDec.c,
+ addons/ooh323c/src/ooh245.h, addons/ooh323c/src/ooSocket.c,
+ addons/ooh323c/src/h323/H235-SECURITY-MESSAGESDec.c,
+ addons/ooh323c/src/oochannels.c,
+ addons/ooh323c/src/ooCmdChannel.c,
+ addons/ooh323c/src/printHandler.c, addons/ooh323c/src/ooSocket.h,
+ addons/ooh323c/src/ooCommon.h, addons/ooh323c/src/ooq931.c,
+ addons/ooh323c/src/ootrace.c: Reworked chan_ooh323 channel
+ module. Many architectural and functional changes. Main changes
+ are threading model chanes (many thread in ooh323 stack instead
+ of one), modifications and improvements in signalling part,
+ additional codecs support (726, speex), t38 mode support. This
+ module tested and used in production environment. (closes issue
+ #15285) Reported by: may213 Tested by: sles, c0w, OrNix Review:
+ https://reviewboard.asterisk.org/r/324/
+
+2009-11-04 21:39 +0000 [r227829-227897] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_dial.c, CHANGES: Added the 'a' option to app dial and
+ modified app_dial to set the answertime when the called channel
+ answers. This change causes answertime to be correct even if the
+ called channel hangs up during an announcement triggered by the
+ A() option. (closes issue #15936) Reported by: falves11 Patches:
+ dial-macro-billsec-fix1.diff uploaded by mnicholson (license 96)
+ dial-caller-answer1.diff uploaded by mnicholson (license 96)
+ Tested by: falves11, mnicholson
+
+ * apps/app_dial.c, /: Merged revisions 227827 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov
+ 2009) | 10 lines This patch modifies the Dial application to
+ monitor the calling channel for hangups while playing back
+ announcements. (closes issue #16005) Reported by: falves11
+ Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson
+ (license 96) Tested by: mnicholson, falves11 Review:
+ https://reviewboard.asterisk.org/r/407/ ........
+
+2009-11-04 20:35 +0000 [r227824] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/unaligned.h: Fixes for gcc 4.4
+
+2009-11-04 20:13 +0000 [r227759] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Modify the SDP parsing code to parse session
+ and media level items separately. With the new code, media level
+ proprieties should no longer be confused with session level
+ proprieties. This change also reorganizes some of the SDP parsing
+ code which should make it easier to manage in the future. (closes
+ issue #14994) Reported by: frawd Tested by: frawd, mnicholson,
+ file Review: https://reviewboard.asterisk.org/r/414/
+
+2009-11-04 19:26 +0000 [r227712-227739] Joshua Colp <jcolp@digium.com>
+
+ * /, static-http/prototype.js: Merged revisions 227735 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r227735 | file | 2009-11-04 15:25:37 -0400 (Wed, 04 Nov
+ 2009) | 5 lines Fix a security issue where it may be possible for
+ someone to execute a cross-site AJAX request exploit.
+ (AST-2009-009) ........
+
+ * /, channels/chan_sip.c: Merged revisions 227700 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227700 | file | 2009-11-04 15:17:39 -0400 (Wed, 04 Nov 2009) | 5
+ lines Fix a security issue where sending a REGISTER with a
+ differing username in the From URI and Authorization header would
+ reveal whether it was valid or not. (AST-2009-008) ........
+
+2009-11-04 16:41 +0000 [r227646] Mark Michelson <mmichelson@digium.com>
+
+ * main/frame.c: Add a couple more casts so that code compiles
+ correctly.
+
+2009-11-04 16:35 +0000 [r227645] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/pbx.h: mmichelson reported a compilation error
+ related to codec bit expansion that should be resolved with a
+ simple include of frame_defs.h
+
+2009-11-04 16:25 +0000 [r227643] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: fix trunk building
+
+2009-11-04 16:17 +0000 [r227579-227615] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c, channels/chan_iax2.c: Two other trunk build
+ fixes (reported by seanbright on #asterisk-dev)
+
+ * addons/format_mp3.c: Fix trunk building
+
+ * main/udptl.c, main/autoservice.c, apps/app_dahdibarge.c,
+ main/frame.c, channels/chan_local.c, main/rtp_engine.c,
+ include/asterisk/autoconfig.h.in, apps/app_record.c,
+ apps/app_test.c, bridges/bridge_softmix.c,
+ apps/app_alarmreceiver.c, codecs/ex_alaw.h, codecs/ex_adpcm.h,
+ formats/format_wav_gsm.c, formats/format_sln16.c,
+ codecs/ex_gsm.h, channels/chan_iax2.c, main/indications.c,
+ res/res_rtp_multicast.c, channels/chan_dahdi.c,
+ include/asterisk/bridging_technology.h, pbx/pbx_spool.c,
+ channels/sig_analog.c, include/asterisk/audiohook.h,
+ channels/chan_skinny.c, configure, main/strcompat.c,
+ include/asterisk/compat.h, formats/format_pcm.c, main/features.c,
+ channels/chan_alsa.c, apps/app_amd.c, formats/format_h263.c,
+ apps/app_url.c, apps/app_externalivr.c, formats/format_jpeg.c,
+ main/bridging.c, codecs/ex_ulaw.h, apps/app_milliwatt.c,
+ formats/format_gsm.c, apps/app_dial.c, main/pbx.c,
+ formats/format_wav.c, channels/chan_bridge.c, apps/app_echo.c,
+ apps/app_fax.c, include/asterisk/slin.h, channels/chan_agent.c,
+ configure.ac, formats/format_ogg_vorbis.c, apps/app_disa.c,
+ include/asterisk/unaligned.h, codecs/ex_speex.h,
+ include/asterisk/channel.h, apps/app_talkdetect.c,
+ channels/iax2-parser.c, apps/app_speech_utils.c,
+ channels/iax2-parser.h, channels/chan_misdn.c,
+ apps/app_waitforring.c, channels/iax2.h, codecs/codec_dahdi.c,
+ main/audiohook.c, apps/app_chanspy.c, formats/format_g726.c,
+ include/asterisk/frame_defs.h (added),
+ include/asterisk/translate.h, include/asterisk/slinfactory.h,
+ channels/chan_unistim.c, channels/chan_vpb.cc,
+ channels/chan_multicast_rtp.c, formats/format_sln.c,
+ apps/app_meetme.c, apps/app_dictate.c, codecs/ex_g722.h,
+ codecs/ex_g726.h, channels/chan_gtalk.c, res/res_musiconhold.c,
+ apps/app_followme.c, formats/format_siren7.c,
+ include/asterisk/abstract_jb.h, main/asterisk.exports,
+ main/channel.c, formats/format_ilbc.c, channels/chan_phone.c,
+ main/dial.c, main/manager.c, funcs/func_volume.c, res/res_agi.c,
+ apps/app_mp3.c, main/app.c, doc/codec-64bit.txt (added),
+ formats/format_h264.c, include/asterisk/rtp_engine.h,
+ include/asterisk/frame.h, formats/format_siren14.c,
+ codecs/ex_ilbc.h, channels/chan_mgcp.c, apps/app_jack.c,
+ res/res_rtp_asterisk.c, apps/app_nbscat.c, channels/chan_sip.c,
+ codecs/ex_lpc10.h, apps/app_festival.c, main/slinfactory.c,
+ main/translate.c, res/res_adsi.c, channels/chan_console.c,
+ channels/h323/chan_h323.h, channels/sig_pri.c, apps/app_queue.c,
+ channels/chan_oss.c, channels/chan_jingle.c,
+ formats/format_vox.c, include/asterisk/bridging.h,
+ main/abstract_jb.c, main/file.c, channels/chan_h323.c,
+ formats/format_g723.c, codecs/codec_ulaw.c, apps/app_sms.c,
+ include/asterisk/pbx.h, main/dsp.c, formats/format_g729.c: Expand
+ codec bitfield from 32 bits to 64 bits. Reviewboard:
+ https://reviewboard.asterisk.org/r/416/
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac:
+ chan_misdn will fail to compile if the redirect_dn member is
+ missing
+
+2009-11-04 08:22 +0000 [r227545] Olle Johansson <oej@edvina.net>
+
+ * main/manager.c: Add destruction of iterators to avoid problems
+ with refcounters (per Russell's review of another patch)
+
+2009-11-04 03:15 +0000 [r227509] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c: Don't crash when state_interface is NULL.
+
+2009-11-03 22:13 +0000 [r227462-227464] Russell Bryant <russell@digium.com>
+
+ * res/res_pktccops.c: Resolve another warning.
+
+ * main/manager.c, pbx/pbx_config.c: Resolve a warning from gcc
+ 4.4.1.
+
+ * channels/chan_mgcp.c: Resolve some dev-mode warnings.
+
+2009-11-03 21:26 +0000 [r227448] David Brooks <dbrooks@digium.com>
+
+ * main/manager.c, include/asterisk/manager.h, tests/test_amihooks.c
+ (added): AMI hook interface This patch, originally submitted by
+ jozza, enables custom modules to send actions to AMI and receive
+ messages from AMI via a hook interface. Included is a simple test
+ module to illustrate the interface. (closes issue #14635)
+ Reported by: jozza Review:
+ https://reviewboard.asterisk.org/r/412/
+
+2009-11-03 21:21 +0000 [r227435] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/cdr.c, apps/app_forkcdr.c, configs/cdr_custom.conf.sample,
+ funcs/func_cdr.c, main/features.c, include/asterisk/cdr.h,
+ CHANGES: This patch adds a sequence field to CDRs that can be
+ combined with the linkedid or uniqueid field to uniquely identify
+ a CDR. (closes issue #15180) Reported by: Nick_Lewis Patches:
+ cdr-sequence10.diff uploaded by mnicholson (license 96) Tested
+ by: mnicholson
+
+2009-11-03 21:16 +0000 [r227424] Joshua Colp <jcolp@digium.com>
+
+ * configs/queues.conf.sample, apps/app_queue.c: Add support for
+ using a hint when configuring a state interface using the format
+ hint:<extension>@<context>. (closes issue #15168) Reported by:
+ p_lindheimer Patches: queue_extenstate5_1.4.svn.patch uploaded by
+ GameGamer43 (license 894)
+
+2009-11-03 19:59 +0000 [r227372] Jason Parker <jparker@digium.com>
+
+ * Makefile, main/Makefile: Fix some build issues on Solaris.
+ (closes issue #14517) (SWP-109) Reported by: asgaroth Patches:
+ bug_14517.diff uploaded by snuffy (license 35) Tested by:
+ asgaroth, snuffy, dougm, qwell
+
+2009-11-03 19:48 +0000 [r227361-227368] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_controlplayback.c: Change warning message to debug
+ message. app_controlplayback outputs a warning, when in fact it
+ is normal. (closes issue #16071) Reported by: atis Patches:
+ controlplayback_warning.patch uploaded by atis (license 242)
+
+ * configs/extensions.conf.sample: Additional fixes to the
+ extensions.conf.sample file. Update the extensions.conf.sample
+ [stdexten] context so that we use the variable instead of
+ requiring it to be passed explicitly. Also updated uses of the
+ [stdexten] context throughout. (closes issue #15858) Reported by:
+ pprindeville Patches: stdexten-context-update.txt uploaded by
+ lmadsen (license 10) Tested by: pprindeville
+
+2009-11-03 18:22 +0000 [r227298] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Fixed a spelling error in the q850 reason
+ header option in the output of sip show settings.
+
+2009-11-03 17:58 +0000 [r227277] Richard Mudgett <rmudgett@digium.com>
+
+ * /: Recorded merge of revisions 227275 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227275 | rmudgett | 2009-11-03 11:55:47 -0600 (Tue, 03 Nov 2009)
+ | 4 lines Make sure the outgoing flag is cleared if a new channel
+ fails to get created for outgoing calls. This is the relevant
+ portion of asterisk/trunk -r226648 ........
+
+2009-11-03 17:56 +0000 [r227276] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_mgcp.c: Code guidelines fixes only
+
+2009-11-03 17:12 +0000 [r227238] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: user.conf entries in SIP were not having
+ their peer type set. (closes issue #16120) Reported by: jsmith
+
+2009-11-03 16:56 +0000 [r227237] Olle Johansson <oej@edvina.net>
+
+ * funcs/func_speex.c: Adding some clarifications to func_speex
+ doxygen docs. The functions needed doesn't exist in Speex 1.05
+ which is what a lot of distros use. 1.2 seems to have been in
+ beta status for years, and does include the sexy functions needed
+ for func_speex to work.
+
+2009-11-03 15:37 +0000 [r227167] Joshua Colp <jcolp@digium.com>
+
+ * /: Merged revisions 227166 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227166 | file | 2009-11-03 11:36:16 -0400 (Tue, 03 Nov 2009) | 5
+ lines Fix a bug where an RPID header could be generated with a
+ blank username in the URI. (closes issue #15909) Reported by:
+ kobaz ........
+
+2009-11-03 15:19 +0000 [r227162] Leif Madsen <lmadsen@digium.com>
+
+ * configs/extensions.conf.sample: Update extensions.conf.sample
+ file to fix incorrect extensions. (closes issue #15857) Reported
+ by: pprindeville Patches: stdexten.patch#2 uploaded by
+ pprindeville (license 347) Tested by: pprindeville
+
+2009-11-03 11:11 +0000 [r227091] Olle Johansson <oej@edvina.net>
+
+ * Makefile, /, channels/chan_sip.c: Merged revisions 227088 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7
+ lines Use proper response code when violating Contact ACL's.
+ https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a
+ quick review. (EDVX-003) ........
+
+2009-11-02 22:29 +0000 [r227049] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/mgcp.conf.sample, include/asterisk/pktccops.h (added),
+ CHANGES, res/res_pktccops.c (added), channels/chan_mgcp.c,
+ configs/res_pktccops.conf.sample (added): Add PacketCable NCS 1.0
+ support for Docsis/Eurodocsis networks (closes issue #12950)
+ Reported by: alea-soluciones Patches:
+ ncs-pktccops-12950-r206803.patch uploaded by alea-soluciones
+ (license 514) Tested by: alea-soluciones, adomjan, urtho,
+ nahuelgreco
+
+2009-11-02 20:59 +0000 [r226973-226974] David Brooks <dbrooks@digium.com>
+
+ * channels/chan_sip.c: SIP channel name uniqueness SIP channel
+ names were supposed to be unique by way of a name suffix derived
+ from the pointer to the channel's private data. Uniqueness was
+ preserved on 32-bit systems, but not on 64-bit systems. This
+ patch, as suggested by kpfleming, replaces this suffix with a
+ simple incremented unsigned int. (closes issue #15152) Reported
+ by: palbrecht Review: https://reviewboard.asterisk.org/r/420/
+
+ * /: SIP channel name uniqueness SIP channel names were supposed to
+ be unique by way of a name suffix derived from the pointer to the
+ channel's private data. Uniqueness was preserved on 32-bit
+ systems, but not on 64-bit systems. This patch, as suggested by
+ kpfleming, replaces this suffix with a simple incremented
+ unsigned int. (closes issue #15152) Reported by: palbrecht
+ Review: https://reviewboard.asterisk.org/r/420/
+
+2009-11-02 20:43 +0000 [r226970] Olle Johansson <oej@edvina.net>
+
+ * main/http.c: Adding external reference for doxygen
+
+2009-11-02 18:08 +0000 [r226890] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 226889 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) |
+ 11 lines Fix a bug where the recorded privacy introduction file
+ would not get removed if the caller hung up while the called
+ party had not yet answered. This was fixed by introducing an
+ argument to the 'n' option which, when enabled, removes the
+ introduction file under all scenarios. This was done to preserve
+ the behavior that has existed for quite some time. (closes issue
+ #14674) Reported by: ulogic Patches: bug14674.patch uploaded by
+ jpeeler (license 325) ........
+
+2009-11-02 17:34 +0000 [r226882] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, UPGRADE.txt,
+ channels/sig_pri.c: DAHDI ISDN channel names will not allow
+ device state to work. (Interim solution.) Since ISDN works like
+ SIP and not analog ports in regard to devices, the device state
+ based on the ISDN channel number could not work. This has not
+ been an issue until the advent of PTMP NT mode. Previously, ISDN
+ lines were used as trunks and did not have to keep track of
+ specific devices. As an interim solution until device states are
+ properly implemented, the channel name is being changed to the
+ following format to use the generic device state support:
+ DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number> Dialplan
+ hints would thus be: exten => xxx,hint,DAHDI/i2/5551212 This will
+ work with the following restrictions: * The number of
+ devices/phones cannot exceed the number of B channels. (i.e., BRI
+ has 2) * Each device/phone can only have one number. No shared
+ MSN's. * The phones/devices probably should not use
+ subaddressing.
+
+2009-11-02 17:15 +0000 [r226812] Tilghman Lesher <tlesher@digium.com>
+
+ * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 226811 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226811 | tilghman | 2009-11-02 11:14:20 -0600 (Mon, 02 Nov 2009)
+ | 8 lines Don't allow two separate instances of safe_asterisk
+ when restarting from the init script. (closes issue #14562)
+ Reported by: davidw Patches: Initially
+ 20091022__issue14562.diff.txt uploaded by tilghman (license 14)
+ Modified to 20091030__Issue14562_diff.txt uploaded by davidw
+ (license 780) Tested by: davidw ........
+
+2009-11-02 14:57 +0000 [r226687] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: This patch
+ adds support for a draft proposal for adding Q.850 reason headers
+ to sip messages. (closes issue #13385) Reported by: adomjan
+ Patches: sip.conf.sample-trunk20090929-reason_q850.patch uploaded
+ by adomjan (license 487) CHANGES-trunk20090929-reason_q850.patch
+ uploaded by adomjan (license 487)
+ chan_sip.c-trunk20090929-reason_q850_atoi_fix.patch uploaded by
+ adomjan (license 487) sip-q850-hangupcause1.diff uploaded by
+ mnicholson (license 96) Tested by: adomjan
+
+2009-10-30 23:26 +0000 [r226648] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_pri.c: Cleanup some flags on
+ DAHDI PRI channel hangup. * Cleanup some flags on DAHDI PRI
+ channel hangup. (sig_pri split) * Make sure the outgoing flag is
+ cleared if a new channel fails to get created for outgoing calls.
+ * Remove some unused flags since sig_pri was split.
+
+2009-10-30 04:08 +0000 [r226606] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/doxygen/architecture.h (added),
+ res/res_rtp_asterisk.c, res/res_rtp_multicast.c,
+ include/asterisk/doxyref.h, contrib/asterisk-ng-doxygen,
+ main/asterisk.c: Add an "Asterisk Architecture Overview" section
+ to the doxygen documentation. This is a side project I've been
+ poking at this week. The intent is to discuss Asterisk
+ architecture in a top down fashion to help new developers
+ understand how Asterisk is put together. There is a ton of stuff
+ to write about, so this will just continue to evolve over time.
+
+2009-10-29 18:13 +0000 [r226532] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_local.c, /, doc/tex/localchannel.tex: Merged
+ revisions 226531 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6
+ lines Add an option to enabling passing music on hold start and
+ stop requests through instead of acting on them in chan_local.
+ (closes issue #14709) Reported by: dimas ........
+
+2009-10-29 12:20 +0000 [r226490] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_local.c: Doxygen documentation update
+
+2009-10-28 20:50 +0000 [r226453] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * build_tools/get_documentation: remove empty awk pattern (//)
+ Solaris 10 nawk doesn't lthe empty pattern ike '//' for 'always'.
+ Just remove that. No pattern at all always matches.
+
+2009-10-28 20:11 +0000 [r226378-226384] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/sip.conf.sample: Merged revisions 226382 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28
+ Oct 2009) | 9 lines Update documentation in sip.conf.sample.
+ Update the documentation in sip.conf.sample in order to make it
+ more clear that directmedia/canreinvite do not cause Asterisk to
+ ignore reINVITEs. It is only used to stop Asterisk from
+ generating a reINVITE, but does not stop it from accepting them
+ if necessary. (closes issue #15644) Reported by: lmadsen ........
+
+ * doc/tex/channelvariables.tex: Merged revisions 226377 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009)
+ | 7 lines Update CALLINGSUBADDR channel variable documentation.
+ (closes issue #15734) Reported by: alecdavis Patches:
+ channelvariables.tex.diff.txt uploaded by alecdavis (license 585)
+ Tested by: alecdavis ........
+
+2009-10-28 18:04 +0000 [r226305] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/linkedlists.h: Merged revisions 226304 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28 Oct 2009)
+ | 2 lines Fix documentation (pointed out by TheDavidFactor on
+ #-dev) ........
+
+2009-10-28 08:47 +0000 [r226227-226270] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * contrib/upstart/asterisk.user.conf: Remove extra cleanup in case
+ we have more than one Asterisk. /var/run would be cleaned on
+ startup on most systems anyway.
+
+ * contrib/upstart/asterisk.user.conf (added): another variation of
+ the upstart script
+
+2009-10-27 21:03 +0000 [r226184] Olle Johansson <oej@edvina.net>
+
+ * Makefile: Adding compile time flags for Snow Leopard, Leopard and
+ some other animals
+
+2009-10-27 20:22 +0000 [r226159] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, /: Merged revisions 226138 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009)
+ | 7 lines Manager output is not always NULL-terminated, so force
+ a NULL at the end of the filestream. (closes issue #15495)
+ Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded
+ by tilghman (license 14) Tested by: pdf ........
+
+2009-10-27 16:48 +0000 [r226099] Terry Wilson <twilson@digium.com>
+
+ * res/res_http_post.c: Don't prepend the URI prefix to the post
+ directory
+
+2009-10-27 13:30 +0000 [r226060] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add
+ support for receiving unsolicited MWI NOTIFY messages. This
+ change adds a configuration option to SIP peers,
+ unsolicited_mailbox, which configures a virtual mailbox to use
+ for received new/old MWI information. This virtual mailbox can
+ then be used by any device supporting MWI. (closes issue #13028)
+ Reported by: AsteriskRocks Patches:
+ bug_13028_chan_sip_external_mwi_20090707.patch uploaded by cmaj
+ (license 830)
+
+2009-10-26 22:46 +0000 [r226018] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * /, configure, configure.ac: detect ARM Linux EABI OSARCH as
+ linux-gnu instead of linux-gnueabi * Set OSARCH to linux-gnu even
+ if host_os is linux-gnueabi * When checking if we are Linux,
+ check OSARCH rather than host_os The newer ARM ABI ("EABI") shows
+ the OS name 'linux-gnueabi' rather than 'linux-gnu' . This patch
+ sets OSARCH to be 'linux-gnu' even in such a case. OSARCH is
+ tested for the value of 'linux-gnu' in one or two places in the
+ tree. This patch also fixes the check libcap to check for $OSARCH
+ rather than $host_os . See also:
+ http://wiki.debian.org/ArmEabiPort Merged revisions 225957 via
+ svnmerge from http://svn.digium.com/svn/asterisk/branches/1.4
+
+2009-10-26 22:04 +0000 [r225955-225956] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/editline/makelist.in, channels/chan_sip.c, UPGRADE.txt,
+ UPGRADE-1.6.txt, doc/lang/language-criteria.txt: Fix building in
+ REF_DEBUG mode.
+
+ * main/astobj2.c: Correct broken logic from revision 225405. The
+ code committed in revision 225405 was broken; instead of removing
+ the unreference code, the logic used to decide when to do it
+ should have been reversed. This patch corrects the situation, and
+ makes reference counting work properly again.
+
+2009-10-26 19:40 +0000 [r225912] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_sip.c: ACL check not present for verifying SIP
+ INVITEs The ACL check in check_peer_ok was missing and has now
+ been restored. The missing check allowed for calls to be made on
+ prohibited networks where an ACL was defined in sip.conf and the
+ allowguest option was set to off. See the AST security advisory
+ below for more information. Merge code associated with
+ AST-2009-007. (closes issue #16091) Reported by: thom4fun
+
+2009-10-26 16:07 +0000 [r225872] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Make conditionals create previous code
+ when libpri/ss7 are present.
+
+2009-10-26 13:29 +0000 [r225767-225836] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_dahdi.c: span numbers in pri debug / error messages
+ Prefix PRI trace messages with the span number. This makes the
+ trace readable even when you have a multi-port device. (closes
+ issue #15054) Reported by: tzafrir Patches:
+ dahdi_pri_debug_spannum.diff uploaded by tzafrir (license 46)
+
+ * channels/chan_dahdi.c: Re-arange code a bit to build in dev-mode
+ without ss7 No change of functionality here. Just localized a
+ variable and indented code into blocks.
+
+ * channels/chan_dahdi.c: Make chan_dahdi build even without PRI /
+ SS7 (Note: still some strange build warnings without SS7 in
+ dev-mode)
+
+2009-10-24 14:40 +0000 [r225727] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: Improve performance of pedantic mode dialog
+ searching in chan_sip. This patch changes chan_sip to use the new
+ astobj2 OBJ_MULTIPLE iterator support to make pedantic mode
+ dialog searching in find_call() not require a linear search of
+ all dialogs in the list of dialogs. This patch does *not* change
+ the dialog matching logic (more on that later), just improves the
+ searching performance.
+
+2009-10-23 16:57 +0000 [r225692] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, CHANGES,
+ channels/sig_pri.c: Add to chan_dahdi ISDN HOLD, Call deflection,
+ and keypad facility support. * Added handling of received
+ HOLD/RETRIEVE messages and the optional ability to transfer a
+ held call on disconnect similar to an analog phone. * Added
+ CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI
+ PTMP. Will reroute/deflect an outgoing call when receive the
+ message. Can use the DAHDISendCallreroutingFacility to send the
+ message for the supported switches. * Added ability to
+ send/receive keypad digits in the SETUP message. Send keypad
+ digits in SETUP message:
+ Dial(DAHDI/g1[/K<keypad_digits>][/extension]) Access any received
+ keypad digits in SETUP message by: ${CHANNEL(keypad_digits)} *
+ Added support for BRI PTMP NT mode.
+
+2009-10-23 16:40 +0000 [r225690] Sean Bright <sean@malleable.com>
+
+ * Makefile, agi/Makefile, agi/agi.xml (added): Optionally build and
+ install the sample AGIs in the agi/ directory.
+
+2009-10-23 14:41 +0000 [r225650] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Fixes an iterator memory leak and
+ uninitialized memory
+
+2009-10-23 14:02 +0000 [r225582] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, /: Merged revisions 225581 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct
+ 2009) | 10 lines Don't force menuselect.makeopts to be rebuilt on
+ every build. For some reason the menuselect.makeopts file was
+ listed as PHONY in the Makefile, resulting in 'make' needing to
+ rebuild it for every build. This then resulted in the embedded
+ module rules being rebuilt on every build, which can be slow and
+ is unnecessary. This patch fixes the problem by properly allowing
+ 'make' to know when the menuselect.makeopts file needs to be
+ rebuilt (defining the proper dependencies). ........
+
+2009-10-22 22:24 +0000 [r225483-225515] Leif Madsen <lmadsen@digium.com>
+
+ * README: Update README documentation. Update the README
+ documentation to correctly describe which CLI command you should
+ use when attempting to get help from the CLI. (closes issue
+ #16064) Reported by: thedavidfactor Patches: readme.patch
+ uploaded by thedavidfactor (license 903)
+
+ * /, doc/valgrind.txt, contrib/valgrind.supp (added): Merged
+ revisions 225484 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009)
+ | 11 lines Clean valgrind output by suppressing false errors.
+ Update valgrind.txt documentation and add valgrind.supp file in
+ order to allow those who are creating valgrind output to have
+ less false errors in the logfile. (closes issue #16007) Reported
+ by: atis Patches: valgrind.txt.diff uploaded by atis (license
+ 242) asterisk2.supp uploaded by atis (license 242) Tested by:
+ atis, amorsen ........
+
+ * include/asterisk/doxyref.h,
+ include/asterisk/doxygen/asterisk-git-howto.h (added): Add
+ Asterisk Git HowTo documentation. Added documentation on how to
+ create a local git repository from SVN. This documentation was
+ added via doxygen. (closes issue #15814) Reported by: tzafrir
+ Patches: git-asterisk-howto uploaded by tzafrir (license 46)
+
+2009-10-22 20:07 +0000 [r225446] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Search for the subaddress only within the
+ extension section of the dial string.
+ Dial(DAHDI/(g|G|r|R)<group#(0-63)>[c|r<cadance#>|d][/extension])
+
+2009-10-22 19:55 +0000 [r225445] David Vossel <dvossel@digium.com>
+
+ * main/tcptls.c, channels/chan_sip.c, apps/app_externalivr.c,
+ include/asterisk/tcptls.h: SIP TCP/TLS: move client connection
+ setup/write into tcp helper thread, various related
+ locking/memory fixes. What this patch fixes 1.Moves sip TCP/TLS
+ connection setup into the TCP helper thread: Connection setup
+ takes awhile and before this it was being done while holding the
+ monitor lock. 2.Moves TCP/TLS writing to the TCP helper thread:
+ Through the use of a packet queue and an alert pipe, the TCP
+ helper thread can now be woken up to write data as well as read
+ data. 3.Locking error: sip_xmit returned an XMIT_ERROR without
+ giving up the tcptls_session lock. This lock has been completely
+ removed from sip_xmit and placed in the new sip_tcptls_write()
+ function. 4.Memory leak: When creating a tcptls_client the
+ tls_cfg was alloced but never freed unless the tcptls_session
+ failed to start. Now the session_args for a sip client are an ao2
+ object which frees the tls_cfg on destruction. 5.Pointer to stack
+ variable: During sip_prepare_socket the creation of a client's
+ ast_tcptls_session_args was done on the stack and stored as a
+ pointer in the newly created tcptls_session. Depending on the
+ events that followed, there was a slight possibility that pointer
+ could have been accessed after the stack returned. Given the new
+ changes, it is always accessed after the stack returns which is
+ why I found it. Notable code changes 1.I broke tcptls.c's
+ ast_tcptls_client_start() function into two functions. One for
+ creating and allocating the new tcptls_session, and a separate
+ one for starting and handling the new connection. This allowed me
+ to create the tcptls_session, launch the helper thread, and then
+ establish the connection within the helper thread. 2.Writes to a
+ tcptls_session are now done within the helper thread. This is
+ done by using an alert pipe to wake up the thread if new data
+ needs to be sent. The thread's sip_threadinfo object contains the
+ alert pipe as well as the packet queue. 3.Since the threadinfo
+ object contains the alert pipe, it must now be accessed outside
+ of the helper thread for every write (queuing of a packet). For
+ easy lookup, I moved the threadinfo objects from a linked list to
+ an ao2_container. (closes issue #13136) Reported by: pabelanger
+ Tested by: dvossel, whys (closes issue #15894) Reported by:
+ dvossel Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/380/
+
+2009-10-22 19:33 +0000 [r225440] Sean Bright <sean@malleable.com>
+
+ * Makefile, utils/Makefile, utils/utils.xml (added),
+ doc/janitor-projects.txt: Add the programs in utils/ to
+ menuselect. Nothing in utils/ is now built by default except for
+ astcanary. Review: https://reviewboard.asterisk.org/r/353/
+
+2009-10-22 19:10 +0000 [r225406] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
+ Permit storage of voicemail secrets in a separate file, located
+ within the spool directory. (closes issue #14276) Reported by:
+ klaus3000 Patches: app_voicemail.c-svn-trunk-r214898.txt uploaded
+ by klaus3000 (license 65) Tested by: jamesgolovich
+
+2009-10-22 18:41 +0000 [r225405] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/astobj2.c: Fix a refcount error introduced by yesterday's
+ OBJ_MULTIPLE commit. When an object is being unlinked from its
+ container *and* being returned to the caller, we do not want to
+ decrement the reference count after unlinking it from the
+ container, as the reference that the container held is what we
+ are returning to the caller... and if it was the only remaining
+ reference to the object, that could result in the object being
+ destroyed.
+
+2009-10-22 17:11 +0000 [r225360] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /, apps/app_meetme.c, include/asterisk/channel.h:
+ Merged revisions 225105 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009)
+ | 4 lines Fix documentation for ast_softhangup() and correct the
+ misuse thereof. (closes issue #16103) Reported by: majorbloodnok
+ ........
+
+2009-10-22 16:33 +0000 [r225357] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac, funcs/func_connectedline.c,
+ include/asterisk/channel.h, CHANGES, channels/sig_pri.c,
+ funcs/func_callerid.c: Add support for calling and called
+ subaddress. Partial support for COLP subaddress. The Telecom
+ Specs in NZ suggests that SUB ADDRESS is always on, so doing
+ "desk to desk" between offices each with an asterisk box over the
+ ISDN should then be possible, without a whole load of DDI numbers
+ required. (closes issue #15604) Reported by: alecdavis Patches:
+ asterisk_subaddr_trunk.diff11.txt uploaded by alecdavis (license
+ 585) Some minor modificatons were made. Tested by: alecdavis,
+ rmudgett Review: https://reviewboard.asterisk.org/r/405/
+
+2009-10-21 21:58 +0000 [r225307] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 225243 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21
+ Oct 2009) | 13 lines IAX2: VNAK loop caused by signaling frames
+ with no destination call number It is possible for the PBX thread
+ to queue up signaling frames before a destination call number is
+ received. This can result in signaling frames being sent out with
+ no destination call number. Since recent versions of Asterisk
+ require accurate destination callnumbers for all Full Frames,
+ this can cause a VNAK loop to occur. To resolve this no signaling
+ frames are sent until a destination callnumber is received, and
+ destination call numbers are now only required for iax_pvt
+ matching when the frame is an ACK. Review:
+ https://reviewboard.asterisk.org/r/413/ ........
+
+2009-10-21 21:15 +0000 [r225244-225245] Kevin P. Fleming <kpfleming@digium.com>
+
+ * doc/tex/manager.tex, channels/chan_sip.c: Add 'mohsuggest'
+ configuration option to 'sip show peer' CLI command and
+ SIPShowPeer AMI action. (closes issue #15990) Reported by:
+ _brent_ Patches: sip_peer_info_mohsuggest-r3.patch uploaded by
+ brent (license 388) Review:
+ https://reviewboard.asterisk.org/r/381/
+
+ * main/channel.c, main/manager.c, apps/app_directed_pickup.c,
+ apps/app_softhangup.c, funcs/func_channel.c,
+ include/asterisk/astobj2.h, res/snmp/agent.c,
+ include/asterisk/channel.h, include/asterisk/lock.h,
+ apps/app_chanspy.c, main/astobj2.c, main/cli.c: Finish
+ implementaton of astobj2 OBJ_MULTIPLE, and convert
+ ast_channel_iterator to use it. This patch finishes the
+ implementation of OBJ_MULTIPLE in astobj2 (the case where
+ multiple results need to be returned; OBJ_NODATA mode already was
+ supported). In addition, it converts ast_channel_iterators (only
+ the targeted versions, not the ones that iterate over all
+ channels) to use this method. During this work, I removed the
+ 'ao2_flags' arguments to the ast_channel_iterator constructor
+ functions; there were no uses of that argument yet, there is only
+ one possible flag to pass, and it made the iterators less
+ 'opaque'. If at some point in the future someone really needs an
+ ast_channel_iterator that does not lock the container, we can
+ provide constructor(s) for that purpose. Review:
+ https://reviewboard.asterisk.org/r/379/
+
+2009-10-21 16:46 +0000 [r225170-225172] Russell Bryant <russell@digium.com>
+
+ * /, main/translate.c: Merged revisions 225171 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225171 | russell | 2009-10-21 11:44:49 -0500 (Wed, 21 Oct 2009)
+ | 2 lines Revert 225169, as this doesn't account for the
+ possibility of a list of frames. ........
+
+ * /, main/translate.c: Merged revisions 225169 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225169 | russell | 2009-10-21 11:39:20 -0500 (Wed, 21 Oct 2009)
+ | 2 lines Isolate the frame returned from ast_translate().
+ ........
+
+2009-10-21 15:42 +0000 [r225102] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c: Apparently, I don't need to specify the ".so"
+ suffix to get a match
+
+2009-10-21 15:35 +0000 [r225089] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add
+ support for specifying the IP address to use for media streams in
+ sip.conf This is the second commit for this and documents the
+ text stream using the configured IP address and fixes a bug in
+ the original patch where the UDPTL stream would also use the
+ different IP address. (closes issue #14729) Reported by: _brent_
+ Patches: media_address.patch uploaded by brent (license 388)
+
+2009-10-21 15:21 +0000 [r225048] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c, CHANGES: Turn on DENOISE filter for all
+ conference participants. (Fixes SWP-238)
+
+2009-10-21 15:04 +0000 [r225034] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Revert
+ media_address commit, I'm going to roll a fix to the SDP
+ generation in the next version.
+
+2009-10-21 14:39 +0000 [r225033] David Vossel <dvossel@digium.com>
+
+ * configs/iax.conf.sample, /, channels/chan_sip.c,
+ configs/sip.conf.sample, channels/chan_iax2.c: Merged revisions
+ 225032 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009)
+ | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller
+ id removes '(', ' ', ')', non-trailing '.', and '-' from the
+ string. This means values such as 555.5555 and test-test result
+ in 555555 and testtest. There are instances, such as Skype
+ integration, where a specific value is passed via caller id that
+ must be preserved unmodified. This patch makes the shrinking of
+ caller id optional in chan_sip and chan_iax in order to support
+ such cases. By default this option is on to preserve previous
+ expected behavior. (closes issue #15940) Reported by: dimas
+ Patches: v2-15940.patch uploaded by dimas (license 88)
+ 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671)
+ Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/408/ ........
+
+2009-10-21 13:34 +0000 [r225003] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add
+ support for specifying the IP address to use for media streams in
+ sip.conf (closes issue #14729) Reported by: _brent_ Patches:
+ media_address.patch uploaded by brent (license 388)
+
+2009-10-21 03:09 +0000 [r224932] Russell Bryant <russell@digium.com>
+
+ * main/frame.c, /, main/translate.c, include/asterisk/dsp.h,
+ codecs/codec_dahdi.c, include/asterisk/frame.h,
+ include/asterisk/translate.h, main/dsp.c: Merged revisions 224931
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009)
+ | 5 lines Isolate frames returned from a DSP instance or codec
+ translator. The reasoning for these changes are the same as what
+ I wrote in the commit message for rev 222878. ........
+
+2009-10-21 02:43 +0000 [r224930] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Make PRI_SUBCMD_xxx handling subaddress
+ friendly.
+
+2009-10-20 22:09 +0000 [r224856] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_speex.c, /, main/audiohook.c: Merged revisions 224855
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009)
+ | 5 lines Pay attention to the return value of the manipulate
+ function. While this looks like an optimization, it prevents a
+ crash from occurring when used with certain audiohook callbacks
+ (diagnosed with SVN trunk, backported to 1.4 to keep the source
+ consistent across versions). ........
+
+2009-10-20 17:47 +0000 [r224774] Joshua Colp <jcolp@digium.com>
+
+ * /, main/features.c: Merged revisions 224773 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5
+ lines Add support for relaying early media in the features
+ attended transfer option. (closes issue #14828) Reported by:
+ licedey ........
+
+2009-10-20 12:44 +0000 [r224738] Matthew Nicholson <mnicholson@digium.com>
+
+ * CHANGES: Added information to CHANGES about the dynamic range
+ compression feature added to dahdi.
+
+2009-10-19 23:47 +0000 [r224671] Kevin P. Fleming <kpfleming@digium.com>
+
+ * res/res_rtp_asterisk.c, /: Merged revisions 224670 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19
+ Oct 2009) | 7 lines Correct timestamp calculations when RTP
+ sample rates over 8kHz are used. While testing some endpoints
+ that support 16kHz and 32kHz sample rates, some log messages were
+ generated due to calc_rxstamp() computing timestamps in a way
+ that produced odd results, so this patch sanitizes the result of
+ the computations. ........
+
+2009-10-19 22:02 +0000 [r224637] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add
+ dynamic range compression support for analog channels. (closes
+ issue AST-29)
+
+2009-10-19 19:49 +0000 [r224567] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 224565 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5
+ lines Do not attempt early media bridging (ie: direct RTP setup)
+ if options are enabled that should prevent it. (closes issue
+ #14763) Reported by: cupotka ........
+
+2009-10-19 19:40 +0000 [r224562] Kevin P. Fleming <kpfleming@digium.com>
+
+ * formats/format_siren14.c: Remove useless debugging message.
+
+2009-10-19 15:50 +0000 [r224527] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/janitor-projects.txt: Remove a completed project and add
+ another
+
+2009-10-19 14:32 +0000 [r224491] Joshua Colp <jcolp@digium.com>
+
+ * channels/sig_pri.h, channels/sig_pri.c: Add a callback to sig_pri
+ which is called when sig_pri is going to queue a control frame on
+ a channel.
+
+2009-10-19 00:05 +0000 [r224446-224448] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Allow ODBC storage to be queried with
+ multiple mailboxes, and remove multiple goto's. This corrects an
+ issue reported on the -users list.
+
+ * configs/res_odbc.conf.sample: Clarify that "forcecommit" is NOT
+ an alias for "autocommit", but instead controls the default
+ disposition of uncommitted transactions.
+
+2009-10-17 16:39 +0000 [r224403] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/app.h, main/app.c: Remove unnecessary typedef
+
+2009-10-17 02:01 +0000 [r224331-224335] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: fix typo, sorry
+
+ * channels/chan_dahdi.c, /, channels/sig_pri.c: Merged revisions
+ 224330 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009)
+ | 13 lines Fix stale caller id data from being reported in AMI
+ NewChannel event The problem here is that chan_dahdi is designed
+ in such a way to set certain values in the dahdi_pvt only once.
+ One of those such values is the configured caller id data in
+ chan_dahdi.conf. For PRI, the configured caller id data could be
+ overwritten during a call. Instead of saving the data and
+ restoring, it was decided that for all non-analog channels it was
+ simply best to not set the configured caller id in the first
+ place and also clear it at the end of the call. (closes issue
+ #15883) Reported by: jsmith ........
+
+2009-10-16 20:40 +0000 [r224261] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 224260 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009)
+ | 18 lines Never released PRI channels when using Busy() or
+ Congestion() dialplan apps. When the Busy() or Congestion()
+ application is used towards ISDN (an ISDN progress is sent), the
+ responding ISDN Disconnect or Release may contain the ISDN cause
+ user busy or one of the congestion causes. In chan_dahdi.c these
+ causes will only set the needbusy or needcongestion flags and not
+ activate the softhangup procedure. Unfortunately only the latter
+ can interrupt the endless wait loop of Busy()/Congestion().
+ Result: PRI channels staying in state busy for the rest of
+ asterisk life or until the other end times out and forces the
+ call to clear. (issue #14292) Reported by: tomaso Patches:
+ disc_rel_userbusy.patch uploaded by tomaso (license 564) (This
+ patch is unrelated to the issue.) ........
+
+2009-10-15 22:33 +0000 [r224225] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/app.h, main/pbx.c, main/app.c: Create an API for
+ adding an optional time unit onto the ends of time periods. Two
+ examples of its use are included, and the usage could be expanded
+ in some cases into certain configuration options where time
+ periods are specified.
+
+2009-10-15 15:57 +0000 [r224178] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_chanspy.c: Readd removed ability to allow listening to
+ one side of the call in app_chanspy (Option o) (closes issue
+ #15675) Reported by: john8675309 Patches:
+ issue15675patchtrunk.txt uploaded by dbrooks (license 790) Tested
+ by: jgutierrez on users list:
+ http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html
+
+2009-10-15 14:37 +0000 [r224144] Doug Bailey <dbailey@digium.com>
+
+ * configs/chan_dahdi.conf.sample: chan_dahdi.conf.sample changes
+ for DTMF CID detect Explains new options for detecting DTMF CID
+ on fxo lines (issue #9096) Reported by: fleed Patches:
+ chan_dahid_sample_config.patch uploaded by sum (license 766)
+
+2009-10-15 06:48 +0000 [r224074-224109] Terry Wilson <twilson@digium.com>
+
+ * res/res_calendar_caldav.c: Properly handle PUT requests for
+ CALENDAR_WRITE()
+
+ * res/res_calendar.c: Add missing 'getnum' field
+
+2009-10-14 17:48 +0000 [r224035] Jeff Peeler <jpeeler@digium.com>
+
+ * configs/sip_notify.conf.sample, channels/chan_sip.c, CHANGES:
+ Allow for adding message body to the SIP NOTIFY message Ability
+ has been added to both manager command SIPnotify as well as
+ console command sip notify. Message body is stored in the
+ "Content" variable. An example is present in sip_notify.conf.
+ (closes issue #13926) Reported by: jthurman Patches:
+ sip-notify-svn189463.diff uploaded by gareth (license 208) Tested
+ by: gareth
+
+2009-10-13 22:14 +0000 [r223992] Terry Wilson <twilson@digium.com>
+
+ * res/res_calendar.c: use Calendar: instead of Calendar/ for
+ devstate
+
+2009-10-13 17:11 +0000 [r223911-223912] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/pbx.h: Fix some doxygen format problems and trim
+ trailing whitespace.
+
+ * res/res_calendar.c: Fix compiler warning.
+
+2009-10-13 01:58 +0000 [r223874-223875] Terry Wilson <twilson@digium.com>
+
+ * apps/app_originate.c: Revert inadvertant code commit to
+ app_originate
+
+ * apps/app_originate.c, include/asterisk/calendar.h,
+ res/res_calendar.c: Fix handling of notification calls w/ the
+ dialing api
+
+2009-10-12 23:48 +0000 [r223832] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 223804 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009)
+ | 8 lines Ensure ringing continues for branched calls after
+ progress is received While waiting for an answer, don't send
+ progress for branched calls for which ringing was sent. (closes
+ issue #15028) Reported by: fnordian ........
+
+2009-10-12 20:58 +0000 [r223756] David Vossel <dvossel@digium.com>
+
+ * configs/iax.conf.sample: Clarifies trunkmaxsize, trunkfreq, and
+ trunkmtu iax2 options SWP-151
+
+2009-10-12 15:32 +0000 [r223652-223693] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /: Recorded merge of revisions 223692 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223692 | kpfleming | 2009-10-12 10:30:40 -0500 (Mon, 12 Oct
+ 2009) | 13 lines Remove automatic switching from T.38 to voice
+ mode in chan_sip. chan_sip has some code to automatically switch
+ from T.38 mode to voice mode when a voice frame is written to the
+ channel while it is in T.38 mode; this was intended to handle the
+ situation when a FAX transmission has ended and the channel is
+ not yet hung up, but is causing problems at the beginning of FAX
+ sessions as well when there are still voice frames 'in flight' at
+ the time the T.38 negotiation completes. This patch removes the
+ automatic switchover. (issue #16025) Reported by: jamicque
+ ........
+
+ * channels/chan_sip.c, apps/app_fax.c: Remove automatic switching
+ from T.38 to voice mode in chan_sip. chan_sip has some code to
+ automatically switch from T.38 mode to voice mode when a voice
+ frame is written to the channel while it is in T.38 mode; this
+ was intended to handle the situation when a FAX transmission has
+ ended and the channel is not yet hung up, but is causing problems
+ at the beginning of FAX sessions as well when there are still
+ voice frames 'in flight' at the time the T.38 negotiation
+ completes. This patch removes the automatic switchover, and
+ changes app_fax to explicitly switch off T.38 mode when the FAX
+ transmission process ends. (closes issue #16025) Reported by:
+ jamicque
+
+2009-10-11 22:19 +0000 [r223617] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Check the proper page for the SENDRPID flag.
+ If a pending reinvite were sent, we might not properly send
+ connected party info since we were checking the wrong flag. This
+ was a rare occurrence, but could still happen nevertheless.
+
+2009-10-11 18:35 +0000 [r223487-223553] Russell Bryant <russell@digium.com>
+
+ * /: Merged revisions 223550 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223550 | russell | 2009-10-11 13:34:37 -0500 (Sun, 11 Oct 2009)
+ | 2 lines Remove a duplicate ao2_iterator_destroy(). ........
+
+ * main/autoservice.c, /: Merged revisions 223485-223486 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223485 | russell | 2009-10-11 12:22:52 -0500 (Sun, 11 Oct 2009)
+ | 6 lines Don't use data outside of its scope. The purpose of
+ this code was to have a hangup frame put on the list of deferred
+ frames. However, the code that read the hangup frame was outside
+ of the scope of where the hangup frame was declared. ........
+ r223486 | russell | 2009-10-11 12:25:06 -0500 (Sun, 11 Oct 2009)
+ | 2 lines Remove some unnecessary code. ........
+
+2009-10-10 20:02 +0000 [r223449] Terry Wilson <twilson@digium.com>
+
+ * res/res_calendar_icalendar.c, res/res_calendar_caldav.c: Fix
+ handling of floating times and dates
+
+2009-10-10 08:30 +0000 [r223413-223415] Olle Johansson <oej@edvina.net>
+
+ * configs/cdr_pgsql.conf.sample: Adding note about TLS usage
+
+ * configs/res_ldap.conf.sample: Add an additional note on TLS
+ support
+
+ * configs/res_ldap.conf.sample: Adding some information on TLS
+ support
+
+2009-10-09 22:04 +0000 [r223370] Terry Wilson <twilson@digium.com>
+
+ * res/res_calendar_icalendar.c, res/res_calendar_caldav.c: Properly
+ return "free" on confirmed events that are free CONFIRMED status
+ doesn't imply busy or free, that is handled with the TRANSP
+ field. Luckily, libical already sets the is_busy status on the
+ span for us.
+
+2009-10-09 20:58 +0000 [r223330] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_fax.c: Initiate T.38 switchover when acting as called
+ party, regardless of FAX direction. SendFAX() and ReceiveFAX()
+ can be given options to indicate whether they should act as the
+ calling or called party; this mode should be used to decide
+ whether to initiate a switchover to T.38, not the direction that
+ the FAX transfer will take place. (closes issue #16039) Reported
+ by: jamicque
+
+2009-10-09 18:34 +0000 [r223273] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, /: Merged revisions 223225 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct
+ 2009) | 8 lines Signal timeouts by returning AST_CONTROL_RINGING
+ when originating calls. (closes issue #15104) Reported by:
+ nblasgen Patches: manager-timeout1.diff uploaded by mnicholson
+ (license 96) Tested by: nblasgen, mnicholson ........
+
+2009-10-09 18:17 +0000 [r223211-223215] Mark Michelson <mmichelson@digium.com>
+
+ * /: Recorded merge of revisions 223213 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri, 09 Oct
+ 2009) | 3 lines Fix potential memory leak in app_dial.c ........
+
+ * apps/app_dial.c: Fix potential memory leaks. ABE-1998
+
+2009-10-09 17:53 +0000 [r223206] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 223205 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009)
+ | 10 lines fixes sip registration using authuser in user.conf
+ (closes issue #14954) Reported by: tornblad Tested by:
+ mmichelson, tornblad, dvossel ........
+
+2009-10-09 17:14 +0000 [r223136] Matthew Nicholson <mnicholson@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c: Don't close the sqlite database when
+ reloading. Only close the database when unloading. (closes issue
+ #15953) Reported by: frawd Patches: sqlite3_rev220097.diff
+ uploaded by frawd (license 610) Tested by: frawd
+
+2009-10-09 16:54 +0000 [r223088-223132] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: 'auth=' did not parse md5 secret correctly
+ (closes issue #15949) Reported by: ebroad Patches:
+ authparsefix.patch uploaded by ebroad (license 878)
+ 15949_trunk.diff uploaded by dvossel (license 671) Tested by:
+ ebroad
+
+ * channels/chan_sip.c: p->peerauth is always empty in
+ transmit_register() When using callbackextension or specifing the
+ peer name in a registration string, the peer's specific auth
+ settings set by the "auth=" strings within the peer definition
+ are not used by the registration. Thanks to ebroad for reporting
+ the issue and providing the patch. (closes issue #15955) Reported
+ by: ebroad Patches: regauthfix.patch uploaded by ebroad (license
+ 878)
+
+2009-10-09 15:00 +0000 [r223016-223053] Terry Wilson <twilson@digium.com>
+
+ * res/res_calendar.c: Don't add Attendees during copy, replace them
+
+ * res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
+ res/res_calendar_caldav.c, include/asterisk/calendar.h,
+ res/res_calendar.c: Remove global variable that makes dlopen
+ unhappy This isn't the best way to do this, but it is the
+ easiest. There are some limitations that are going to need to be
+ addressed at some point with reloads and when I (or someone else)
+ work on that, then the API can be updated to handle passing the
+ private config data that the calendar tech modules need in a
+ better way as well.
+
+2009-10-08 22:57 +0000 [r222947-223015] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixed comment line for do_magic_pickup
+
+ * channels/chan_sip.c: Deadlock between ast_cel_report_event and
+ ast_do_masquerade chan_sip calls pbx_exec on a pvt's owner
+ channel while only the pvt lock is held. Since pbx_exec calls
+ ast_cel_report_event which attempts to lock the channel, invalid
+ locking order occurs. Channels should be locked before pvt's.
+ (closes issue #15512) Reported by: lmsteffan Patches:
+ ast_cel_deadlock_15512.diff uploaded by dvossel (license 671)
+
+ * channels/chan_sip.c: makes externtcpport and externtlsport static
+ variables externtcpport and externtlsport need to be declared as
+ static variables. Thanks to russell for finding and pointing this
+ out.
+
+2009-10-08 19:52 +0000 [r222880] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/file.h, main/frame.c, /, main/file.c,
+ include/asterisk/frame.h: Merged revisions 222878 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08
+ Oct 2009) | 44 lines Make filestream frame handling safer by
+ isolating frames before returning them. This patch is related to
+ a number of issues on the bug tracker that show crashes related
+ to freeing frames that came from a filestream. A number of fixes
+ have been made over time while trying to figure out these
+ problems, but there re still people seeing the crash. (Note that
+ some of these bug reports include information about other
+ problems. I am specifically addressing the filestream frame crash
+ here.) I'm still not clear on what the exact problem is. However,
+ what is _very_ clear is that we have seen quite a few problems
+ over time related to unexpected behavior when we try to use
+ embedded frames as an optimization. In some cases, this
+ optimization doesn't really provide much due to improvements made
+ in other areas. In this case, the patch modifies filestream
+ handling such that the embedded frame will not be returned.
+ ast_frisolate() is used to ensure that we end up with a
+ completely mallocd frame. In reality, though, we will not
+ actually have to malloc every time. For filestreams, the frame
+ will almost always be allocated and freed in the same thread.
+ That means that the thread local frame cache will be used. So,
+ going this route doesn't hurt. With this patch in place, some
+ people have reported success in not seeing the crash anymore.
+ (SWP-150) (AST-208) (ABE-1834) (issue #15609) Reported by: aragon
+ Patches: filestream_frisolate-1.4.diff2.txt uploaded by russell
+ (license 2) Tested by: aragon, russell (closes issue #15817)
+ Reported by: zerohalo Tested by: zerohalo (closes issue #15845)
+ Reported by: marhbere Review:
+ https://reviewboard.asterisk.org/r/386/ ........
+
+2009-10-08 19:35 +0000 [r222873] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/netsock.h, main/netsock.c: fixes an
+ ast_netsock_list memory leak. ABE-1998 Review:
+ https://reviewboard.asterisk.org/r/395/
+
+2009-10-08 16:44 +0000 [r222799] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/misdn_config.c: Merged revisions 222797 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r222797 | rmudgett | 2009-10-08 11:33:06 -0500 (Thu, 08
+ Oct 2009) | 12 lines Fix memory leak if chan_misdn config
+ parameter is repeated. Memory leak when the same config option is
+ set more than once in an misdn.conf section. Why must this be
+ considered? Templates! Defining a template with default port
+ options and later adding to or overriding some of them. Patches:
+ memleak-misdn.patch JIRA ABE-1998 ........
+
+2009-10-07 22:58 +0000 [r222761] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, main/pbx.c, channels/chan_misdn.c,
+ channels/chan_sip.c, main/features.c, include/asterisk/channel.h:
+ Deadlock in channel masquerade handling Channels are stored in an
+ ao2_container. When accessing an item within an ao2_container the
+ proper locking order is to first lock the container, and then the
+ items within it. In ast_do_masquerade both the clone and original
+ channel must be locked for the entire duration of the function.
+ The problem with this is that it attemptes to unlink and link
+ these channels back into the ao2_container when one of the
+ channel's name changes. This is invalid locking order as the
+ process of unlinking and linking will lock the ao2_container
+ while the channels are locked!!! Now, both the channels in
+ do_masquerade are unlinked from the ao2_container and then locked
+ for the entire function. At the end of the function both channels
+ are unlocked and linked back into the container with their new
+ names as hash values. This new method of requiring all channels
+ and tech pvts to be unlocked before ast_do_masquerade() or
+ ast_change_name() required several changes throughout the code
+ base. (closes issue #15911) Reported by: russell Patches:
+ masq_deadlock_trunk.diff uploaded by dvossel (license 671) Tested
+ by: dvossel, atis (closes issue #15618) Reported by: lmsteffan
+ Patches: deadlock_local_attended_transfers_trunk.diff uploaded by
+ dvossel (license 671) Tested by: lmsteffan, dvossel Review:
+ https://reviewboard.asterisk.org/r/387/
+
+2009-10-07 21:56 +0000 [r222692] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 222691 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r222691 | rmudgett | 2009-10-07 16:51:24 -0500 (Wed, 07
+ Oct 2009) | 14 lines chan_misdn.c:process_ast_dsp() memory leak
+ misdn.conf: astdtmf must be set to "yes". With "no", buffer loss
+ does not occur. The translated frame "f2" when passing through
+ ast_dsp_process() is not freed whenever it is not used further in
+ process_ast_dsp(). Then in the end it is never ever freed.
+ Patches: translate.patch JIRA ABE-1993 ........
+
+2009-10-07 20:08 +0000 [r222652] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Change ringt (ring timeout) styles to be
+ consistent across chan_dahdi. (closes issue #15684) Reported by:
+ alecdavis Patches: chan_dahdi.bug15684.diff2.txt uploaded by
+ alecdavis (license 585) Tested by: alecdavis
+
+2009-10-07 18:57 +0000 [r222614-222615] Olle Johansson <oej@edvina.net>
+
+ * res/res_config_ldap.c: Formatting, moving error messages to
+ ERROR, removing references to unexisting debug output. No
+ functionality changes.
+
+ * cel/cel_pgsql.c, res/res_config_pgsql.c, cdr/cdr_pgsql.c: Use
+ extref for doxygen references to external libraries (in this case
+ PostgreSQL)
+
+2009-10-07 18:04 +0000 [r222548] Jason Parker <jparker@digium.com>
+
+ * configs/queues.conf.sample: Remove 'keepstats' queue option from
+ sample config, as it's no longer used.
+ https://reviewboard.asterisk.org/r/115/ (closes issue #15820)
+ Reported by: kshumard
+
+2009-10-07 17:44 +0000 [r222543] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 222542 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r222542 | dvossel | 2009-10-07 12:41:21 -0500 (Wed, 07 Oct 2009)
+ | 8 lines crash on transfer handle_invite_replaces() attempts to
+ uplock a pvt's owner channel without first verifing that it
+ exists. (issue #16027) ........
+
+2009-10-06 23:56 +0000 [r222463] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 222462 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r222462 | jpeeler | 2009-10-06 18:51:19 -0500 (Tue, 06
+ Oct 2009) | 8 lines Add missing unlock(s) in dahdi_read (two
+ cases in trunk) (closes issue #15683) Reported by: alecdavis
+ ........
+
+2009-10-06 22:49 +0000 [r222398-222399] David Vossel <dvossel@digium.com>
+
+ * CHANGES: Updates CHANGES to reflect the new externtcpport and
+ externtlsport sip options
+
+ * channels/chan_sip.c, configs/sip.conf.sample: contact header port
+ ignored transport when using externip This patch adds support for
+ TCP/TLS in the Contact header when using NAT, specifically
+ externip or externhost. The original issue was that Asterisk sent
+ 5060 as the port in the contact header whether TLS was used or
+ not. Additionally, this patch adds 2 config options to sip.conf,
+ specifically externtcpport and externtlsport. This allows a user
+ to specify different external ports for TCP and TLS other than
+ those used internally, this is especially useful in in a PAT/port
+ redirection setup. Thanks to ebroad for reporting the issue and
+ providing the patch! (closes issue #15880) Reported by: ebroad
+ Patches: portmap.patch uploaded by ebroad (license 878)
+ externtXXport_v2.patch uploaded by ebroad (license 878) Tested
+ by: ebroad Review: https://reviewboard.asterisk.org/r/392/
+
+2009-10-06 20:35 +0000 [r222351] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Fix 222298 (crash during destruction of
+ second channel when variable set with setvar). I mistakenly
+ reasoned that setvar would be used on all channels. Since it can
+ be set per channel, give each dahdi channel a copy of the
+ variable. (related to #15899)
+
+2009-10-06 19:31 +0000 [r222309] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_pgsql.c, cdr/cdr_pgsql.c: Change schema query to
+ involve the use of an optional schema parameter. This change is
+ done in such a way as to allow the driver to continue to function
+ with older databases which don't have these features. (closes
+ issue #16000) Reported by: jamicque Patches:
+ 20091002__issue16000.diff.txt uploaded by tilghman (license 14)
+ 20091002__issue16000__1.6.1.diff.txt uploaded by tilghman
+ (license 14) Tested by: jamicque
+
+2009-10-06 19:24 +0000 [r222298] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Fix crash during destruction of second
+ channel when variable set with setvar. The setvar line in
+ chan_dahdi.conf is shared among all the channels, so make sure to
+ only free the resources only when the last channel is destroyed.
+ (closes issue #15899) Reported by: tzafrir
+
+2009-10-06 19:17 +0000 [r222273] Tilghman Lesher <tlesher@digium.com>
+
+ * res/ael/pval.c: When we call a gosub routine, the variables
+ should be scoped to avoid contaminating the caller. This affected
+ the ~~EXTEN~~ hack, where a subroutine might have changed the
+ value before it was used in the caller. Patch by myself, tested
+ by ebroad on #asterisk
+
+2009-10-06 16:17 +0000 [r222237] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_dahdi.c: Make sure digit events are not reported as
+ "ERROR" dahdievent_to_analogevent used a simple switch statement
+ to convert DAHDI event numbers to "ANALOG_*" event numbers.
+ However "digit" events (DAHDI_EVENT_PULSEDIGIT,
+ DAHDI_EVENT_DTMFDOWN, DAHDI_EVENT_DTMFUP) are accompannied by the
+ digit in the low word of the event number. This fix makes
+ dahdievent_to_analogevent() return the event number as-is for
+ such an event. This is also required to fix #15924 (in addition
+ to r222108).
+
+2009-10-06 01:24 +0000 [r222110-222176] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_sip.c, funcs/func_dialgroup.c,
+ include/asterisk/astobj2.h, res/res_phoneprov.c,
+ channels/chan_console.c, res/res_musiconhold.c, apps/app_queue.c,
+ channels/chan_iax2.c, main/astobj2.c, res/res_odbc.c,
+ res/res_calendar.c, res/res_clialiases.c: Recorded merge of
+ revisions 222152 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct
+ 2009) | 20 lines Fix ao2_iterator API to hold references to
+ containers being iterated. See Mantis issue for details of what
+ prompted this change. Additional notes: This patch changes the
+ ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum
+ instead of a macro, with a name that fits our naming policy;
+ also, it is now necessary to call ao2_iterator_destroy() on any
+ iterator that has been created. Currently this only releases the
+ reference to the container being iterated, but in the future this
+ could also release other resources used by the iterator, if the
+ iterator implementation changes to use additional resources.
+ (closes issue #15987) Reported by: kpfleming Review:
+ https://reviewboard.asterisk.org/r/383/ ........
+
+ * main/udptl.c, channels/chan_sip.c, configs/udptl.conf.sample,
+ UPGRADE.txt, configs/sip.conf.sample: Allow non-compliant T.38
+ endpoints to be supportable via configuration option. Many T.38
+ endpoints incorrectly send the maximum IFP frame size they can
+ accept as the T38FaxMaxDatagram value in their SDP, when in fact
+ this value is supposed to be the maximum UDPTL payload size
+ (datagram size) they can accept. If the value they supply is
+ small enough (a commonly supplied value is '72'), T.38 UDPTL
+ transmissions will likely fail completely because the UDPTL
+ packets will not have enough room for a primary IFP frame and the
+ redundancy used for error correction. If this occurs, the
+ Asterisk UDPTL stack will emit log messages warning that data
+ loss may occur, and that the value may need to be overridden.
+ This patch extends the 't38pt_udptl' configuration option in
+ sip.conf to allow the administrator to override the value
+ supplied by the remote endpoint and supply a value that allows
+ T.38 FAX transmissions to be successful with that endpoint. In
+ addition, in any SIP call where the override takes effect, a
+ debug message will be printed to that effect. This patch also
+ removes the T38FaxMaxDatagram configuration option from
+ udptl.conf.sample, since it has not actually had any effect for a
+ number of releases. In addition, this patch cleans up the T.38
+ documentation in sip.conf.sample (which incorrectly documented
+ that T.38 support was passthrough only). (issue #15586) Reported
+ by: globalnetinc
+
+2009-10-05 19:20 +0000 [r222108] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h: Add a few missing events to
+ analog_handle_event. The reported bug was actually only for
+ pulsedigit, dtmfup, and dtmfdown handling. Also added recognition
+ for fax events (just some verbose output) and fixed handling for
+ the ec_disabled_event. In order to make comparing the analog
+ version of events to the DAHDI events easier, the ordering has
+ been changed to follow that of the DAHDI events. (closes issue
+ #15924) Reported by: tzafrir
+
+2009-10-02 17:34 +0000 [r222030] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 222026 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r222026 | dvossel | 2009-10-02 12:32:13 -0500 (Fri, 02
+ Oct 2009) | 3 lines Removes unnecessary unlock, clarifies a
+ memcpy. ........
+
+2009-10-02 16:59 +0000 [r221920-221971] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/astobj2.c: Merged revisions 221970 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009)
+ | 2 lines Ensure the result of the hash function is positive.
+ Negative array offsets suck. ........
+
+ * main/logger.c: Initialize a variable that we check immediately
+ upon startup. (closes issue #15973) Reported by: atis
+
+2009-10-02 01:49 +0000 [r221844-221881] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c: Whitespace change.
+
+ * channels/misdn/isdn_lib.c: Whitespace change.
+
+ * channels/misdn/isdn_lib_intern.h, /, channels/misdn/isdn_lib.c:
+ Merged revisions 221769 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009)
+ | 26 lines Occasionally losing use of B channels in chan_misdn. I
+ have not been able to reproduce the problem of losing channels.
+ However, I have seen in the code a reentrancy problem that might
+ give these symptoms. The reentrancy patch does several things: 1)
+ Guards B channel and B channel structure allocation. 2) Makes the
+ B channel structure find routines more precise in locating
+ records. 3) Never leave a B channel allocated if we received
+ cause 44. The last item may cause temporary outgoing call
+ problems, but they should clear when the line becomes idle.
+ (closes issue #15490) Reported by: slutec18 Patches:
+ issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett
+ (license 664) Tested by: rmudgett, slutec18 (closes issue #15458)
+ Reported by: FabienToune Patches:
+ issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett
+ (license 664) Tested by: FabienToune, rmudgett, slutec18 ........
+
+2009-10-02 00:08 +0000 [r221777-221781] Tilghman Lesher <tlesher@digium.com>
+
+ * main/say.c: One more off-by-one in trunk
+
+ * main/rtp_engine.c, /, main/say.c, main/asterisk.c: Merged
+ revisions 221776 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009)
+ | 2 lines Fix a bunch of off-by-one errors ........
+
+2009-10-01 20:18 +0000 [r221709] Richard Mudgett <rmudgett@digium.com>
+
+ * UPGRADE.txt, CHANGES: Move DAHDI/ISDN channel naming note from
+ CHANGES to UPGRADE.txt.
+
+2009-10-01 20:09 +0000 [r221705] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Revision 220906 (a merge from 1.4) was not
+ merged correctly, causing a problem with non-dynamic peers.
+
+2009-10-01 19:48 +0000 [r221701] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, CHANGES: Prevent
+ deadlock if chan_dahdi attempts to change PRI channel names. The
+ PRI channels can no longer change the channel name if a different
+ B channel is selected during call negotiation. To prevent using
+ the channel name to infer what B channel a call is using and to
+ avoid name collisions, the channel name format is changed. The
+ new channel naming for PRI channels is:
+ DAHDI/ISDN-<span>-<sequence-number>
+
+2009-10-01 19:33 +0000 [r221697] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: outbound tls connections were not defaulting
+ to port 5061 (closes issue #15854) Reported by: dvossel Patches:
+ sip_port_config_trunk.diff uploaded by dvossel (license 671)
+ Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/357/
+
+2009-10-01 16:27 +0000 [r221592-221627] Kevin P. Fleming <kpfleming@digium.com>
+
+ * UPGRADE.txt: Sync up UPGRADE.txt with the 1.6.2 version.
+
+ * main/udptl.c, configs/udptl.conf.sample: Remove ability to
+ control T.38 FAX error correction from udptl.conf. chan_sip has
+ had the ability to control T.38 FAX error correction mode on a
+ per-peer (or global) basis for a couple of releases now, which is
+ where it should have been all along. This patch removes the
+ ability to configure it in udptl.conf, but issues a warning if
+ the user tries to do, telling them to look at sip.conf.sample for
+ how to configure it now. For any SIP peers that are T.38 enabled
+ in sip.conf, there is already a default for FEC error correction
+ even if the user does not specify any mode, so this change will
+ not turn off error correction by default, it will have the same
+ default value that has been in the udptl.conf sample file.
+
+2009-10-01 15:26 +0000 [r221589] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 221588 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct
+ 2009) | 2 lines Use unsigned ints for portinuri flags. ........
+
+2009-10-01 07:00 +0000 [r221554] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Simplify code for porturi, use TRUE/FALSE
+ constructs when it's just TRUE or FALSE.
+
+2009-09-30 23:04 +0000 [r221484] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Cleaned up merge from r221432
+
+2009-09-30 21:15 +0000 [r221436] Matthias Nick <mnick@digium.com>
+
+ * apps/app_queue.c: Prevents from division by zero
+
+2009-09-30 20:40 +0000 [r221432] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+ 221360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep
+ 2009) | 10 lines Fix SRV lookup and Request-URI generation in
+ chan_sip. This patch adds a new field "portinuri" to the sip
+ dialog struct and the sip peer struct. That field is used during
+ RURI generation to determine if the port should be included in
+ the RURI. It is also used in some places to determine if an SRV
+ lookup should occur. (closes issue #14418) Reported by: klaus3000
+ Tested by: klaus3000, mnicholson Review:
+ https://reviewboard.asterisk.org/r/369/ ........
+
+2009-09-30 19:42 +0000 [r221368] Matthias Nick <mnick@digium.com>
+
+ * configs/cdr_custom.conf.sample, /, funcs/func_strings.c: Merged
+ revisions 221153,221157,221303 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) |
+ 2 lines check bounds - prevents for buffer overflow ........
+ r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) |
+ 8 lines added a new dialplan function 'CSV_QUOTE' and changed the
+ cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr
+ Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by:
+ mnick ........ r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed,
+ 30 Sep 2009) | 2 lines changed the prototype definition of
+ csv_quote ........
+
+2009-09-30 18:47 +0000 [r221266-221300] Terry Wilson <twilson@digium.com>
+
+ * res/res_rtp_asterisk.c: Remove spurious debug
+
+ * res/res_rtp_asterisk.c, main/rtp_engine.c, channels/chan_sip.c,
+ include/asterisk/rtp_engine.h: Use rtp properties instead of
+ adding a callback Thanks, Josh.
+
+ * res/res_rtp_asterisk.c, main/rtp_engine.c, /,
+ channels/chan_sip.c, configs/sip.conf.sample,
+ include/asterisk/rtp_engine.h: Merged revisions 221086 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009)
+ | 25 lines Change the SSRC by default when our media stream
+ changes Be default, change SSRC when doing an audio stream
+ changes Asterisk doesn't honor marker bit when reinvited to
+ already-bridged RTP streams,resulting in far-end stack discarding
+ packets with "old" timestamps that areactually part of a new
+ stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is
+ a reinvite, unless the 'constantssrc' is set to true in sip.conf.
+ The original issue reported to Digium support detailed the
+ following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based
+ Application Server Call comes in fromITSP, Asterisk dials the app
+ server which sends a re-invite back toAsterisk--not to negotiate
+ to send media directly to the ITSP, but to indicatethat it's
+ changing the stream it's sending to Asterisk. The app
+ servergenerates a new SSRC, sequence numbers, timestamps, and
+ sets the marker bit on the new stream. Asterisk passes through
+ the teimstamp of the new stream, butdoes not reset the SSRC,
+ sequence numbers, or set the marker bit. When the timestamp on
+ the new stream is older than the timestamp on the originalstream,
+ the ITSP (which doesn't know there has been any change) discards
+ the newframes because it thinks they are too old. This patch
+ addresses this by changing the SSRC on a stream update unless
+ constantssrc=true is set in sip.conf. Review:
+ https://reviewboard.asterisk.org/r/374/ ........
+
+2009-09-30 16:56 +0000 [r221201] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 221200 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009)
+ | 7 lines Avoid a potential NULL dereference. (closes issue
+ #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt
+ uploaded by tilghman (license 14) Tested by: kobaz ........
+
+2009-09-30 15:11 +0000 [r221085-221090] Sean Bright <sean@malleable.com>
+
+ * apps/app_voicemail.c: Modify VoiceMailMain()'s a() argument to
+ allow mailboxes to be specified by name. (closes issue #14740)
+ Reported by: pj Patches: issue14740_09022009.diff uploaded by
+ seanbright (license 71) Tested by: seanbright, lmadsen
+
+ * apps/app_voicemail.c: Clarify documentation for VoiceMailMain()'s
+ a() option. We require box numbers, not names as the
+ documentation implies. (issue #14740) Reported by: pj Patches:
+ __20090729-app_voicemail-documentation.patch uploaded by lmadsen
+ (license 10) Tested by: seanbright, lmadsen
+
+2009-09-30 04:32 +0000 [r221044] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_lock.c: Allow locks to be inherited through a
+ masquerade without causing starvation. (closes issue #14859)
+ Reported by: atis Patches: 20090821__issue14859.diff.txt uploaded
+ by tilghman (license 14) 20090925__issue14859__1.6.1.diff.txt
+ uploaded by tilghman (license 14) Tested by: atis, tilghman
+
+2009-09-29 21:28 +0000 [r220920-220995] Mark Michelson <mmichelson@digium.com>
+
+ * main/cel.c: Fix channel reference leak. ast_cel_report_event
+ would geet a reference to the bridged channel. However, certain
+ return paths, such as if CEL was not enabled, would result in a
+ reference leak. All return paths now properly unref the channel.
+ (closes issue #15991) Reported by: mmichelson
+
+ * main/rtp_engine.c: Get rid of annoying and cryptic debug
+ messages.
+
+2009-09-29 19:57 +0000 [r220906] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 220873 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009)
+ | 9 lines Reduce CPU usage related to building a peer merely for
+ devicestates. This fixes a 100% CPU problem in the SIP driver,
+ found by profiling the driver while the problem was occurring.
+ (closes issue #14309) Reported by: pkempgen Patches:
+ 20090924__issue14309.diff.txt uploaded by tilghman (license 14)
+ Tested by: pkempgen, vrban ........
+
+2009-09-29 19:49 +0000 [r220904] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_confbridge.c: Fix options 'm' and 's'. They were swapped
+ in the code. Also document the fact that app_confbridge does not
+ automatically answer the channel. (closes issue #15964) Reported
+ by: shrift
+
+2009-09-29 16:58 +0000 [r220833] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c: Make deletion of temporary greetings work
+ properly with IMAP_STORAGE When imapgreetings was set to yes, the
+ message was being deleted but wasn't actually being expunged.
+ When imapgreetings was set to no, the file based message was not
+ being deleted at all. All good now! (closes issue #14949)
+ Reported by: noahisaac Patches: vm_tempgreeting_removal.patch
+ uploaded by noahisaac (license 748), modified by me
+
+2009-09-28 21:02 +0000 [r220792] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_pri.c: Miscellaneous minor
+ changes.
+
+2009-09-28 19:11 +0000 [r220721] Sean Bright <sean@malleable.com>
+
+ * /, Makefile.rules: Merged revisions 220717 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220717 | seanbright | 2009-09-28 15:09:25 -0400 (Mon, 28 Sep
+ 2009) | 3 lines When selecting DONT_OPTIMIZE in menuselect,
+ explicitly pass -O0 to the compiler so we override any default
+ optimization levels for a particular install. ........
+
+2009-09-28 19:10 +0000 [r220718] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_sip.c: Fix building of registration entry in
+ build_peer when using callbackextension Check for remotesecret
+ option was unintentionally always true, which therefore caused
+ the secret option to never be used. Thanks to dvossel for
+ pointing out the exact fix. (closes issue #15943) Reported by:
+ tpsast
+
+2009-09-28 15:27 +0000 [r220672] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/sig_pri.c: Locking issues dealing
+ with service_lock. * Removed unneeded and uninitialized
+ service_lock. * Fixed potential locking imbalance in
+ pri_dchannel():PRI_EVENT_RESTART. * Fixed verbose message typo in
+ pri_dchannel():PRI_EVENT_RESTART.
+
+2009-09-27 20:40 +0000 [r220629] Michiel van Baak <michiel@vanbaak.info>
+
+ * funcs/func_callerid.c: add name argument for the CALLERID
+ dialplan function to the xml documentation. Pointed out to me on
+ IRC by snuff-home. Thanks
+
+2009-09-26 15:10 +0000 [r220586] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/aes.h: Allow AES to compile, when OpenSSL is not
+ present.
+
+2009-09-25 19:56 +0000 [r220543] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Reduce indentation in sig_pri_available().
+
+2009-09-25 14:50 +0000 [r220494-220496] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/manager.c: Eliminate unnecessary include of version.h in
+ manager.c. Including version.h here causes this file to get
+ recompiled after every commit or update, which is not needed.
+
+ * main/channel.c: Correct sense of logic test committed in revision
+ 220494.
+
+ * main/channel.c: Don't use hash-based lookups for
+ ast_channel_get_by_name_prefix(). ast_channel_get_full() tries to
+ use OBJ_POINTER to optimize name-based channel lookups, but this
+ will not work properly when the channel's full name was not
+ supplied; for name-prefix searches, there is no value in doing a
+ hash-based lookup, and in fact doing so could result in many
+ channels being skipped.
+
+2009-09-25 10:54 +0000 [r220457] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * channels/chan_jingle.c, configs/jabber.conf.sample,
+ include/asterisk/jabber.h, channels/chan_gtalk.c, CHANGES,
+ doc/jabber.txt, res/res_jabber.c: Add JABBER_RECEIVE as a
+ dialplan function, implement SendText in Jingle channels
+ JABBER_RECEIVE (along with JabberSend) makes Asterisk interact
+ with users over XMPP to process calls. SendText can be used
+ instead of JabberSend in the context of XMPP based voice channels
+ (chan_gtalk and chan_jingle). (closes issue #12569) Reported by:
+ eech55 Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo
+ Review: https://reviewboard.asterisk.org/r/88/
+
+2009-09-24 22:53 +0000 [r220417] Tilghman Lesher <tlesher@digium.com>
+
+ * UPGRADE.txt, main/asterisk.c: Change the default behavior of Set,
+ AGI, and pbx_realtime to 1.6 behavior by default (starting in
+ 1.6.3).
+
+2009-09-24 20:37 +0000 [r220365] David Vossel <dvossel@digium.com>
+
+ * main/tcptls.c: fixes tcptls_session memory leak caused by ref
+ count error (closes issue #15939) Reported by: dvossel Review:
+ https://reviewboard.asterisk.org/r/375/
+
+2009-09-24 20:29 +0000 [r220344] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_dial.c, main/features.c, include/asterisk/features.h:
+ Add bridge related dial flags to the bridge app Most of the
+ functionality here is gained simply by setting the feature flag
+ on the bridge config. However, the dial limit functionality has
+ been moved from app_dial to the features code and has been made
+ public so both app_dial and the bridge app can use it. (closes
+ issue #13165) Reported by: tim_ringenbach Patches:
+ app_bridge_options_r138998.diff uploaded by tim ringenbach
+ (license 540), modified by me
+
+2009-09-24 19:57 +0000 [r220295] Olle Johansson <oej@edvina.net>
+
+ * configs/sip.conf.sample: Documentation in the commit messages is
+ soon forgotten, please add it to the docs in the product.
+
+2009-09-24 19:41 +0000 [r220289] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /, apps/app_disa.c, apps/app_playback.c: Merged
+ revisions 220288 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009)
+ | 6 lines Implicitly sending a progress signal breaks some
+ applications. Call Progress() in your dialplan if you explicitly
+ want progress to be sent. (Reverts change 216430, closes issue
+ #15957) Reported by: Pavel Troller on the Asterisk-Dev mailing
+ list
+ http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
+ ........
+
+2009-09-24 18:19 +0000 [r220217] Sean Bright <sean@malleable.com>
+
+ * Makefile, /: Merged revisions 220213 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220213 | seanbright | 2009-09-24 14:18:18 -0400 (Thu, 24 Sep
+ 2009) | 1 line Resolve parallel build warnings. Reported by Klaus
+ Darilion on the asterisk-dev mailing list. ........
+
+2009-09-24 16:33 +0000 [r220174] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Ensure the numeric portion of the
+ P-Asserted-Identity header is properly escaped.
+
+2009-09-24 14:44 +0000 [r220100] Sean Bright <sean@malleable.com>
+
+ * Makefile, build_tools/mkpkgconfig, /: Merged revisions 220099 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220099 | seanbright | 2009-09-24 10:41:57 -0400 (Thu, 24 Sep
+ 2009) | 2 lines Remove the remaining bashisms in the
+ Makefile/mkpkgconfig ........
+
+2009-09-24 08:36 +0000 [r220028] Michiel van Baak <michiel@vanbaak.info>
+
+ * build_tools/mkpkgconfig, /: Merged revisions 220027 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r220027 | mvanbaak | 2009-09-24 10:33:50 +0200 (Thu, 24
+ Sep 2009) | 7 lines mkpkgconfig does not need bash so make it use
+ /bin/sh This fixes building on all systems that don't have bash
+ at /bin/bash Reported by _ys on #asterisk-dev Tested by _ys on
+ #asterisk-dev ........
+
+2009-09-24 07:39 +0000 [r219951-219987] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_directory.c: Fix two possible crashes, one only in 1.6.1
+ and one in 1.6.1 forward. (closes issue #15739) Reported by:
+ DLNoah, jeffg Patches: 20090914__issue15739.diff.txt uploaded by
+ tilghman (license 14) 20090922__issue15739.diff.txt uploaded by
+ tilghman (license 14) Tested by: DLNoah, jeffg
+
+ * configs/mgcp.conf.sample, CHANGES, channels/chan_mgcp.c: Add
+ support for 'setvar=' for MGCP device lines, like other channel
+ drivers provide. (closes issue #14818) Reported by:
+ alea-soluciones Patches:
+ chan_mgcp-setvars-svn-trunk-r219899.patch uploaded by alea
+ (license 514)
+
+ * doc/lang/language-criteria.txt: Update fax number to the legal
+ fax, not the generic fax. (closes issue #15946) Reported by:
+ jtodd Patches: leif-is-a-wuss.txt uploaded by jtodd (license 870)
+ Tested by: jparker, tilghman, jtodd, russellb, mmichelson,
+ seanbright, kpfleming, and the rest of the usual suspects
+
+2009-09-23 17:46 +0000 [r219895] Leif Madsen <lmadsen@digium.com>
+
+ * include/asterisk/doxyref.h,
+ include/asterisk/doxygen/mantisworkflow.h (added): Add Mantis
+ work flow documention. This commit adds the doxygen changes that
+ I've made to describe the Mantis work flow documentation for the
+ open source issue tracker. This should make it easier to
+ determine the flow of issues through the issue tracker, and what
+ those statuses mean. (closes issue #15902) Reported by: lmadsen
+ Patches: mantisworkflow.h uploaded by lmadsen (license 10)
+ Review: https://reviewboard.asterisk.org/r/367/
+
+2009-09-22 21:43 +0000 [r219818] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 219816 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22
+ Sep 2009) | 10 lines When IMAP variables were changed during a
+ reload, Voicemail did not use the new values. This change
+ introduces a configuration version variable, which ensures that
+ connections with the old values are not reused but are allowed to
+ expire normally. (closes issue #15934) Reported by:
+ viniciusfontes Patches: 20090922__issue15934.diff.txt uploaded by
+ tilghman (license 14) Tested by: viniciusfontes ........
+
+2009-09-21 16:59 +0000 [r219721] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 219720 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r219720 | dvossel | 2009-09-21 11:55:53 -0500 (Mon, 21
+ Sep 2009) | 3 lines Reverting merge 219520. This change was not
+ necessary. ........
+
+2009-09-20 17:55 +0000 [r219654] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/file.c: Merged revisions 219653 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009)
+ | 8 lines Really stop the stream, when ast_closestream() is
+ called. (closes issue #15129) Reported by: bmh Patches:
+ 20090918__issue15129.diff.txt uploaded by tilghman (license 14)
+ Review: https://reviewboard.asterisk.org/r/372/ ........
+
+2009-09-19 02:59 +0000 [r219587] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 219586 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r219586 | russell | 2009-09-18 21:51:13 -0500 (Fri, 18
+ Sep 2009) | 6 lines Make sure the iax_pvt exists before
+ dereferencing it. This fixes the latest crash posted on issue
+ 15609. (issue #15609) ........
+
+2009-09-18 23:20 +0000 [r219451-219520] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 219519 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18
+ Sep 2009) | 9 lines iax2 frame double free The iax frame's
+ retrans sched id was written over right before iax2_frame_free
+ was called. In iax2_frame_free that retrans id is used to delete
+ the sched item. By writing over the retrans field before the
+ sched item could be deleted, it was possible for a retransmit to
+ occur on a freed frame. ........
+
+ * /, channels/chan_sip.c: Merged revisions 219450 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009)
+ | 14 lines via-header branches not updated correctly on INVITE
+ INVITE requests must always contain a new unique branch id. When
+ a new branch id is created for an INVITE, the dialog's
+ invite_branch variable must be updated so CANCEL requests use the
+ correct branch id. (closes issue #15262) Reported by: maniax
+ Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety
+ (license 608) invite_new_branch_trunk.diff uploaded by dvossel
+ (license 671) Tested by: maniax, dvossel ........
+
+2009-09-18 13:54 +0000 [r219412] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Missing value setting line for
+ maxsecs/maxmessage (closes issue #15696) Reported by:
+ fhackenberger Patches: maxsecs.patch uploaded by fhackenberger
+ (license 592)
+
+2009-09-17 22:37 +0000 [r219371] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes deadlock when performing directed
+ pickup w Invite/replaces (closes issue #15340) Reported by:
+ lmsteffan Patches: deadlock.patch uploaded by lmsteffan (license
+ 779) Tested by: lmsteffan
+
+2009-09-17 22:22 +0000 [r219324] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 219320 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219320 | mmichelson | 2009-09-17 17:20:50 -0500 (Thu, 17 Sep
+ 2009) | 6 lines Send a 100 Trying response when we detect a
+ spiral. This was problematic during spiral tests at SIPit...
+ along with some other things as well. ........
+
+2009-09-17 21:59 +0000 [r219304] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 219303 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009)
+ | 21 lines INVITE w/Replaces deadlock fix This patch cleans up
+ the locking logic in chan_sip.c's handle_invite_replaces()
+ function as well as making use of ast_do_masquerade() rather than
+ forcing the masquerade on an ast_read(). The code had several
+ redundant unlocks that would result in 'freed more times than
+ we've locked!' errors. I cleaned these up as well as moving all
+ the unlock logic to the end of the function. This patch should
+ also resolve the issue people were having with the replacecall
+ channel never being unlocked with one legged calls. (closes issue
+ #15151) Reported by: irroot Patches: invite_w_replaces_1.4.diff
+ uploaded by dvossel (license 671) Tested by: irroot, dvossel
+ Review: https://reviewboard.asterisk.org/r/371/ ........
+
+2009-09-17 19:57 +0000 [r219264] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Ensure no spaces exist before "refresher="
+ when doing the comparison.
+
+2009-09-17 16:25 +0000 [r219230] Sean Bright <sean@malleable.com>
+
+ * apps/app_chanspy.c: Get this compiling under dev-mode.
+
+2009-09-17 15:18 +0000 [r219139] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, /, include/asterisk/cdr.h: Merged revisions
+ 219136 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep
+ 2009) | 10 lines Prevent a potential race condition and crash
+ when hanging up a channel by removing the channel from the
+ channel list before begining channel tear down. This fix may
+ potentially cause problems with CDR backends that access the
+ channel a CDR is associated with via the channel list. This fix
+ makes the channel unavabile at the time when the CDR backend is
+ invoked. This has been documented in include/asterisk/cdr.h.
+ (closes issue #15316) Reported by: vmarrone Tested by: mnicholson
+ Review: https://reviewboard.asterisk.org/r/362/ ........
+
+2009-09-17 00:58 +0000 [r219007-219105] Tilghman Lesher <tlesher@digium.com>
+
+ * CHANGES, apps/app_chanspy.c: Add the 'E' option to exit ChanSpy,
+ once the single channel it spied upon hangs up. In addition,
+ there's a bit of cleanup to the arguments and documentation, in
+ which I discovered that the last feature added to this
+ application duplicated an option (oops!) and changed that option
+ so that it now works. (closes issue #14909) Reported by: junky
+ Patches: __20090901-spy_hangup_trunk.diff uploaded by lmadsen
+ (license 10) Tested by: amilcar, junky, flujan, lmadsen
+
+ * /, main/config.c, configs/extensions.conf.sample: Merged
+ revisions 219023 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009)
+ | 8 lines Properly deal with quotes in the arguments of '#exec'
+ includes. (closes issue #15583) Reported by: pkempgen Patches:
+ 20090726__issue15583.diff.txt uploaded by tilghman (license 14)
+ 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license
+ 169) Tested by: pkempgen ........
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac: Detect
+ whether we actually have the long double type, before looking for
+ those functions. (closes issue #15017) Reported by: tzafrir
+ Patches: 20090916__issue15017.diff.txt uploaded by tilghman
+ (license 14) Tested by: tzafrir
+
+2009-09-16 20:32 +0000 [r218973] Sean Bright <sean@malleable.com>
+
+ * res/res_jabber.c: Remove some unused defines from res_jabber.
+ (closes issue #15359) Reported by: snuffy Patches:
+ bug_res_jabber_unused_defines.diff uploaded by snuffy (license
+ 35)
+
+2009-09-16 19:25 +0000 [r218933] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Reverse order of args to fread. This way, we
+ don't always write a null byte into byte 1 of the buffer (closes
+ issue #15905) Reported by: ebroad Patches: freadfix.patch
+ uploaded by ebroad (license 878) Tested by: ebroad
+
+2009-09-16 18:31 +0000 [r218918] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: On TCP and TLS connections do not attempt to
+ stop retransmission of the packet internally. This was preventing
+ responses from being properly processed because the packet was
+ not being found causing handle_response to return prematurely.
+
+2009-09-16 18:06 +0000 [r218868] David Brooks <dbrooks@digium.com>
+
+ * main/pbx.c, /: Merged revisions 218867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009)
+ | 13 lines Fixes CID pattern matching behavior to mirror that of
+ extension pattern matching. Pattern matching for extensions uses
+ a type of scoring system, giving values for specificity to each
+ character in the pattern. Unfortunately, this is done character
+ by character, in order. This does lead to some less specific
+ patterns being first in line for matching, but it will usually
+ get the job done. This patch merely brings CID matching to the
+ same level as extension matching. This patch does not attempt to
+ tackle the problem shared by extension matching. (closes issue
+ #14708) Reported by: klaus3000 ........
+
+2009-09-16 13:34 +0000 [r218799] Russell Bryant <russell@digium.com>
+
+ * contrib/firmware/iax/iaxy.bin (removed), /, UPGRADE.txt: Merged
+ revisions 218798 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009)
+ | 9 lines Remove the IAXy firmware from Asterisk. The firmware
+ can now be found on downloads.digium.com, where the rest of our
+ binary downloads live. This was the last part of our Asterisk
+ tarballs that was considered non-free by Debian. :-) (closes
+ issue #15838) Reported by: paravoid ........
+
+2009-09-15 22:33 +0000 [r218731] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 218730 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15
+ Sep 2009) | 6 lines If the user enters the same password as
+ before, don't signal an error when the change does nothing.
+ (closes issue #15492) Reported by: cbbs70a Patches:
+ 20090713__issue15492.diff.txt uploaded by tilghman (license 14)
+ ........
+
+2009-09-15 19:22 +0000 [r218687] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: upward bound checking for port string to int
+ conversion
+
+2009-09-15 16:15 +0000 [r218586] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218578 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218578 | mnicholson | 2009-09-15 11:03:54 -0500 (Tue, 15 Sep
+ 2009) | 8 lines Send request contact header field with response
+ to registrer queries instead of the address of record. (closes
+ issue #14438) Reported by: ravindrad Patches: regquerypatch
+ uploaded by ravindrad (license 684) Tested by: ravindrad ........
+
+2009-09-15 16:12 +0000 [r218583] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Add some changes related to 218430. *
+ Remove thread_spawned in handle_init_event since it was never
+ used * Always check handle_init_event in case a channel is
+ destroyed
+
+2009-09-15 16:04 +0000 [r218579] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_followme.c: Merged revisions 218577 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009)
+ | 9 lines Ensure FollowMe sets language in channels it creates.
+ Also, not in the original bug report, but related fields are
+ accountcode and musicclass, and the inheritance of datastores.
+ (closes issue #15372) Reported by: Romik Patches:
+ 20090828__issue15372.diff.txt uploaded by tilghman (license 14)
+ Tested by: cervajs ........
+
+2009-09-15 15:40 +0000 [r218504-218566] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Use a better method of ensuring
+ null-termination of the buffer while reading the SDP when using
+ TCP.
+
+ * channels/chan_sip.c: Ensure that SDP read from TCP socket is
+ null-terminated.
+
+2009-09-15 15:02 +0000 [r218500] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /: Merged revisions 218497 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218497 | kpfleming | 2009-09-15 10:55:58 -0400 (Tue, 15 Sep
+ 2009) | 1 line Use proper hostname for downloading sound files.
+ ........
+
+2009-09-15 14:59 +0000 [r218499] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix off-by-one error when reading SDP sent
+ over TCP.
+
+2009-09-15 10:24 +0000 [r218465] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_dahdi.c: Fix false error message on
+ DAHDI_EVENT_REMOVED (RESULT_SUCCESS == 0)
+
+2009-09-14 22:38 +0000 [r218430] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/sig_analog.h: Merged revisions 218401 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009)
+ | 11 lines Fix handling of DAHDI_EVENT_REMOVED event to prevent
+ crash in do_monitor. After talking to rmudgett about some of his
+ recent iflist locking changes, it was determined that the only
+ place that would destroy a channel without being explicitly to do
+ so was in handle_init_event. The loop to walk the interface list
+ has been modified to wait to destroy the channel until the
+ dahdi_pvt of the channel to be destroyed is no longer needed.
+ (closes issue #15378) Reported by: samy ........
+
+2009-09-14 20:08 +0000 [r218365] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Add support for multiple interface lists.
+ Also unlink the sig_pri_pri.pvts[] pointer in
+ destroy_dahdi_pvt().
+
+2009-09-14 19:29 +0000 [r218361] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configs/voicemail.conf.sample, sounds/Makefile,
+ apps/app_voicemail.c: Recorded merge of revisions 218331 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009)
+ | 4 lines Don't say "Please try again" if we don't give the user
+ another chance to try again. (issue #15055, SWP-129) Reported by:
+ jthurman ........
+
+2009-09-14 18:16 +0000 [r218295] Joshua Colp <jcolp@digium.com>
+
+ * main/features.c: Do not attempt to add a parking extension if an
+ error occurred while reading the configuration.
+
+2009-09-14 14:57 +0000 [r218224] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, apps/app_directed_pickup.c: Merged revisions 218223 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep
+ 2009) | 8 lines Ensure we don't pickup ourselves when doing
+ pickup by exten. (closes issue #15100) Reported by: lmsteffan
+ Patches: (modified) pickup.patch uploaded by lmsteffan (license
+ 779) ........
+
+2009-09-13 17:34 +0000 [r218184] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_phone.c: gcc 4.4: Remove a nop memset size 0 that
+ annoys gcc This memset doesn't write beyond the end of the
+ buffer. (tmpbuf has size of 4).
+
+2009-09-13 05:51 +0000 [r218150] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c: get rid of mfcr2 monitor thread condition,
+ is problematic
+
+2009-09-12 13:08 +0000 [r218107] Michiel van Baak <michiel@vanbaak.info>
+
+ * res/res_rtp_asterisk.c: use the actual given ip address for 'rtp
+ set debug ip <foo>' instead of the word 'ip' (closes issue
+ #15711) Reported by: davidw Patches: 2009082800-rtpdebug.diff.txt
+ uploaded by mvanbaak (license 7) Tested by: davidw
+
+2009-09-11 05:58 +0000 [r217990-218050] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Check the origination priority for more matches, not
+ the current priority. Found by Pavel Troller on the -dev list.
+
+ * /, apps/app_queue.c: Merged revisions 217989 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009)
+ | 3 lines Don't ring another channel, if there's not enough time
+ for a queue member to answer. (Fixes AST-228) ........
+
+2009-09-10 23:49 +0000 [r217954-217987] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
+ Cleanup approach in 217804 and don't reach inside the sig_pvt.
+
+ * channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h: Allow do not disturb to be set on analog
+ channels via the CLI and AMI.
+
+2009-09-10 23:12 +0000 [r217916] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/iax-friends.sql, channels/chan_sip.c,
+ channels/chan_iax2.c: Make calltoken support work with realtime
+ users and peers. In the course of this, I also found that the
+ results of ast_gethostbyname were being used incorrectly in both
+ chan_iax2 and chan_sip, so both have been fixed.
+
+2009-09-10 22:31 +0000 [r217873-217912] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Cleaned up chan_dahdi iflist handling and
+ locking. * Fixed walking the iflist so it is always done with the
+ iflock locked. * Simplified iflist walking routines. * Created
+ chan_dahdi iflist insertion and extraction routines. * Fixed
+ duplicate_pseudo() malloc fail handling. * Fixed infinite loop in
+ action_dahdishowchannels() when showing a single channel.
+
+ * channels/chan_dahdi.c: Miscellaneous minor changes.
+
+2009-09-10 21:07 +0000 [r217807] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 217806 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10
+ Sep 2009) | 22 lines IAX2 encryption regression The IAX2 Call
+ Token security patch inadvertently broke the use of encryption
+ due to the reorganization of code in the socket_process()
+ function. When encryption is used, an incoming full frame must
+ first be decrypted before the information elements can be parsed.
+ The security release mistakenly moved IE parsing before
+ decryption in order to process the new Call Token IE. To resolve
+ this, decryption of full frames is once again done before looking
+ into the frame. This involves searching for an existing callno,
+ checking the pvt to see if encryption is turned on, and
+ decrypting the packet before the internal fields of the full
+ frame are accessed. (closes issue #15834) Reported by: karesmakro
+ Patches: iax2_encryption_fix_1.4.diff uploaded by dvossel
+ (license 671) Tested by: dvossel, karesmakro Review:
+ https://reviewboard.asterisk.org/r/355/ ........
+
+2009-09-10 20:52 +0000 [r217744-217804] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Fix crash during attended transfer over
+ PRI. The owner pointers in the sig_pri_chan structure were not
+ getting updated in dahdi_fixup.
+
+ * channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h: Stop caller id transmission when offhook
+ event detected. This fixes the problem that would occur if an
+ analog phone was picked up while the caller id was being sent.
+ The caller id before sent the whole spill even after pickup and
+ is now corrected.
+
+2009-09-10 19:39 +0000 [r217730] Matthias Nick <mnick@digium.com>
+
+ * res/res_musiconhold.c: Sets the correct musicclass after an
+ announcement (closes issue #15279) Reported by: mbeckwell
+ Patches: patch.txt uploaded by mnick (license ) Tested by: mnick
+ (closes issue #15832) Reported by: mbeckwell Patches: patch.txt
+ uploaded by mnick (license 874) Tested by: mnick
+
+2009-09-10 18:29 +0000 [r217663] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Don't assign UINT_MAX to an INT.
+
+2009-09-10 18:17 +0000 [r217638] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_odbc.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Verify support
+ for wide ODBC character types before using them. (closes issue
+ #15870) Reported by: nic_bellamy
+
+2009-09-10 12:06 +0000 [r217593] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Include ActionID in all events that are
+ responsed to AMI Action SIPShowRegistry (closes issue #15868)
+ Reported by: nic_bellamy Patches:
+ manager_SIPshowregistry_actionid.patch uploaded by nic bellamy
+ (license 299)
+
+2009-09-10 00:35 +0000 [r217560] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Fix available() for SS7, MFC/R2, and
+ pseudo channels.
+
+2009-09-09 21:48 +0000 [r217524] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c: ast_log replaced for ast_verbose in MFCR2
+ event notifications
+
+2009-09-09 20:09 +0000 [r217482] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Don't report transfer success until we
+ actually know. 1xx messages are not final. Related to #12713
+ Patch by oej A big thank you to file for finally fixing the
+ transfer() dialplan application. I've been waiting for years for
+ this. Great work!
+
+2009-09-09 18:52 +0000 [r217445] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * res/res_phoneprov.c: gcc 4.4 fix: union instead of cast gcc 4.4
+ has more strict rules for aliasing. It doesn't like a struct
+ sockaddr_in pointer pointing to a struct sockaddr. So we make it
+ a union.
+
+2009-09-09 12:11 +0000 [r217408] Sean Bright <sean@malleable.com>
+
+ * main/manager.c: Properly terminate the response to the manager
+ Ping action. In passing, correct the formatting of the Timestamp
+ attribute so that there is a space after the colon and before the
+ value. (closes issue #15861) Reported by: Ivan
+
+2009-09-09 10:39 +0000 [r217367-217368] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Not having any TLS session to write to is a
+ serious XMIT_ERROR.
+
+ * channels/chan_sip.c: Formatting and doxygen updates
+
+2009-09-08 23:37 +0000 [r217331-217332] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h, channels/sig_pri.c: Fix memory leak of
+ sig_xxx private structures.
+
+ * channels/chan_dahdi.c: Miscellaneous minor code cleanup in
+ mkintf().
+
+2009-09-08 22:17 +0000 [r217286] Sean Bright <sean@malleable.com>
+
+ * apps/app_meetme.c: Fix compilation of app_meetme. Reported by
+ ebroad in #asterisk-bugs
+
+2009-09-08 21:17 +0000 [r217236] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Remove duplicate entry in the sig_pri_pri
+ private pointer array.
+
+2009-09-08 20:28 +0000 [r217199] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 217156 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009)
+ | 7 lines When MOH is playing on the channel, announcements sent
+ through the conference are not heard. (closes issue #14588)
+ Reported by: voipas Patches: 20090716__issue14588__2.diff.txt
+ uploaded by tilghman (license 14) Tested by: lmadsen, twisted,
+ tilghman ........
+
+2009-09-08 20:06 +0000 [r217158] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/event.h: Add doxygen to ast_event_subscribe for
+ the description. Most importantly, note that a NULL description
+ will cause a crash, as I just experienced that firsthand.
+
+2009-09-08 18:06 +0000 [r217113] Russell Bryant <russell@digium.com>
+
+ * addons/format_mp3.c: Fix audio problems with format_mp3. This
+ problem was introduced when the AST_FRIENDLY_OFFSET patch was
+ merged. I'm surprised that nobody noticed any trouble when
+ testing that patch, but this fixes the code that fills in the
+ buffer to start filling in after the offset portion of the
+ buffer. (closes issue #15850) Reported by: 99gixxer Patches:
+ issue15850.diff1.txt uploaded by russell (license 2) Tested by:
+ 99gixxer
+
+2009-09-08 16:37 +0000 [r217074] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac: Ensure
+ that the default autoconf CFLAGS are not used. A recent change to
+ the configure script that allows the user to specify CFLAGS
+ and/or LDFLAGS to the script had the unfortunate side effect of
+ letting autoconf's default CFLAGS (-g -O2) feed in to the rest of
+ the build system, thereby overriding the DONT_OPTIMIZE setting in
+ menuselect. That problem is now corrected.
+
+2009-09-08 15:30 +0000 [r217033] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_limit.c: Remove what appears to be an unnecessary define.
+ (closes issue #15851) Reported by: tzafrir
+
+2009-09-08 15:23 +0000 [r217015] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * contrib/scripts/live_ast: live_ast: Fix asterisk.conf instead of
+ regenerating it * Don't write asterisk.conf from scratch. Fix the
+ existing one. * Pass extra 'make' command-line arguments to
+ 'install' and 'samples'. * Fix some extra typos. closes issue
+ #15019 .
+
+2009-09-08 14:26 +0000 [r216993] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: caller id number empty parse_uri was not
+ being given the correct scheme's, as a result, uri parsing did
+ not parse the username correctly. One of the side effects of this
+ is an empty caller id. (closes issue #15839) Reported by: ebroad
+ Patches: blank_cidv2.patch uploaded by ebroad (license 878)
+ parse_uri_fix.diff uploaded by dvossel (license 671) Tested by:
+ ebroad, dvossel
+
+2009-09-07 20:23 +0000 [r216883-216956] Olle Johansson <oej@edvina.net>
+
+ * doc/manager_1_1.txt: Fixing formatting
+
+ * doc/manager_1_1.txt: Add new actions under "new actions" and not
+ in the top of the document
+
+ * channels/chan_sip.c: Moving another function declared in the
+ middle of forward declarations. Please follow the structure of
+ the source code, thanks. Chan_sip is messy enough as it is :-)
+
+ * channels/chan_sip.c: Move "deprecated_username" to a flag like
+ the others - unsigned int blah:1
+
+ * channels/chan_sip.c: - Doxygen additions - Remove unused string
+ in sip_registry -- "random" - Someone added a function in the
+ middle of all forward declarations... Weird. Moved it out of that
+ section.
+
+ * channels/chan_sip.c: Clean up the "offered_media" code - Add
+ variable for number of known media streams instead of hardcoding
+ in definition of sip_pvt - Rename "text" to "codecs" - beacuse
+ it's what it is - Add documentation for future developers so that
+ we make sure that we define new sdp media types for SRTP-variants
+
+2009-09-07 17:15 +0000 [r216846] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/func_odbc.conf.sample, funcs/func_odbc.c, CHANGES: Allow
+ multiple rows to be fetched within the normal mode of operation.
+
+2009-09-07 16:35 +0000 [r216652-216842] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Make sure we reset global_exclude_static at
+ channel reload
+
+ * channels/chan_sip.c: Move capability into sip_cfg. While at it,
+ make sure we reset it at channel reload.
+
+ * channels/chan_sip.c: Move global_regcontext into the sip_cfg
+ structure
+
+ * channels/chan_sip.c: Move contact_ha to sip_cfg structure
+
+ * channels/chan_sip.c: Doxygen updates
+
+ * channels/chan_sip.c: Since it's possible to have more than 999
+ calls, I'm changing the call counter roof to something higher.
+
+ * channels/chan_sip.c: add doxygen and remove duplicate declaration
+ of variable
+
+ * channels/chan_sip.c: After many years, remove VOCAL_DATA_HACK
+ definition
+
+ * channels/chan_sip.c: Remove unneeded header files (tested on
+ Linux and OS/X)
+
+ * channels/chan_sip.c: Don't send MESSAGE with sendtext() if
+ recepient doesn't allow MESSAGE requests
+
+ * channels/chan_sip.c: Add some doxygen
+
+ * channels/chan_sip.c: Fix typo
+
+ * channels/chan_sip.c: If there is no session timer in the INVITE,
+ set it to default value (not unset minimum = -1) Patch by oej
+ closes issue #15621 Reported by: fnordian Tested by: atis
+
+ * configs/sip.conf.sample: Update sip.conf.sample documentation,
+ reorganize a bit
+
+ * channels/chan_sip.c: Simplify the code in this function
+
+2009-09-04 19:32 +0000 [r216594] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: sip peer matching by address only with
+ TCP/TLS This patch removes the contact header matching logic and
+ adds logic to match all tcp/tls connections by ip only. Thanks to
+ oej for finding the issue and suggesting solutions. Review:
+ https://reviewboard.asterisk.org/r/354/
+
+2009-09-04 19:29 +0000 [r216593] Sean Bright <sean@malleable.com>
+
+ * apps/app_voicemail.c: Use ast_free() instead of free().
+
+2009-09-04 17:50 +0000 [r216547-216551] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/lock.h: Fix trunk breakage.
+
+ * main/pbx.c, UPGRADE-1.6.txt: Enable turning off the application
+ delimiter warning with the 'dontwarn' option. Suggested on the
+ -dev list, and implemented in an alternate way by me.
+
+2009-09-04 15:05 +0000 [r216506] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, main/utils.c: Merged revisions 216435 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009)
+ | 2 lines make asterisk compile under devmode with DEBUG_THREADS
+ enabled on OpenBSD ........
+
+2009-09-04 14:02 +0000 [r216438] Olle Johansson <oej@edvina.net>
+
+ * main/pbx.c, /, channels/chan_sip.c, apps/app_disa.c,
+ configs/sip.conf.sample, apps/app_playback.c: Merged revisions
+ 216430 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27
+ lines Make apps send PROGRESS control frame for early media and
+ fix too early media issue in SIP The issue at hand is that some
+ legacy (dying) PBX systems send empty media frames on PRI links
+ *before* any call progress. The SIP channel receives these frames
+ and by default signals 183 Session progress and starts sending
+ media. This will cause phones to play silence and ignore the
+ later 180 ringing message. A bad user experience. The fix is
+ twofold: - We discovered that asterisk apps that support early
+ media ("noanswer") did not send any PROGRESS frame to indicate
+ early media. Fixed. - We introduce a setting in chan_sip so that
+ users can disable any relay of media frames before the outbound
+ channel actually indicates any sort of call progress. In 1.4,
+ 1.6.0 and 1.6.1, this will be disabled for backward
+ compatibility. In later versions of Asterisk, this will be
+ enabled. We don't assume that it will change your Asterisk phone
+ experience - only for the better. We encourage third-party
+ application developers to make sure that if they have
+ applications that wants to send early media, add a PROGRESS
+ control frame transmission to make sure that all channel drivers
+ actually will start sending early media. This has not been the
+ default in Asterisk previous to this patch, so if you got
+ inspiration from our code, you need to update accordingly. Sorry
+ for the trouble and thanks for your support. This code has been
+ running for a few months in a large scale installation (over 250
+ servers with PRI and/or BRI links to old PBX systems). That's no
+ proof that this is an excellent patch, but, well, it's tested :-)
+ ........
+
+2009-09-04 14:00 +0000 [r216431-216437] Michiel van Baak <michiel@vanbaak.info>
+
+ * include/asterisk/lock.h: make sure canlog is set so we can
+ compile with DEBUG_THREADS enabled on OpenBSD
+
+ * /: Recorded merge of revisions 216432 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r216432 | mvanbaak | 2009-09-04 15:53:09 +0200 (Fri, 04 Sep 2009)
+ | 2 lines make chan_sip compile under devmode again ........
+
+ * /: Recorded merge of revisions 216369 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r216369 | mvanbaak | 2009-09-04 15:16:29 +0200 (Fri, 04 Sep 2009)
+ | 4 lines Make sure 'start' is always initialized. This is the
+ same as rev 216222 in trunk but 1.4 is affected as well ........
+
+2009-09-04 13:14 +0000 [r216368] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Do not treat every SIP peer as if they were
+ configured with insecure=port. There was a problem in the
+ function responsible for doing peer matching by IP address and
+ port number such that during the second pass for checking for a
+ peer configured with insecure=port, it would end up treating
+ every peer as if it had been configured that way. These changes
+ fix the logic in the peer IP and port comparison callback to
+ handle insecure=port checking properly. This problem was
+ introduced when SIP peers were converted to astobj2. Many thanks
+ to dvossel for noticing this while working on another peer
+ matching issue.
+
+2009-09-04 12:05 +0000 [r216335] Olle Johansson <oej@edvina.net>
+
+ * doc/janitor-projects.txt: Adding to the janitor list. For new
+ readers: The janitor list is a list of tasks we need help with in
+ the Asterisk project. Taking up one of these is often a good way
+ to get into Asterisk development and getting a lot of developers
+ in the project to be grateful. It's stuff we could spend time on
+ when the bug tracker is empty, when our employers hasn't filled
+ our task lists and our servers is running bugfree and happily
+ without any issues. If you want to start working on one of these
+ small projects, feel free to ask for help in the #asterisk-dev
+ channel on IRC or asterisk-dev mailing list. We'll be more than
+ happy to help you to start and reach goal. Thank you for your
+ help. </end of long commit message>
+
+2009-09-04 10:48 +0000 [r216264] Russell Bryant <russell@digium.com>
+
+ * /, doc/IAX2-security.txt (added): Merged revisions 216263 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r216263 | russell | 2009-09-04 05:48:00 -0500
+ (Fri, 04 Sep 2009) | 9 lines Merged revisions 216262 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04
+ Sep 2009) | 2 lines Add a plain text version of the IAX2 security
+ document. ........ ................
+
+2009-09-04 06:08 +0000 [r216222] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/astobj2.c: make sure 'start' is always initialized. Makes
+ asterisk compile with --enable-dev-mode
+
+2009-09-03 21:09 +0000 [r216186] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_pri.c: Lets try not to use
+ C++ keywords for variable names.
+
+2009-09-03 19:40 +0000 [r216094] Doug Bailey <dbailey@digium.com>
+
+ * include/asterisk/callerid.h, channels/chan_dahdi.c,
+ channels/sig_analog.c, channels/sig_analog.h: Added detection
+ DTMF CID without polarity change alert. Added detection of DTMF
+ tone energy levels on FXO channels in chan_dahdi monitoring loop
+ so DTMF CID can be detected without the need of a polarity change
+ precursor. (closes issue #9096) Reported by: fleed Patches:
+ 9096-chan_dahdi-trunk.diff uploaded by dbailey (license 819)
+ Tested by: cyberplant, sum, maturs
+
+2009-09-03 19:38 +0000 [r216009-216092] Russell Bryant <russell@digium.com>
+
+ * /, UPGRADE.txt: Merged revisions 216085 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r216085 | russell | 2009-09-03 14:36:46 -0500
+ (Thu, 03 Sep 2009) | 9 lines Merged revisions 216080 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03
+ Sep 2009) | 2 lines Add a note about IAX2 to UPGRADE.txt.
+ ........ ................
+
+ * /, doc/IAX2-security.pdf (added): Merged revisions 216008 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r216008 | russell | 2009-09-03 13:44:58 -0500
+ (Thu, 03 Sep 2009) | 9 lines Merged revisions 216005 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03
+ Sep 2009) | 2 lines Add IAX2 security document related to
+ AST-2009-006. ........ ................
+
+2009-09-03 18:42 +0000 [r216006] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/file.c, doc/lang/language-criteria.txt (added): Document
+ language prompt submission process. This patch adds a document
+ describing the language prompt submission process, licensing
+ terms and other issues related to that process. In addition, it
+ modifies the sound file searching process to support language
+ codes with any number of suffices (not limited to just "xx" or
+ "xx_YY"), so that prompts can be named with gender,
+ customer/company, etc. suffices as well. (closes issue #15771)
+ Reported by: jtodd Patches: language-criteria.txt uploaded by
+ jtodd
+
+2009-09-03 16:31 +0000 [r215955] David Vossel <dvossel@digium.com>
+
+ * configs/iax.conf.sample, include/asterisk/acl.h,
+ channels/iax2-parser.h, include/asterisk/astobj2.h,
+ channels/iax2.h, main/acl.c, channels/chan_iax2.c,
+ channels/iax2-parser.c, main/astobj2.c: Merge code associated
+ with AST-2009-006 (closes issue #12912) Reported by: rathaus
+ Tested by: tilghman, russell, dvossel, dbrooks
+
+2009-09-03 13:02 +0000 [r215891] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Add known internal IP address when
+ autodomain=yes (closes issue #14573) Reported by: pj Patches:
+ sip-internip-autodomain1.diff uploaded by mnicholson (license 96)
+ modified by oej Tested by: pj
+
+2009-09-03 05:57 +0000 [r215838] Michiel van Baak <michiel@vanbaak.info>
+
+ * doc/manager_1_1.txt: Document that SIPshowpeer and SKINNYshowline
+ now include the configured parkinglot in their response. Prodded
+ by snuff-work on #asterisk-dev IRC channel
+
+2009-09-03 03:43 +0000 [r215800-215801] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Default the callback extension to "s". This
+ is a regression. (closes issue #15764) Reported by: elguero
+ Change-type: bugfix
+
+ * include/asterisk.h: Revert attempt to standardize with
+ _POSIX_C_SOURCE. This did not function in the way that was
+ intended, causing more compatibility issues than it solved. It is
+ best, therefore, that it be simply removed. (Discussed with
+ kpfleming; agreement to remove was reached.)
+
+2009-09-02 23:31 +0000 [r215758] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 215682 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r215682 | twilson | 2009-09-02 16:41:22 -0500 (Wed, 02 Sep 2009)
+ | 18 lines Re-send non-100 provisional responses to prevent
+ cancellation From section 13.3.1.1 of RFC 3261: If the UAS
+ desires an extended period of time to answer the INVITE, it will
+ need to ask for an "extension" in order to prevent proxies from
+ canceling the transaction. A proxy has the option of canceling a
+ transaction when there is a gap of 3 minutes between responses in
+ a transaction. To prevent cancellation, the UAS MUST send a
+ non-100 provisional response at every minute, to handle the
+ possibility of lost provisional responses. (closes issue #11157)
+ Reported by: rjain Tested by: twilson Review:
+ https://reviewboard.asterisk.org/r/315/ ........
+
+2009-09-02 23:25 +0000 [r215757] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, CHANGES, channels/sig_pri.c: Made
+ chan_dahdi able to ignore incoming calls that are not in a MSN
+ list for ISDN PTMP CPE spans.
+
+2009-09-02 21:39 +0000 [r215681] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: port string to int conversion using sscanf
+ There are several instances where a port is parsed from a uri or
+ some other source and converted to an int value using atoi(), if
+ for some reason the port string is empty, then a standard port is
+ used. This logic is used over and over, so I created a function
+ to handle it in a safer way using sscanf().
+
+2009-09-02 21:23 +0000 [r215622-215665] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_sip.c, channels/chan_skinny.c: add Parkinglot info
+ to sip show peer <foo> and skinny show line <foo> If we had this
+ from the start, debugging the 'parking not using configured
+ parkinglot' bug would have been easier.
+
+ * main/features.c: - lock channel before looking for a channel
+ variable - Init the parkings list member of struct parkinglot.
+ Thanks Sean for the explanation why this should be here.
+
+2009-09-02 19:49 +0000 [r215608] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c: Fix issue where
+ DTMF CID detect was placing channels into signed linear mode made
+ analog_set_linear_mode return back the mode that was being
+ overwritten so it could be restored later.
+
+2009-09-02 18:37 +0000 [r215567] Tilghman Lesher <tlesher@digium.com>
+
+ * main/Makefile, main/app.c: Close up to the soft open file limit
+ (same on Linux, but varies drastically on OS X). Also, a Makefile
+ fix for Darwin (OS X). (closes issue #14542) Reported by: jtodd
+ Patches: 20090901__issue14542.diff.txt uploaded by tilghman
+ (license 14) Tested by: jtodd, tilghman Change-type: bugfix
+
+2009-09-02 17:26 +0000 [r215522] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: SIP uri parsing cleanup Now, the scheme
+ passed to parse_uri can either be a single scheme, or a list of
+ schemes ',' delimited. This gets rid of the whole problem of
+ having to create two buffers and calling parse_uri twice to check
+ for separate schemes. Review:
+ https://reviewboard.asterisk.org/r/343/
+
+2009-09-02 16:20 +0000 [r215479] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: like in chan_sip's sip_new skinny should
+ copy the configured parkinglot from a line to the newly created
+ channel. This makes callparking honor the configured parkinglot
+ for skinny lines as well.
+
+2009-09-02 16:08 +0000 [r215466] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: SIP support for keep-alive event keep-alive
+ events are used by Sipura/Linksys for NAT keepalive. There
+ currently don't appear to be any problems with NAT, but everytime
+ a keep-alive event is received, Asterisk responds with a "489 Bad
+ event". This error may indicate to a user that NAT problems exist
+ just because this even is not supported. Now, rather than respond
+ with an error, the packet is consumed and a "200 ok" is sent just
+ to indicate we received the packet. (issue #15084) Patches:
+ chan_sip.keepalive.v1.diff uploaded by IgorG (license 20)
+
+2009-09-02 15:56 +0000 [r215419-215462] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_sip.c: Honor configured parkinglot when parking and
+ retrieving parked calls Thank oej for pointing out the fact that
+ sip_new did not copy parkinglot from the peer into the newly
+ created channel. (closes issue #15538) Reported by: gracedman
+ Patches: 2009090100_sipnewparkinglot-161.diff.txt uploaded by
+ mvanbaak (license 7) With mod by me to also fix callparking as
+ well (this uploaded patch only fixed retrieving a parked call)
+ Tested by: gracedman, mvanbaak
+
+ * include/asterisk.h: Let's compile again on OpenBSD
+
+2009-09-02 06:23 +0000 [r215382] Olle Johansson <oej@edvina.net>
+
+ * CHANGES, res/res_mutestream.c (added): Adding MUTEAUDIO()
+ dialplan function and MuteAudio AMI action (pinepeach) Review:
+ https://reviewboard.asterisk.org/r/345/
+
+2009-09-02 01:16 +0000 [r215338] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * /, apps/app_softhangup.c: Merged revisions 215270 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r215270 | dhubbard | 2009-09-01 18:04:52 -0500 (Tue, 01
+ Sep 2009) | 12 lines Use strrchr() so SoftHangup will correctly
+ truncate multi-hyphen channel names In general channel names are
+ in the form Foo/Bar-Z, but the channel name could have multiple
+ hyphens and look like Foo/B-a-r-Z. Use strrchr to truncate the
+ channel name at the last hyphen. (closes issue #15810) Reported
+ by: dhubbard Patches: dw-softhangup-1.4.patch uploaded by
+ dhubbard (license 733) ........
+
+2009-09-01 23:41 +0000 [r215222-215301] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c, funcs/func_channel.c, CHANGES: Add
+ MASTER_CHANNEL() dialplan function, as well as a useful usage.
+ (closes issue #13140) Reported by: cpina Patches:
+ 20090807__issue13140.diff.txt uploaded by tilghman (license 14)
+ Tested by: lmadsen Change-type: feature
+
+ * channels/chan_sip.c: Fix register such that lines with a
+ transport string, but without an authuser, parse correctly.
+ (AST-228)
+
+2009-09-01 20:44 +0000 [r215212] Russell Bryant <russell@digium.com>
+
+ * addons/format_mp3.c: Fix memory corruption caused by format_mp3.
+ format_mp3 claimed that it provided AST_FRIENDLY_OFFSET in frames
+ returned by read(). However, it lied. This means that other parts
+ of the code that attempted to make use of the offset buffer would
+ end up corrupting the fields in the ast_filestream structure.
+ This resulted in quite a few crashes due to unexpected values for
+ fields in ast_filestream. This patch closes out quite a few bugs.
+ However, some of these bugs have been open for a while and have
+ been an area where more than one bug has been discussed. So with
+ that said, anyone that is following one of the issues closed
+ here, if you still have a problem, please open a new bug report
+ for the specific problem you are still having. If you do, please
+ ensure that the bug report is based on the newest version of
+ Asterisk, and that this patch is applied if format_mp3 is in use.
+ Thanks! (closes issue #15109) Reported by: jvandal Tested by:
+ aragon, russell, zerohalo, marhbere, rgj (closes issue #14958)
+ Reported by: aragon (closes issue #15123) Reported by:
+ axisinternet (closes issue #15041) Reported by: maxnuv (closes
+ issue #15396) Reported by: aragon (closes issue #15195) Reported
+ by: amorsen Tested by: amorsen (closes issue #15781) Reported by:
+ jensvb (closes issue #15735) Reported by: thom4fun (closes issue
+ #15460) Reported by: marhbere
+
+2009-09-01 19:50 +0000 [r215161] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/frame.c: Ensure that frame dumps of
+ AST_CONTROL_T38_PARAMETERS frames are properly decoded.
+
+2009-09-01 14:40 +0000 [r215110] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Removing whitespace that causes red dots in
+ reviewboard
+
+2009-08-31 22:02 +0000 [r215069-215070] Tilghman Lesher <tlesher@digium.com>
+
+ * main/http.c: Fix a trunk compilation warning.
+
+ * main/manager.c: Properly initialize the session to prevent a
+ crash. (closes issue #15774) Reported by: lasko Patches:
+ 20090831__issue15774.diff.txt uploaded by tilghman (license 14)
+ Tested by: lasko
+
+2009-08-31 18:17 +0000 [r215023] Olle Johansson <oej@edvina.net>
+
+ * funcs/func_volume.c: By copying this code I got bad comments in
+ reviewboard... Better fix the original.
+
+2009-08-31 16:18 +0000 [r214819-214945] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 214940 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r214940 | tilghman | 2009-08-31 11:16:52 -0500 (Mon, 31
+ Aug 2009) | 7 lines Also unlock the "other" channel, when
+ returning, due to glare. (closes issue #15787) Reported by:
+ tim_ringenbach Patches: chan_local.diff uploaded by tim
+ ringenbach (license 540) Tested by: tim_ringenbach ........
+
+ * Makefile: Force Darwin on ppc platforms to compile with a target
+ level that supports aliasing.
+
+ * include/asterisk.h, main/poll.c: Various patches, to enable
+ Asterisk to once again compile on Mac OS X. One note on defining
+ _POSIX_C_SOURCE: while this feature test macro works to require
+ certain behaviors on Linux, it works differently on *BSD
+ platforms to REMOVE certain API calls that are not in the POSIX
+ specification, such as vasprintf(3). Thus, defining it while
+ depending upon vasprintf (and other extensions to the POSIX
+ standard) to be defined is a recipe to ensure that Asterisk is
+ only buildable on Linux. Hence, this define which was meant to
+ INCREASE portability, effectively ensures the opposite.
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ pbx/pbx_lua.c: If lua is detected with the lua5.1 prefix (or
+ not), adjust the include path accordingly. Based upon feedback to
+ a release announcement on the -users list. See
+ http://lists.digium.com/pipermail/asterisk-users/2009-August/236954.html
+
+2009-08-28 22:44 +0000 [r214777] Russell Bryant <russell@digium.com>
+
+ * configure: Update configure script so that CONFIG_CFLAGS and
+ CONFIG_LDFLAGS doesn't break the build.
+
+2009-08-28 20:14 +0000 [r214702] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 214701 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214701 | tilghman | 2009-08-28 15:13:32 -0500 (Fri, 28 Aug 2009)
+ | 8 lines Modify comment to be a bit more accurate. We have kept
+ this comment around long enough, that it's pretty clear that
+ we're keeping the code, because changing the code would require a
+ pretty fundamental architectural shift. We've also taken
+ criticism in some quarters, because it was believed that it was
+ referring to the code being nasty. No, the code isn't nasty, just
+ the operation itself is rather odd. Fixed for eternity (probably
+ not). ........
+
+2009-08-28 20:01 +0000 [r214696] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, include/asterisk/autoconfig.h.in, configure.ac,
+ makeopts.in: Ensure that CFLAGS and/or LDFLAGS provided to
+ configure script are preserved. Cross-compilation environments
+ want to provide 'defaults' for compiler and linker options, and
+ frequently do this by specifying CFLAGS and LDFLAGS in the
+ environment or as command-line arguments to the configure script.
+ This patch modifies the configure script and Makefile to preserve
+ these settings and ensure they are used in the build process.
+
+2009-08-28 19:13 +0000 [r214654] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Move discardremoteholdretrieval test so it
+ applies only to the specific notification indicator values.
+
+2009-08-28 18:41 +0000 [r214650] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/sched.h: Fix some incorrect documentation of
+ sched_thread functions.
+
+2009-08-28 16:50 +0000 [r214360-214611] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c: Remove unnecessary define for Solaris
+ (closes issue #15358) Reported by: snuffy Patches:
+ bug_res_moh_remove_unneeded_include.diff uploaded by snuffy
+ (license 35)
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ autoconf/libcurl.m4 (added): Merged revisions 214517 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r214517 | tilghman | 2009-08-27 16:45:34 -0500 (Thu, 27
+ Aug 2009) | 7 lines Use autoconf to detect libcurl, as this
+ enables cross-compilation checks, something we didn't allow
+ before. (closes issue #15714) Reported by: pprindeville Patches:
+ 20090813__issue15714.diff.txt uploaded by tilghman (license 14)
+ Tested by: pprindeville ........
+
+ * main/manager.c: Ensure that we check for the special value
+ CONFIG_STATUS_FILEINVALID. (closes issue #15786) Reported by:
+ a_villacis Patches:
+ asterisk-1.6.2.0-beta4-manager-fix-crash-on-include-nonexistent-file.patch
+ uploaded by a villacis (license 660) (Plus a few of my own, to
+ catch the remaining places within manager.c where it could have
+ been a problem)
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ autoconf/ast_ext_lib.m4: Merged revisions 214436 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r214436 | tilghman | 2009-08-27 11:53:58 -0500 (Thu, 27
+ Aug 2009) | 2 lines One more build system change, to make the
+ descriptions look better, if we have better information. ........
+
+ * /, configure, include/asterisk/autoconfig.h.in,
+ autoconf/ast_ext_lib.m4: Merged revisions 214357 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r214357 | tilghman | 2009-08-27 11:03:50 -0500 (Thu, 27
+ Aug 2009) | 3 lines Make autoheader descriptions render correctly
+ in our autoconfig.h file. (Figured out while working with issue
+ #14906) ........
+
+2009-08-27 15:57 +0000 [r214309-214355] Jeff Peeler <jpeeler@digium.com>
+
+ * doc/tex/channelvariables.tex: Add forgotten documentation for new
+ channel variables added in 214309.
+
+ * main/features.c, CHANGES: Add two new dialplan variables when
+ using features Added DYNAMIC_FEATURENAME which holds the last
+ triggered dynamic feature. Added DYNAMIC_PEERNAME which holds the
+ unique channel name on the other side and is set when a dynamic
+ feature is triggered. (closes issue #14663) Reported by: tamiel
+ Patches: 20090313_features.diff uploaded by tamiel (license 712)
+ Tested by: tamiel
+
+2009-08-26 21:56 +0000 [r214272] Richard Mudgett <rmudgett@digium.com>
+
+ * configs/chan_dahdi.conf.sample: Minor punctuation change.
+
+2009-08-26 16:53 +0000 [r214199] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Typo fix ("SIP/2.0 XXX" is 11 chars, not 10)
+ (closes issue #15362) Reported by: klaus3000 Patches:
+ chan_sip.c_logmessagefix_patch.txt uploaded by klaus3000 (license
+ 65)
+
+2009-08-26 16:38 +0000 [r214195] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /: Merged revisions 214194 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214194 | dvossel | 2009-08-26 11:36:42 -0500 (Wed, 26 Aug 2009)
+ | 19 lines ast_write() ignores ast_audiohook_write() results In
+ ast_write(), if a channel has a list of audiohooks, those lists
+ are written to and the resulting frame is what ast_write() should
+ continue with. The problem was the returned audiohook frame was
+ not being handled at all, and the original frame passed into it
+ did not contain the mixed audio, so essentially audio was being
+ lost. One result of this was chan_spy's whisper mode no longer
+ worked. To complicate the issue, frames passed into ast_write may
+ either be a single frame, or a list of frames. So, as the list of
+ frames is processed in the audiohook_write, the returned frames
+ had to be added to a new list. (closes issue #15660) Reported by:
+ corruptor Tested by: dvossel ........
+
+2009-08-25 22:39 +0000 [r213900-214152] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac: Not
+ all versions of gnu-linux use glibc, which contains iconv. Some
+ (especially embedded systems) don't have iconv at all. (closes
+ issue #15169) Reported by: pprindeville
+
+ * /, main/say.c: Merged revisions 214068-214069 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214068 | tilghman | 2009-08-25 14:26:50 -0500 (Tue, 25 Aug 2009)
+ | 6 lines Fix pronunciation of German dates. (closes issue
+ #15273) Reported by: Benjamin Kluck Patches: say_c.patch uploaded
+ by Benjamin Kluck (license 803) ........ r214069 | tilghman |
+ 2009-08-25 14:28:42 -0500 (Tue, 25 Aug 2009) | 2 lines I should
+ always compile before committing... ........
+
+ * pbx/pbx_dundi.c: DUNDILOOKUP function in 1.6 should use comma
+ delimiters. (closes issue #15322) Reported by: chappell Patches:
+ dundilookup-0015322.patch uploaded by chappell (license 8)
+
+ * main/pbx.c, /: Merged revisions 213970 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r213970 | tilghman | 2009-08-25 01:34:44 -0500 (Tue, 25 Aug 2009)
+ | 7 lines Improve error message by informing user exactly which
+ function is missing a parethesis. (closes issue #15242) Reported
+ by: Nick_Lewis Patches: pbx.c-funcparenthesis.patch2 uploaded by
+ dbrooks (license 790) pbx.c-funcparenthesis-1.4.diff uploaded by
+ loloski (license 68) ........
+
+ * Makefile: The DTD should be installed in the same path as the
+ rest of the XML documentation. (closes issue #15344) Reported by:
+ tzafrir Patches: makefile_appdocs_dtd.diff uploaded by tzafrir
+ (license 46)
+
+ * Makefile, /: Merged revisions 213899 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r213899 | tilghman | 2009-08-24 21:40:22 -0500 (Mon, 24 Aug 2009)
+ | 4 lines Use the default runlevels for Debian derivatives,
+ instead of making up our own. (closes issue #14730) Reported by:
+ pkempgen ........
+
+2009-08-24 16:43 +0000 [r213833] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c: Fix storage of greetings when using
+ IMAP_STORAGE The store macro was not getting called preventing
+ storage of IMAP greetings at all. This has been corrected along
+ with fixing checking if the imapgreetings option is turned on to
+ store the greeting in IMAP. Lastly, the attachment filename was
+ incorrectly using the full path instead of just the basename,
+ which was causing problems with retrieval of the greeting.
+ (closes issue #14950) Reported by: noahisaac (closes issue
+ #15729) Reported by: lmadsen
+
+2009-08-24 04:46 +0000 [r213790] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c: improve handling of
+ openr2_chan_disconnect_call API failure, unlikely, but happened
+ on openr2 library bug
+
+2009-08-21 23:18 +0000 [r213748] Richard Mudgett <rmudgett@digium.com>
+
+ * configure, configure.ac, channels/sig_pri.c: Update configure
+ script for libpri COLP feature dependency requirements.
+
+2009-08-21 22:36 +0000 [r213738] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Clarifying comments in sip_register, and
+ removing a dead section
+
+2009-08-21 22:22 +0000 [r213716] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Register request line contains wrong address
+ when user domain and register host differ (closes issue #15539)
+ Reported by: Nick_Lewis Patches: chan_sip.c-registraraddr.patch
+ uploaded by Nick (license 657) register_domain_fix_1.6.2 uploaded
+ by dvossel (license 671) Tested by: Nick_Lewis, dvossel
+
+2009-08-21 21:39 +0000 [r213697] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_voicemail.c: Ensure that realtime mailboxes properly
+ report status on subscription. This patch modifies
+ app_voicemail's response to mailbox status subscriptions (via the
+ internal event system) to ensure that a subscription triggers an
+ explicit poll of the mailbox, so the subscriber can get an
+ immediate cached event with that status. Previously, the cache
+ was only populated with the status of non-realtime mailboxes.
+ (closes issue #15717) Reported by: natmlt
+
+2009-08-21 21:02 +0000 [r213635] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes sip register parsing when user@domain
+ is used (issue #15008) (issue #15672)
+
+2009-08-21 16:53 +0000 [r213560] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk.h, /: Merged revisions 213559 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r213559 | tilghman | 2009-08-21 11:52:53 -0500 (Fri, 21 Aug 2009)
+ | 7 lines Permit DEBUG_FD_LEAKS to be used with C++ source files.
+ (closes issue #15698) Reported by: slavon Patches:
+ 20090817__issue15698.diff.txt uploaded by tilghman (license 14)
+ Tested by: slavon, tilghman ........
+
+2009-08-21 16:04 +0000 [r213494] Jason Parker <jparker@digium.com>
+
+ * /, configs/queues.conf.sample: Merged revisions 213493 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r213493 | qwell | 2009-08-21 11:03:21 -0500 (Fri, 21 Aug 2009) |
+ 5 lines Clarify queues.conf comments to specify that variables
+ should be set in the dialplan. (closes issue #15755) Reported by:
+ trendboy ........
+
+2009-08-21 04:09 +0000 [r213454] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c: increment the mfcr2 monitor count when
+ clearing the call request
+
+2009-08-21 03:48 +0000 [r213450] Terry Wilson <twilson@digium.com>
+
+ * main/loader.c: Make LOAD_ORDER actually work
+
+2009-08-20 22:13 +0000 [r213414] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c: Add original position, when logging a caller
+ entering a queue. (closes issue #15146) Reported by: arabe
+ Patches: asterisk-trunk.patch uploaded by arabe (license 786)
+
+2009-08-20 21:33 +0000 [r213404] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c: Fix greeting retrieval from IMAP Properly
+ check for the current voicemail state and if it doesn't exist,
+ create it. (closes issue #14597) Reported by: wtca Patches:
+ 14597_v2.patch uploaded by mmichelson (license 60) Tested by:
+ jpeeler
+
+2009-08-20 20:29 +0000 [r213327] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/features.c: Fix a crash by checking the proper pointer for
+ validity before deferencing it. (closes issue #15751) Reported
+ by: atis Patches: ast_bridge_call_peer_cdr.patch uploaded by atis
+ (license 242)
+
+2009-08-20 19:56 +0000 [r213284] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.exports (added), /: Merged revisions 213283
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r213283 | jpeeler | 2009-08-20 14:53:34 -0500 (Thu, 20 Aug 2009)
+ | 2 lines Make all the symbols for the C-client callbacks global
+ ........
+
+2009-08-20 15:29 +0000 [r213248] Tilghman Lesher <tlesher@digium.com>
+
+ * addons/res_config_mysql.c: Select uncommented lines, not
+ commented ones. (closes issue #15746) Reported by: makoto
+
+2009-08-20 03:26 +0000 [r213216] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c: fixed bug caused by calling ast_request
+ without calling ast_call on an R2 channel, ie, CHANISAVAIL
+
+2009-08-19 22:38 +0000 [r213179] Jason Parker <jparker@digium.com>
+
+ * main/ulaw.c, main/alaw.c: Fix compile when certain G711
+ menuselect options are enabled. (closes issue #15697) Reported
+ by: slavon
+
+2009-08-19 21:21 +0000 [r213113] David Vossel <dvossel@digium.com>
+
+ * /, apps/app_mixmonitor.c: Merged revisions 213103 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r213103 | dvossel | 2009-08-19 16:18:37 -0500 (Wed, 19
+ Aug 2009) | 8 lines Fixes memory leak caused by incorrectly
+ freeing mixmonitor (closes issue #15699) Reported by: edantie
+ Patches: mixmonitor.patch uploaded by edantie (license 862)
+ ........
+
+2009-08-19 21:05 +0000 [r213093-213098] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: Better parsing for
+ the "register" line Allows characters that are otherwise used as
+ delimiters to be used within certain fields (like the secret).
+ (closes issue #15008, closes issue #15672) Reported by: tilghman
+ Patches: 20090818__issue15008.diff.txt uploaded by tilghman
+ (license 14) Tested by: lmadsen, tilghman
+
+ * channels/chan_sip.c: If we have realtime caching enabled, 'sip
+ reload' must purge users/peers, even if the config files haven't
+ changed. (closes issue #12869) Reported by: bcnit Patches:
+ 20090819__issue12869__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: lasko
+
+2009-08-19 15:32 +0000 [r213046] Russell Bryant <russell@digium.com>
+
+ * main/features.c: Don't blow up on a NULL cdr. Reported in
+ #asterisk-dev.
+
+2009-08-18 23:53 +0000 [r213007] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, CHANGES, channels/sig_pri.c: Add COLP support
+ to chan_dahdi/sig_pri. Add Connected Line Presentation (COLP)
+ support to chan_dahdi/libpri as an addition to issue 8824. This
+ is the chan_dahdi/sig_pri portion. COLP support is now available
+ for any switch for which libpri supports COLP (currently ETSI
+ PTP, ETSI PTMP, and Q.SIG) with this patch. (closes issue #14068)
+ Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/340/
+
+2009-08-18 20:33 +0000 [r212922-212939] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /: Remove some accidentally-committed properties.
+
+ * CREDITS, /, UPGRADE-1.4.txt, sounds/sounds.xml,
+ build_tools/prep_tarball, sounds/Makefile, doc/tex/asterisk.tex:
+ Convert this branch to Opsound music-on-hold. For more details:
+ http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/
+
+2009-08-18 19:49 +0000 [r212857-212883] Tilghman Lesher <tlesher@digium.com>
+
+ * addons/res_config_mysql.c: Clarify some of the error messages, to
+ help upgraders.
+
+ * configs/extconfig.conf.sample: Make the default extconfig.conf
+ match entries with the sample res_mysql.conf. This eliminates a
+ future source of possible confusion with the configuration of
+ 1.6.1 and higher.
+
+2009-08-18 18:57 +0000 [r212844] Olle Johansson <oej@edvina.net>
+
+ * apps/app_meetme.c: Small doxygen changes
+
+2009-08-18 16:38 +0000 [r212764] Sean Bright <sean@malleable.com>
+
+ * main/manager.c, /: Merged revisions 212763 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r212763 | seanbright | 2009-08-18 12:36:00 -0400 (Tue, 18 Aug
+ 2009) | 11 lines Delay the creation of temporary files until we
+ have a valid manager command to handle. Without this patch,
+ asterisk creates a temporary file before determining if the
+ specified command is valid. If invalid, we weren't properly
+ cleaning up the file. (closes issue #15730) Reported by: zmehmood
+ Patches: M15730.diff uploaded by junky (license 177) Tested by:
+ zmehmood ........
+
+2009-08-18 16:29 +0000 [r212758] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/misdn/isdn_lib.c: Merged revisions 212727 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r212727 | rmudgett | 2009-08-18 11:00:56 -0500 (Tue, 18 Aug 2009)
+ | 1 line Removed some deadwood and added some doxygen comments.
+ ........
+
+2009-08-17 20:40 +0000 [r212672] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk.h: Relax check for XOPEN_VERSION. It's not clear
+ that we actually require XOPEN_VERSION to be 600 or greater at
+ this time, so skip the check for now.
+
+2009-08-17 19:57 +0000 [r212627] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Check the return value of opendir(3), or we
+ may crash. (closes issue #15720) Reported by: tobias_e
+
+2009-08-17 18:50 +0000 [r212574-212581] Sean Bright <sean@malleable.com>
+
+ * channels/chan_agent.c: Correct spelling of AGENTACCEPTDTMF in
+ chan_agent. (closes issue #15668) Reported by: davidw
+
+ * main/logger.c: Correct the return value check for
+ ast_safe_system. The logic here was reversed as ast_safe_system
+ returns -1 on error and not on success. Fix suggested by
+ reporter. (closes issue #15667) Reported by: loic
+
+2009-08-17 16:50 +0000 [r212506] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/misdn_config.c: Merged revisions 212498 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r212498 | jpeeler | 2009-08-17 11:34:56 -0500 (Mon, 17
+ Aug 2009) | 12 lines Fix segfault when reloading chan_misdn. If
+ more ports were specified than configured in misdn.conf a reload
+ would crash asterisk. The problem was the unconfigured port was
+ using data from the previously configured port. When the data for
+ an unconfigured port was freed a crash would result from the
+ double free. (closes issue #12113) Reported by: agupta Patches:
+ bug12113.patch uploaded by jpeeler (license 325) ........
+
+2009-08-17 16:25 +0000 [r212463] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk.h, main/xml.c: Define our desires for POSIX and
+ X/OPEN API features properly. Based on a post on the gcc-help
+ mailing list and some subsequent reading, we can increase our
+ portability to various platforms by directly defining the POSIX
+ and X/OPEN API feature sets we wish to have available. This patch
+ does that, and also includes a double-check to ensure that the
+ system we are compiling on can actually provide the requested
+ feature sets.
+
+2009-08-17 15:42 +0000 [r212431] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+ 212430 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 Fix
+ uninitialized variable causing random MWI indications. (closes
+ issue #15727) Reported by: doda Patches: dahdi_changes.patch
+ uploaded by doda (license 853) ........ r212430 | rmudgett |
+ 2009-08-17 10:36:28 -0500 (Mon, 17 Aug 2009) | 1 line Fix
+ uninitialized variable. ........
+
+2009-08-16 19:27 +0000 [r212390] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp_engine.c, include/asterisk/rtp_engine.h: Add two more
+ API calls for getting the current glue and channel in bridging
+ code.
+
+2009-08-15 11:36 +0000 [r212339-212343] Michiel van Baak <michiel@vanbaak.info>
+
+ * res/res_calendar.c: cast time_t type variables to long where
+ needed. This makes res_calendar.c compile on OpenBSD and the same
+ cast is used in a lot of other places where time_t type vars are
+ used. (closes issue #15656) Reported by: mvanbaak Patches:
+ 2009081100-rescalendarcompilefix.diff.txt uploaded by mvanbaak
+ (license 7)
+
+ * main/xmldoc.c: Add an empty line after each option when printing
+ the documentation of a function/application. This will make
+ reading the docs on the CLI way more easy. (closes issue #15694)
+ Reported by: mvanbaak Patches:
+ 2009081100-extralinesoptionlist.diff.txt uploaded by mvanbaak
+ (license 7)
+
+2009-08-14 23:07 +0000 [r212287-212291] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/sig_analog.c: Add braces where missing and a few
+ whitespace fixes in sig_analog (closes issue #15678) Reported by:
+ alecdavis Patches: sig_analog_mainly_braces.diff.txt uploaded by
+ alecdavis (license 585) Tested by: alecdavis
+
+ * channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h: More code that somehow got left out of
+ sig_analog * confirmanswer option now respected * check and set
+ waiting for dialtone timer * unneeded needcallerid flag removed
+ from analog_subchannel * ss_astchan does not need to be a void
+ pointer * swap_channels callback updated to trunk * analog_hangup
+ now resets channel to default law
+
+2009-08-14 17:36 +0000 [r212249] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_curl.c: Add SSL_VERIFYPEER, as requested on the -users
+ list
+
+2009-08-13 17:33 +0000 [r212199] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: Send a generic return result when we
+ receive a CallDeflection facility message in chan_misdn. ETSI
+ 300-196 implies that a facility return result without arguments
+ does not have the operation-value. This fact implies for ETSI
+ that you can only use the invoke-id to match requests with
+ responses.
+
+2009-08-13 16:44 +0000 [r212161] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp_engine.c, include/asterisk/rtp_engine.h: Add an API call
+ for retrieving the engine in use by an RTP instance.
+
+2009-08-13 15:46 +0000 [r212113] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: Ensure that T38FaxVersion is put into
+ outgoing SDP in the proper case.
+
+2009-08-13 13:51 +0000 [r212067] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Check an actual populated variable when
+ seeing if we need to do video or not.
+
+2009-08-13 11:37 +0000 [r212027] Gavin Henry <ghenry@suretecsystems.com>
+
+ * contrib/scripts/asterisk.ldap-schema,
+ contrib/scripts/asterisk.ldif: Fixed typo (closes issue #15710)
+ Reported by: suretec
+
+2009-08-12 23:14 +0000 [r211947-211957] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 211953 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211953 | mnicholson | 2009-08-12 18:04:02 -0500 (Wed, 12 Aug
+ 2009) | 10 lines This patch adds additional checking when
+ generating queue log TRANSFER events. The additional checks
+ prevent generation of false TRANSFER events in certain
+ situations. (closes issue #14536) Reported by: aragon Patches:
+ queue-log-xfer-fix1.diff uploaded by mnicholson (license 96)
+ Tested by: aragon, mnicholson ........
+
+ * channels/chan_sip.c, configs/sip.conf.sample: This patch adds
+ support for choosing a realm based on the domain in the From or
+ To header in the incoming request. Eligible domains are taken
+ from the domains list in the config file. This functionality is
+ enabled when domainsasrealm is enabled in the config file.
+ (closes issue #11361) Reported by: arkadia Patches:
+ sip_realm_mnich_to_added_2.patch uploaded by arkadia (license
+ 233) Tested by: arkadia
+
+2009-08-12 20:47 +0000 [r211908] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h: Fix chan_dahdi option ringtimeout
+ dahdi_read relies on the dahdi_pvt copy of ringt which was not
+ getting set in sig_analog. This patch adds a callback to do so.
+ (closes issue #15288) Reported by: alecdavis Patches:
+ chan_dahdi.ringtimeout.diff.txt uploaded by alecdavis (license
+ 585) Tested by: alecdavis
+
+2009-08-12 19:53 +0000 [r211876] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Make asterisk handle 423 Interval Too Short
+ messages better. This change uses separate values for the
+ acceptable minimum expiry provided by the 423 error and the
+ expiry value stored in the configuration file. Previously, the
+ value pulled from the configuration file would be overwritten.
+ (closes issue #14366) Reported by: Nick_Lewis Patches:
+ sip-expiry-fix1.diff uploaded by mnicholson (license 96)
+ chan_sip.c-reqexpiry.patch uploaded by Nick (license 657) Tested
+ by: mnicholson
+
+2009-08-12 16:00 +0000 [r211767] Gavin Henry <ghenry@suretecsystems.com>
+
+ * res/res_config_ldap.c, contrib/scripts/asterisk.ldap-schema,
+ contrib/scripts/asterisk.ldif: Added three new attributes and
+ applied a patch to res_config_ldap.c attributetype (
+ AstAccountSubscribeContext NAME 'AstAccountSubscribeContext' DESC
+ 'Asterisk subscribe context' EQUALITY caseIgnoreMatch SUBSTR
+ caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15)
+ attributetype ( AstAccountIpAddr NAME 'AstAccountIpAddr' DESC
+ 'Asterisk aaccount IP address' EQUALITY caseIgnoreMatch SUBSTR
+ caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15)
+ attributetype ( AstAccountUserAgent NAME 'AstAccountUserAgent'
+ DESC 'Asterisk account user context' EQUALITY caseIgnoreMatch
+ SUBSTR caseIgnoreSubstringsMatch SYNTAX
+ 1.3.6.1.4.1.1466.115.121.1.15) and patch
+ fix_empty_attributes_1.6.1.4_v2.patch (closes issue #13725)
+ Reported by: macogeek Patches:
+ fix_empty_attributes_1.6.1.4_v2.patch uploaded by xvisor (license
+ 863) Tested by: suretec
+
+2009-08-12 10:11 +0000 [r211732] Russell Bryant <russell@digium.com>
+
+ * channels/chan_jingle.c, channels/chan_unistim.c,
+ channels/chan_skinny.c, channels/chan_h323.c,
+ channels/chan_gtalk.c, channels/chan_mgcp.c: Always specify which
+ RTP engine is desired for a new RTP instance. This fixes a crash
+ reported in #asterisk-dev where chan_mgcp unexpectedly allocated
+ an RTP instance from res_rtp_multicast, since by not specifying
+ an engine, you get the first one in the list of engines.
+
+2009-08-10 23:21 +0000 [r211675] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Encapsulate testing for which signaling
+ styles are used by sig_pri. Created the
+ dahdi_sig_pri_lib_handles() function and SIG_PRI_LIB_HANDLE_CASES
+ macro to simplify testing for which signaling styles are handled
+ by sig_pri.
+
+2009-08-10 19:49 +0000 [r211539-211584] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/CODING-GUIDELINES, /: Merged revisions 211583 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r211583 | tilghman | 2009-08-10 14:48:48 -0500 (Mon, 10
+ Aug 2009) | 1 line Conversion specifiers, not format specifiers
+ ........
+
+ * cel/cel_pgsql.c, funcs/func_speex.c, funcs/func_rand.c,
+ apps/app_dahdibarge.c, main/frame.c, addons/chan_ooh323.c,
+ apps/app_readfile.c, /, apps/app_record.c,
+ apps/app_alarmreceiver.c, cdr/cdr_adaptive_odbc.c,
+ res/res_http_post.c, channels/chan_iax2.c, main/indications.c,
+ main/config.c, main/cli.c, pbx/pbx_loopback.c,
+ channels/chan_dahdi.c, pbx/pbx_spool.c, res/res_smdi.c,
+ channels/chan_skinny.c, main/features.c, main/http.c, main/pbx.c,
+ funcs/func_sprintf.c, funcs/func_timeout.c, apps/app_privacy.c,
+ codecs/codec_speex.c, channels/chan_agent.c, funcs/func_math.c,
+ apps/app_disa.c, apps/app_morsecode.c, channels/iax2-provision.c,
+ funcs/func_cut.c, apps/app_talkdetect.c, main/netsock.c,
+ res/res_config_curl.c, channels/chan_misdn.c,
+ apps/app_waitforring.c, funcs/func_channel.c, apps/app_macro.c,
+ addons/cdr_mysql.c, pbx/pbx_config.c, apps/app_mixmonitor.c,
+ apps/app_chanspy.c, main/asterisk.c, res/res_odbc.c,
+ cel/cel_adaptive_odbc.c, main/timing.c, apps/app_voicemail.c,
+ doc/CODING-GUIDELINES, addons/app_mysql.c, utils/muted.c,
+ apps/app_meetme.c, main/utils.c, res/res_musiconhold.c,
+ cdr/cdr_pgsql.c, apps/app_followme.c, res/res_config_sqlite.c,
+ main/enum.c, utils/frame.c, channels/misdn_config.c,
+ main/channel.c, res/ael/pval.c, main/cdr.c, funcs/func_enum.c,
+ channels/chan_phone.c, main/manager.c, apps/app_setcallerid.c,
+ apps/app_osplookup.c, funcs/func_odbc.c, res/res_agi.c,
+ apps/app_minivm.c, channels/xpmr/xpmr.c, res/res_config_ldap.c,
+ apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c,
+ res/res_config_pgsql.c, funcs/func_dialplan.c, main/dnsmgr.c,
+ channels/chan_sip.c, res/res_limit.c, apps/app_waitforsilence.c,
+ agi/eagi-test.c, main/acl.c, apps/app_waituntil.c,
+ apps/app_originate.c, channels/sig_pri.c, apps/app_queue.c,
+ channels/chan_oss.c, agi/eagi-sphinx-test.c,
+ channels/chan_usbradio.c, res/snmp/agent.c, pbx/pbx_dundi.c,
+ apps/app_sms.c, utils/extconf.c, apps/app_stack.c,
+ apps/app_verbose.c, addons/app_saycountpl.c, main/dsp.c,
+ addons/res_config_mysql.c: AST-2009-005
+
+2009-08-10 18:01 +0000 [r211475] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: add manager events when a skinny device
+ registers/unregisters like we have in chan_sip (closes issue
+ #15499) Reported by: arifzaman Patches:
+ 2009072600-skinnymanagerevents.diff.txt uploaded by mvanbaak
+ (license 7)
+
+2009-08-10 17:17 +0000 [r211435] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_pri.c: Fix PRI/BRI channels
+ when in alarm condition to only be marked for hangup if T309 is
+ not enabled.
+
+2009-08-10 15:53 +0000 [r211392] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
+ Restoring some code to sig_pri. Not sure if it is really needed.
+ Putting some DSP code back into sig_pri that was removed by the
+ chan_dahdi/sig_pri reorganization.
+
+2009-08-10 15:46 +0000 [r211390] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Fix up some issues with getting a channel by
+ "name". Even though the get_channel_by_name() API advertised that
+ you could search by name or uniqueid (just as the old API did),
+ searching by uniqueid was not actually implemented. This patch
+ fixes that problem. The ast_channel_get_full() function now makes
+ a second search attempt by uniqueid if the parameter was a name.
+ The channel comparison function also now knows how to compare by
+ unqieueid. Finally, a bug was fixed in passing where OBJ_POINTER
+ was being passed in some scenarios where it should not have been.
+
+2009-08-10 14:07 +0000 [r211347] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix retrieval of the port used for the video
+ stream when adding SDP to a SIP message. (closes issue #15121)
+ Reported by: jsmith
+
+2009-08-09 15:42 +0000 [r211232-211275] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/astfd.c: Merged revisions 211274 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211274 | tilghman | 2009-08-09 10:41:01 -0500 (Sun, 09 Aug 2009)
+ | 2 lines Small oops. Clear the flags which have been checked.
+ ........
+
+ * apps/app_stack.c: Check for NULL frame, before dereferencing
+ pointer. (closes issue #15617) Reported by: rain
+
+2009-08-07 23:30 +0000 [r211191-211197] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Fixed some unsafe down cast pointer
+ operations for sig_pri. You cannot cast the struct
+ dahdi_pvt.sig_pvt pointer to a specific signaling private pointer
+ without first checking that it is in fact pointing to the correct
+ signaling private structure.
+
+ * channels/sig_pri.c: Fix static on line when PRI does overlap
+ dialing. The wrong encoding law was used because = was used when
+ it should have been ==.
+
+2009-08-07 20:12 +0000 [r211113] Russell Bryant <russell@digium.com>
+
+ * /: Recorded merge of revisions 211112 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211112 | russell | 2009-08-07 15:11:31 -0500 (Fri, 07 Aug 2009)
+ | 4 lines Resolve a deadlock involving app_chanspy and
+ masquerades. (ABE-1936) ........
+
+2009-08-07 18:17 +0000 [r211040] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 211038 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009)
+ | 14 lines QUEUE_MEMBER_LIST _really_ wants the interface name,
+ not the membername. This is a partial revert of revision 82590,
+ which was an attempted cleanup, but in reality, it broke
+ QUEUE_MEMBER_LIST, which has always been intended as a method by
+ which component interfaces could be queried from the queue.
+ Membername isn't useful here, because that field cannot be used
+ to obtain further information about the member. See the
+ documentation on QUEUE_MEMBER_LIST, RemoveQueueMember,
+ QUEUE_MEMBER_PENALTY, and the various AMI commands which take a
+ member argument for further justification. (closes issue #15664)
+ Reported by: rain Patches: app_queue-queue_member_list.diff
+ uploaded by rain (license 327) ........
+
+2009-08-07 13:08 +0000 [r210992] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c: Workaround broken T.38 endpoints that offer tiny
+ MaxDatagram sizes. Some T.38 endpoints treat T38FaxMaxDatagram as
+ the maximum IFP size that should be sent to them, rather than the
+ maximum packet payload size. If such an endpoint also requests
+ UDPRedundancy as the error correction mode, we'll end up
+ calculating a tiny maximum IFP size, so small as to be unusable.
+ This patch sets a lower bound on what we'll consider the remote's
+ maximum IFP size to be, assuming that endpoints that do this
+ really can accept larger packets than they've offered to accept.
+ (closes issue #15649) Reported by: dazza76
+
+2009-08-06 21:46 +0000 [r210908-210914] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 210913 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210913 | tilghman | 2009-08-06 16:45:01 -0500 (Thu, 06 Aug 2009)
+ | 7 lines Because channel information can be accessed outside of
+ the channel thread, we must lock the channel prior to modifying
+ it. (closes issue #15397) Reported by: caspy Patches:
+ 20090714__issue15397.diff.txt uploaded by tilghman (license 14)
+ Tested by: caspy ........
+
+ * include/asterisk/app.h, main/app.c, apps/app_stack.c: Allow Gosub
+ to recognize quote delimiters without consuming them. (closes
+ issue #15557) Reported by: rain Patches:
+ 20090723__issue15557.diff.txt uploaded by tilghman (license 14)
+ Tested by: rain Review: https://reviewboard.asterisk.org/r/316/
+
+2009-08-06 20:15 +0000 [r210866-210869] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_analog.c: Miscellaneous minor fixes to sig_analog. *
+ Sanity adjustments to __analog_ss_thread for sig_analog
+ environment. * Deleted some duplicated code. * Fixed
+ analog_ss_thread_start passing the wrong pointer.
+
+ * channels/sig_pri.c: Sanity adjustments to pri_ss_thread for
+ sig_pri environment.
+
+2009-08-06 17:47 +0000 [r210817] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Accept additional T.38 reinvites after an
+ initial one has been handled. Discussion of this subject has
+ yielded that it is not actually acceptable to change T.38
+ parameters after the initial reinvite but declining is harsh and
+ can cause the fax to fail when it may be possible to allow it to
+ continue. This patch changes things so that additional T.38
+ reinvites are accepted but parameter changes ignored. This gives
+ the fax a fighting chance. (closes issue #15610) Reported by:
+ huangtx2009
+
+2009-08-06 16:07 +0000 [r210777] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, apps/app_fax.c,
+ configure.ac: Minor improvements to app_fax. This patch makes
+ some small changes to handle watchdog timeouts in a better way,
+ and also uses a 'cleaner' method of including the spandsp header
+ files. (closes issue #14769) Reported by: andrew Patches:
+ app_fax-20090406.diff uploaded by andrew (license 240)
+ v1-14769.patch uploaded by dimas (license 88) Tested by: freh,
+ deti, caspy, dimas, sgimeno, Dovid
+
+2009-08-05 23:44 +0000 [r210640-210732] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.c: Fix potential deadlock issue with
+ USERUSERINFO channel variable.
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
+ More changes from chan_dahdi that did not make it into sig_pri. *
+ Q.SIG channel mapping option. * discardremoteholdretrieval
+ option. * libPRI debug defines. * pri_set_overlapdial() now set
+ correctly. * pthread creation of pri_ss_thread now matches.
+
+ * /, channels/sig_pri.c: Merged revisions 210575 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009)
+ | 14 lines Dialplan starts execution before the channel setup is
+ complete. * Issue 15655: For the case where dialing is complete
+ for an incoming call, dahdi_new() was asked to start the PBX and
+ then the code set more channel variables. If the dialplan hungup
+ before these channel variables got set, asterisk would likely
+ crash. * Fixed potential for overlap incoming call to erroneously
+ set channel variables as global dialplan variables if the
+ ast_channel structure failed to get allocated. * Added missing
+ set of CALLINGSUBADDR in the dialing is complete case. (closes
+ issue #15655) Reported by: alecdavis ........
+
+2009-08-05 18:49 +0000 [r210564] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/imapstorage.tex, /: Merged revisions 210563 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05
+ Aug 2009) | 11 lines Update imapstorage.txt documentation.
+ Updated the imapstorage.txt documentation to reflect that issues
+ with c-client versions older than 2007 seem to cause crashing
+ issues that are not seen with more recent versions. Documentation
+ has been updated to reflect this. (closes issue #14496) Reported
+ by: vbcrlfuser Patches: __20090727-imap-documentation-patch.txt
+ uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson,
+ dbrooks ........
+
+2009-08-05 14:09 +0000 [r210522] Russell Bryant <russell@digium.com>
+
+ * main/file.c: Revert some silly code that snuck into trunk from my
+ working copy. Sorry!
+
+2009-08-05 08:03 +0000 [r210478] Michiel van Baak <michiel@vanbaak.info>
+
+ * addons/mp3: ignore the .i files when compiling in 'DONT_OPTIMIZE'
+ in the addons/mp3 directory
+
+2009-08-04 17:46 +0000 [r210353-210387] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
+ Fix CALLERID() values for sig_pri on incoming calls.
+
+ * main/channel.c, include/asterisk/channel.h: Initial minimum
+ ast_party_caller support.
+
+ * channels/chan_dahdi.c: Removed some dead code.
+
+2009-08-04 15:35 +0000 [r210302] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c: Fix broken call pickup The find_channel_by_group
+ callback was only looking at the channel that was attempting to
+ make the pickup instead of the other channels in the container.
+
+2009-08-04 14:53 +0000 [r210190-210238] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, /: Merged revisions 210237 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210237 | kpfleming | 2009-08-04 09:51:39 -0500 (Tue, 04 Aug
+ 2009) | 10 lines Eliminate spurious compiler warnings from system
+ headers on *BSD platforms. Ensure that system headers located in
+ /usr/local/include are actually treated as system headers by the
+ compiler, and not as local headers which are subject to warnings
+ from the -Wundef compiler option and others. (closes issue
+ #15606) Reported by: mvanbaak ........
+
+ * contrib/scripts/realtime_pgsql.sql, channels/chan_sip.c,
+ channels/chan_skinny.c, configs/mgcp.conf.sample,
+ doc/res_config_sqlite.txt, doc/tex/phoneprov.tex, UPGRADE.txt,
+ configs/res_ldap.conf.sample, configs/sip.conf.sample,
+ configs/skinny.conf.sample, channels/chan_mgcp.c,
+ doc/chan_sip-perf-testing.txt: Rename 'canreinvite' option to
+ 'directmedia', with backwards compatibility. It is clear from
+ multiple mailing list, forum, wiki and other sorts of posts that
+ users don't really understand the effects that the 'canreinvite'
+ config option actually has, and that in some cases they think
+ that setting it to 'no' will actually cause various other
+ features (T.38, MOH, etc.) to not work properly, when in fact
+ this is not the case. This patch changes the proper name of the
+ option to what it should have been from the beginning
+ ('directmedia'), but preserves backwards compatibility for
+ existing configurations.
+
+2009-08-03 18:05 +0000 [r210094-210154] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_pri.c: Changes from
+ chan_dahdi that did not make it into sig_pri. * Moved
+ SUPPORT_USERUSER to sig_pri.c * Fix PRI_DEADLOCK_AVOIDANCE
+ parameter. * Whitespace changes. * Added missing unlock in
+ pri_dchannel():PRI_EVENT_RING case. * Balanced curly braces. *
+ ast_debug/ast_log changes from chan_dahdi. * sig_pri_indicate()
+ should default to return -1 if the indication is not handled.
+
+ * channels/sig_pri.h, channels/sig_analog.c, channels/sig_pri.c:
+ Trim trailing whitespace.
+
+2009-08-03 14:29 +0000 [r210027] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c: Fix order and redundancy of channel rename
+ manager events in ast_do_masquerade. Patch contributed by Mark
+ Spencer.
+
+2009-08-03 14:01 +0000 [r209993] Matthew Nicholson <mnicholson@digium.com>
+
+ * addons/chan_mobile.c, configs/chan_mobile.conf.sample: Add an
+ 'sms' option to mobile.conf to manually enable or disable SMS
+ support. (closes issue #15071) Reported by: ughnz Patches:
+ optional-sms1.diff uploaded by mnicholson (license 96) Tested by:
+ ughnz, mnicholson
+
+2009-08-01 23:33 +0000 [r209958-209959] Bradley Latus <brad.latus@gmail.com>
+
+ * doc/tex/realtime.tex: Update documentation in relation to
+ UnixODBC (closes issue #15516) Reported by: snuffy Patches:
+ bug_odbc_tex_update_v2.diff uploaded by snuffy (license 35)
+
+ * doc/CODING-GUIDELINES: (closes issue #15515)
+
+2009-08-01 11:29 +0000 [r209835-209887] Russell Bryant <russell@digium.com>
+
+ * /, main/db1-ast/mpool/mpool.c: Merged revisions 209879 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209879 | russell | 2009-08-01 06:27:25 -0500 (Sat, 01 Aug 2009)
+ | 5 lines Resolve a valgrind warning about a read from
+ uninitialized memory. (issue #15396) Reported by: aragon ........
+
+ * /, apps/app_milliwatt.c: Merged revisions 209838 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01
+ Aug 2009) | 13 lines Modify how Playtones() is used in
+ Milliwatt() to resolve gain issue. When Milliwatt() was changed
+ internally to use Playtones() so that the proper tone was used,
+ it introduced a drop in gain in the output signal. So, use the
+ playtones API directly and specify a volume argument such that
+ the output matches the gain of the original Milliwatt() code.
+ (closes issue #15386) Reported by: rue_mohr Patches:
+ issue_15386.rev2.diff uploaded by russell (license 2) Tested by:
+ rue_mohr ........
+
+ * main/event.c: Fix ast_event_queue_and_cache() to actually do the
+ cache() part. (closes issue #15624) Reported by: ffossard Tested
+ by: russell
+
+2009-08-01 01:04 +0000 [r209760-209761] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile: Revert accidental Makefile change.
+
+ * Makefile, channels/chan_dahdi.c, channels/chan_misdn.c, /,
+ main/Makefile, channels/misdn/ie.c, pbx/pbx_config.c,
+ utils/frame.c: Merged revisions 209759 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul
+ 2009) | 7 lines Minor changes inspired by testing with latest
+ GCC. The latest GCC (what will become 4.5.x) has a few new
+ warnings, that in these cases found some either downright buggy
+ code, or at least seriously poorly designed code that could be
+ improved. ........
+
+2009-07-31 21:53 +0000 [r209711] Russell Bryant <russell@digium.com>
+
+ * main/event.c: Fix some places where ast_event_type was used
+ instead of ast_event_ie_type.
+
+2009-07-31 17:57 +0000 [r209673-209674] Mark Michelson <mmichelson@digium.com>
+
+ * configs/sip.conf.sample: Add configuration sample code for
+ previous commit.
+
+ * channels/chan_sip.c: Improve chan_sip's ability to determine what
+ methods should and should not be used in a dialog. The previous
+ effort here was to store what a peer is capable of receiving by
+ parsing REGISTER requests from the peer and keeping that
+ information for as long as the registration was active. The
+ problem with this is that there are a great number of SIP devices
+ which give no indication of the methods allowed in their REGISTER
+ requests, and it is unreasonable to try to guess what the device
+ may or may not support. In addition, some SIP devices have been
+ found to claim support for a specific method, but their handling
+ the method is less than ideal, or they are actually lying. With
+ this patch, we now determine what methods a device supports by
+ parsing the Allow header we receive from them, and we do this
+ with each new dialog. In addition, a configuration option has
+ been added so that an administrator can essentially blacklist
+ certain methods from being used with certain peers if the admin
+ knows that support for a specific method is dodgy or nonexistent.
+ ABE-1822
+
+2009-07-30 23:37 +0000 [r209623] Sean Bright <sean@malleable.com>
+
+ * configure, configure.ac, makeopts.in: Allow passing 'noisy' to
+ configure's --enable-dev-mode argument to turn on verbose builds.
+ (closes issue #15607) Reported by: mvanbaak Patches:
+ 20090730_issue15607.patch uploaded by seanbright (license 71)
+ Tested by: seanbright
+
+2009-07-30 23:31 +0000 [r209619] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/sig_pri.h, channels/sig_pri.c: Add missing ifdef-s for
+ service maintenance message functionality (closes issue #15614)
+ Reported by: fabled
+
+2009-07-30 16:07 +0000 [r209554] David Brooks <dbrooks@digium.com>
+
+ * channels/sig_pri.h, apps/app_forkcdr.c, channels/chan_dahdi.c,
+ contrib/init.d/rc.debian.asterisk, addons/chan_ooh323.c,
+ addons/ooh323c/src/ooGkClient.h, funcs/func_math.c,
+ apps/app_sms.c, codecs/lpc10/pitsyn.c, channels/chan_console.c,
+ include/asterisk/abstract_jb.h: Fixes numerous spelling errors.
+ Patch submitted by alecdavis. (closes issue #15595) Reported by:
+ alecdavis
+
+2009-07-30 14:38 +0000 [r209516] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix a crash that can result if text codecs
+ are allowed but textsupport is disabled. (closes issue #15596)
+ Reported by: fabled Patches: sip-red.patch uploaded by fabled
+ (license 448)
+
+2009-07-29 21:46 +0000 [r209453-209484] Matthew Nicholson <mnicholson@digium.com>
+
+ * addons/chan_mobile.c: This patch adds the ability to send a CUSD
+ command to a bluetooth device. (closes issue #15278) Reported by:
+ Artem Patches: cusd5.patch uploaded by Artem (license 800) Tested
+ by: mnicholson, Artem Review:
+ https://reviewboard.asterisk.org/r/274/
+
+ * addons/chan_mobile.c: Fixed a comment for hfp_parse_clip
+
+2009-07-28 13:49 +0000 [r209400] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_usbradio.c, include/asterisk/utils.h,
+ channels/chan_sip.c, channels/chan_alsa.c,
+ channels/chan_console.c, channels/chan_oss.c, main/poll.c: Define
+ side-effect-safe MIN and MAX macros and remove duplicate
+ definitions from various files.
+
+2009-07-28 00:20 +0000 [r209317-209331] Tilghman Lesher <tlesher@digium.com>
+
+ * sounds/sounds.xml: Regex FTL
+
+ * /, sounds/sounds.xml: Merged revisions 209315 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009)
+ | 2 lines Publish French extra sounds ........
+
+2009-07-27 21:43 +0000 [r209256-209279] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_fax.c: Cleanup T.38 negotiation changes. Convert
+ LOG_NOTICE messages about T.38 negotiation in debug level 1
+ messages, clean up some looping logic, and correct an improper
+ use of ast_free() for freeing an ast_frame.
+
+ * apps/app_fax.c: Make T.38 switchover in ReceiveFAX synchronous.
+ In receive mode, if the channel that ReceiveFAX is running on
+ supports T.38, we should *always* attempt to switch T.38, rather
+ than listening for an incoming CNG tone and only triggering on
+ that. The channel may be using a low-bitrate codec that distorts
+ the CNG tone, the sending FAX endpoint may not send CNG at all,
+ or there could be a variety of other reasons that we don't detect
+ it, but in all those cases if T.38 is available we certainly want
+ to use it.
+
+2009-07-27 20:54 +0000 [r209132-209235] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_rtp_asterisk.c: Gracefully handle malformed RTP text
+ packets. AST-2009-004
+
+ * res/res_musiconhold.c: Honor channel's music class when using
+ realtime music on hold. (closes issue #15051) Reported by: alexh
+ Patches: 15051.patch uploaded by mmichelson (license 60) Tested
+ by: alexh
+
+ * main/udptl.c, /, configs/udptl.conf.sample: Merged revisions
+ 209131 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul
+ 2009) | 18 lines Allow for UDPTL to use only even-numbered ports
+ if desired. There are some VoIP providers out there that will not
+ accept SDP offers with odd numbered UDPTL ports. While it is my
+ personal opinion that these VoIP providers are misinterpreting
+ RFC 2327, it really is not a big deal to play along with their
+ silly little games. Of course, since restricting UDPTL ports to
+ only even numbers reduces the range of available ports by half,
+ so the option to use only even port numbers is off by default. A
+ user can enable the behavior by setting use_even_ports=yes in
+ udptl.conf. (closes issue #15182) Reported by: CGMChris Patches:
+ 15182.patch uploaded by mmichelson (license 60) Tested by:
+ CGMChris ........
+
+2009-07-27 16:33 +0000 [r209098] David Brooks <dbrooks@digium.com>
+
+ * channels/chan_dahdi.c, channels/chan_vpb.cc, res/res_smdi.c,
+ include/asterisk/module.h, main/features.c, pbx/pbx_dundi.c,
+ res/res_jabber.c, addons/chan_mobile.c, apps/app_rpt.c,
+ main/loader.c: Fixing typos. Replaces "recieved" with "received"
+ and "initilize" with "initialize" (closes issue #15571) Reported
+ by: alecdavis
+
+2009-07-27 15:38 +0000 [r209056] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile: Restore explicit export of ASTCFLAGS/ASTLDFLAGS and
+ underscore-variants to sub-makes. During the recent Makefile
+ improvements I made, it seemed the 'make' was automatically
+ carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes, so
+ I removed the explict export of them. However, there are some
+ circumstances where make does this, and some where it does not,
+ so I've brought them back to ensure they are always exported. I
+ also removed an extraneous double setting of _ASTLDFLAGS on *BSD
+ platforms.
+
+2009-07-27 01:20 +0000 [r208924] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/translate.c, channels/chan_iax2.c: Merged revisions
+ 208923 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009)
+ | 2 lines Fix logic errors from 208746 ........
+
+2009-07-26 14:00 +0000 [r208886] Michiel van Baak <michiel@vanbaak.info>
+
+ * contrib/scripts/install_prereq: add OpenBSD to the install_prereq
+ script
+
+2009-07-25 12:28 +0000 [r208813-208848] Michiel van Baak <michiel@vanbaak.info>
+
+ * contrib/scripts/install_prereq: libxml2-dev is needed as well by
+ default.
+
+ * configs/cli_aliases.conf.sample, main/cli.c: add default alias
+ reload to run module reload. Requiring 'module reload' to reload
+ everything, including core etc makes russell very unhappy. The
+ default configuration already loads the 'friendly' aliases
+ template. Added 'reload=module reload' to that template. Also
+ removed the comment in main/cli.c that reload should come back.
+
+2009-07-25 06:23 +0000 [r208749] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_skinny.c, main/translate.c,
+ channels/chan_iax2.c: Merged revisions 208746 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009)
+ | 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly
+ trivial changes, but I did not know of any other way to fix the
+ "dereferencing type-punned pointer will break strict-aliasing
+ rules" error without creating a tmp variable in chan_skinny.
+ ........
+
+2009-07-24 21:12 +0000 [r208593-208709] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_dundi.c: Remove trailing whitespace.
+
+ * main/cli.c: Note that "reload" needs to be added back. I keep
+ getting annoyed at having to type "module reload" to reload
+ everything, so I'm adding a note that we need to add "reload"
+ back. "module reload" doesn't really make sense as the command to
+ reload everything, including the core.
+
+ * main/cli.c: Don't log a warning for something that does not
+ affect operation.
+
+ * apps/app_dial.c, /: Merged revisions 208592 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009)
+ | 7 lines Do not log an ERROR if autoservice_stop() returns -1.
+ This does not indicate an error. A return of -1 just means that
+ the channel has been hung up. (reported in #asterisk-dev)
+ ........
+
+2009-07-24 18:31 +0000 [r208588] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 208587 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul
+ 2009) | 10 lines Only send a BYE when hanging up a channel that
+ is up. For cases where Asterisk sends an INVITE and receives a
+ non 2XX final response, Asterisk would follow the INVITE
+ transaction by immediately sending a BYE, which was unnecessary.
+ (closes issue #14575) Reported by: chris-mac ........
+
+2009-07-24 15:02 +0000 [r208548] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, channels/chan_sip.c, include/asterisk/udptl.h:
+ Resolve a T.38 negotiation issue left over from the udptl-updates
+ merge. The udptl-updates branch that was merged yesterday failed
+ to properly send back T.38 SDP responses with the correct error
+ correction mode, if the incoming SDP from the other end caused us
+ to change error correction modes. This patch corrects that
+ situation.
+
+2009-07-24 14:35 +0000 [r208542] Michiel van Baak <michiel@vanbaak.info>
+
+ * contrib/scripts/install_prereq: use aptitude for debian based
+ systems The function to check wether we need to install packages
+ was using dpkg-query which was gives wrong output on Debian 5
+ Also, the apt-get has been replaced with aptitude because
+ aptitude is now the preferred way to handle packages on Debian
+ (closes issue #15570) Reported by: mvanbaak Patches:
+ 2009072400_installprereq-aptitude.diff uploaded by mvanbaak
+ (license 7)
+
+2009-07-23 22:32 +0000 [r208464-208504] Kevin P. Fleming <kpfleming@digium.com>
+
+ * UPGRADE.txt: T.38 change note is not necessary in this branch
+
+ * main/channel.c, main/udptl.c, main/frame.c, main/rtp_engine.c,
+ channels/chan_sip.c, apps/app_fax.c, UPGRADE.txt,
+ include/asterisk/udptl.h, include/asterisk/frame.h: Rework of
+ T.38 negotiation and UDPTL API to address interoperability
+ problems Over the past couple of months, a number of issues with
+ Asterisk negotiating (and successfully completing) T.38 sessions
+ with various endpoints have been found. This patch attempts to
+ address many of them, primarily focused around ensuring that the
+ endpoints' MaxDatagram size is honored, and in addition by
+ ensuring that T.38 session parameter negotiation is performed
+ correctly according to the ITU T.38 Recommendation. The major
+ changes here are: 1) T.38 applications in Asterisk (app_fax) only
+ generate/receive IFP packets, they do not ever work with UDPTL
+ packets. As a result of this, they cannot be allowed to generate
+ packets that would overflow the other endpoints' MaxDatagram size
+ after the UDPTL stack adds any error correction information. With
+ this patch, the application is told the maximum *IFP* size it can
+ generate, based on a calculation using the far end MaxDatagram
+ size and the active error correction mode on the T.38 session.
+ The same is true for sending *our* MaxDatagram size to the remote
+ endpoint; it is computed from the value that the application says
+ it can accept (for a single IFP packet) combined with the active
+ error correction mode. 2) All treatment of T.38 session
+ parameters as 'capabilities' in chan_sip has been removed; these
+ parameters are not at all like audio/video stream capabilities.
+ There are strict rules to follow for computing an answer to a
+ T.38 offer, and chan_sip now follows those rules, using the
+ desired parameters from the application (or channel) that wants
+ to accept the T.38 negotiation. 3) chan_sip now stores and
+ forwards ast_control_t38_parameters structures for tracking 'our'
+ and 'their' T.38 session parameters; this greatly simplifies
+ negotiation, especially for pass-through calls. 4) Since T.38
+ negotiation without specifying parameters or receiving the final
+ negotiated parameters is not very worthwhile, the AST_CONTROL_T38
+ control frame has been removed. A note has been added to
+ UPGRADE.txt about this removal, since any out-of-tree
+ applications that use it will no longer function properly until
+ they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review:
+ https://reviewboard.asterisk.org/r/310/
+
+2009-07-23 19:34 +0000 [r208388] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 208386 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208386 | mmichelson | 2009-07-23 14:24:21 -0500 (Thu, 23 Jul
+ 2009) | 17 lines Fix a problem where a 491 response could be sent
+ out of dialog. This generalizes the fix for issue 13849. The
+ initial fix corrected the problem that Asterisk would reply with
+ a 491 if a reinvite were received from an endpoint and we had not
+ yet received an ACK from that endpoint for the initial INVITE it
+ had sent us. This expansion also allows Asterisk to appropriately
+ handle an INVITE with authorization credentials if Asterisk had
+ not received an ACK from the previous transaction in which
+ Asterisk had responded to an unauthorized INVITE with a 407.
+ (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch
+ uploaded by mmichelson (license 60) Tested by: klaus3000 ........
+
+2009-07-23 19:21 +0000 [r208383] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 208380 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r208380 | jpeeler | 2009-07-23 14:19:53 -0500 (Thu, 23
+ Jul 2009) | 6 lines Only set the priindication setting when not
+ performing a reload (closes issue #14696) Reported by: fdecher
+ ........
+
+2009-07-23 16:29 +0000 [r208314] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 208312 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208312 | mmichelson | 2009-07-23 11:29:18 -0500 (Thu, 23 Jul
+ 2009) | 3 lines Remove inaccurate XXX comment. ........
+
+2009-07-23 15:59 +0000 [r208267] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
+ Fix sending of interface identifier unconditionally in sig_pri
+ The wrong logic was being used in chan_dahdi to convert a
+ sig_pri_chan to the proper libpri channel number. The most
+ significant bit must only be set only when trunk groups are being
+ used. (closes issue #15452) Reported by: alecdavis Patches:
+ bug15452.patch uploaded by jpeeler (license 325) Tested by:
+ alecdavis
+
+2009-07-23 15:46 +0000 [r208229-208263] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 208262 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208262 | mmichelson | 2009-07-23 10:43:07 -0500 (Thu, 23 Jul
+ 2009) | 8 lines Properly handle 183 responses which do not
+ contain an SDP. (closes issue #15442) Reported by: ffloimair
+ Patches: 15442.patch uploaded by mmichelson (license 60) Tested
+ by: tkarl, ffloimair ........
+
+ * channels/chan_sip.c: Fix potential crash if p->owner is NULL.
+ Problem was observed when a call-forwarding loop was accidentally
+ configured. ABE-1906
+
+2009-07-23 01:31 +0000 [r208193] Russell Bryant <russell@digium.com>
+
+ * main/cel.c: Resolve compiler warning on mac.
+
+2009-07-22 22:42 +0000 [r208155] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Reset the fax buffers back to default
+ settings regardless of signaling in use - Pointed out by Matt F.
+ Also in the case of not using a signaling module, set the law
+ back to the default as well.
+
+2009-07-22 22:35 +0000 [r208151] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/compat.h, main/strcompat.c,
+ main/asterisk.exports: Merged revisions 208083 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208083 | tilghman | 2009-07-22 15:23:53 -0500 (Wed, 22 Jul 2009)
+ | 4 lines Export symbols for functions included in our
+ compatibility headers. (closes issue #15556) Reported by: smw1218
+ ........
+
+2009-07-22 21:43 +0000 [r208113] Jason Parker <jparker@digium.com>
+
+ * apps/app_festival.c: Restore an int declaration on PPC platforms.
+ This x is one crafty little bugger... It was used for 2 different
+ things (one of which was only done on PPC) in 1.4. One of the
+ uses were removed in trunk, and with it went the declaration.
+ (closes issue #14038) Reported by: ffloimair
+
+2009-07-22 16:49 +0000 [r208052] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_realtime.c: Clarify documentation on 'realtime update2'
+ to show more than one condition. (closes issue #15357) Reported
+ by: snuffy Patches: bug_fix_doc_update2.diff uploaded by snuffy
+ (license 35) (slightly modified by me)
+
+2009-07-22 14:35 +0000 [r208018] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/channel.h: Remove trailing whitespace.
+
+2009-07-22 14:35 +0000 [r208017] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_directed_pickup.c: Fix the crash in directed pickups.
+ For real this time. A shallow pointer copy was causing an
+ ast_party_connected_line structure to be freed multiple times,
+ thus causing a crash. (closes issue #15441) Reported by:
+ lmsteffan Patches: 15441.patch uploaded by mmichelson (license
+ 60) Tested by: lmsteffan
+
+2009-07-21 22:51 +0000 [r207950] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/sig_pri.c: Do not dial digits when none were specified
+ for sig_pri based calls (closes issue #15524) Reported by:
+ elguero Patches: pri-sig-no-dest-set.patch uploaded by elguero
+ (license 37)
+
+2009-07-21 22:45 +0000 [r207946] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_strings.c: Merged revisions 207945 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21
+ Jul 2009) | 8 lines Force an error if a blank is passed to QUOTE
+ (because the documentation states the argument is not optional).
+ This change makes URIENCODE and QUOTE behave similarly, since the
+ documentation states that the argument is not optional, for both.
+ (closes issue #15439) Reported by: pkempgen Patches:
+ 20090706__issue15439.diff.txt uploaded by tilghman (license 14)
+ ........
+
+2009-07-21 22:24 +0000 [r207934] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: whitespace fix only
+
+2009-07-21 22:22 +0000 [r207925] Russell Bryant <russell@digium.com>
+
+ * doc/CODING-GUIDELINES: Note that we use tabs instead of spaces
+ for indentation. I'm surprised this was never actually in here...
+
+2009-07-21 22:02 +0000 [r207854-207902] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Fix my_is_off_hook to check rxbits only
+ for FXS signaling
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+ 207827 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009)
+ | 9 lines Wait for wink before dialing when using E&M wink
+ signaling There was already code for other signaling types in
+ dahdi_handle_event to handle dialing if a dial operation dial
+ string was present. Simply add SIG_EMWINK to the list. (closes
+ issue #14434) Reported by: araasch ........
+
+2009-07-21 14:29 +0000 [r207723] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c, /: Merged revisions 207714 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207714 | mmichelson | 2009-07-21 09:26:00 -0500 (Tue, 21 Jul
+ 2009) | 5 lines Document default timeout for AMI originations.
+ AST-224 ........
+
+2009-07-21 13:28 +0000 [r207680] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules,
+ res/Makefile, pbx/Makefile, Makefile.rules, channels/Makefile,
+ doc/video_console.txt, Makefile, utils/Makefile, codecs/Makefile,
+ agi/Makefile, addons/Makefile, funcs/Makefile,
+ codecs/lpc10/Makefile, main/db1-ast/Makefile: Merged revisions
+ 207647 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul
+ 2009) | 12 lines Ensure that user-provided CFLAGS and LDFLAGS are
+ honored. This commit changes the build system so that
+ user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to
+ the compiler/linker *after* all flags provided by the build
+ system itself, so that the user can effectively override the
+ build system's flags if desired. In addition, ASTCFLAGS and
+ ASTLDFLAGS can now be provided *either* in the environment before
+ running 'make', or as variable assignments on the 'make' command
+ line. As a result, the use of COPTS and LDOPTS is no longer
+ necessary, so they are no longer documented, but are still
+ supported so as not to break existing build systems that supply
+ them when building Asterisk. ........
+
+2009-07-20 23:08 +0000 [r207522-207551] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_directed_pickup.c: Okay, that didn't fix the crash. It
+ didn't really do anything useful.
+
+ * apps/app_directed_pickup.c: Initialize connected line instance
+ when doing a directed pickup. This helps to prevent a crash which
+ may occur due to our freeing garbage due to a struct being
+ uninitialized.
+
+2009-07-20 20:45 +0000 [r207484] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: reg->username is parsed only once on sip
+ reload The registration string can contain an expanded user
+ portion of the form user@domain. This expanded user portion was
+ stored in reg->username and parsed each time there is a
+ registration refresh. Now, the domain portion of the user is
+ parsed and stored separately in the regdomain field. (closes
+ issue #14331) Reported by: Nick_Lewis Patches:
+ chan_sip.c.domainparse3.patch uploaded by Nick (license 657)
+ Tested by: Nick_Lewis, dvossel
+
+2009-07-20 19:48 +0000 [r207424] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 207423 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul
+ 2009) | 33 lines Answer video SDP offers properly when
+ videosupport is not enabled. Copied from Review board: In issue
+ 12434, the reporter describes a situation in which audio and
+ video is offered on the call, but because videosupport is
+ disabled in sip.conf, Asterisk gives no response at all to the
+ video offer. According to RFC 3264, all media offers should have
+ a corresponding answer. For offers we do not intend to actually
+ reply to with meaningful values, we should still reply with the
+ port for the media stream set to 0. In this patch, we take note
+ of what types of media have been offered and save the information
+ on the sip_pvt. The SDP in the response will take into account
+ whether media was offered. If we are not otherwise going to
+ answer a media offer, we will insert an appropriate m= line with
+ the port set to 0. It is important to note that this patch is
+ pretty much a bandage being applied to a broken bone. The patch
+ *only* helps for situations where video is offered but
+ videosupport is disabled and when udptl_pt is disabled but T.38
+ is offered. Asterisk is not guaranteed to respond to every media
+ offer. Notable cases are when multiple streams of the same type
+ are offered. The 2 media stream limit is still present with this
+ patch, too. In trunk and the 1.6.X branches, things will be a bit
+ different since Asterisk also supports text in SDPs as well.
+ (closes issue #12434) Reported by: mnnojd Review:
+ https://reviewboard.asterisk.org/r/311 Review:
+ https://reviewboard.asterisk.org/r/313 ........
+
+2009-07-20 16:36 +0000 [r207361] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 207360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009)
+ | 9 lines Only do the chan->fdno check in ast_read() in a
+ developer build. I changed this check to only happen in a
+ dev-mode build. I also added a comment explaining what is going
+ on. I also made it so that detection of this situation does not
+ affect ast_read() operation. (closes issue #14723) Reported by:
+ seadweller ........
+
+2009-07-18 04:17 +0000 [r207318] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, CHANGES: Merged 207316 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
+ .......... r207316 | rmudgett | 2009-07-17 23:05:05 -0500 (Fri,
+ 17 Jul 2009) | 20 lines Fixed incoming calls being matched to
+ MSNs without type-of-number prefix added. For an incoming ISDN
+ call the dialed.number is incorrectly matched against the
+ configured MSNs in misdn.conf. The numbers passed to the dialplan
+ include the configured prefix for the dialed.number_type, whereas
+ the check against the configured MSNs (to decide if the call is
+ accepted at all), is executed without the configured prefix.
+ e.g., dialed.number = 241168020, TON = national, configured
+ national prefix is "0". (This is the TON which is used by ISDN
+ providers in the Netherlands.) In chan_misdn.c:cb_events() in
+ case EVENT_SETUP the call to misdn_cfg_is_msn_valid() uses the
+ unnormalized number 241168020, but 57 lines later the call to
+ read_config() adds the prefix, and the dialed.number is now
+ 0241168020, which is then used in the dialplan.
+ misdn_cfg_is_msn_valid() must use the normalized number, too.
+ JIRA ABE-1912
+
+2009-07-18 04:16 +0000 [r207317] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Flag field in wrong position. Reported by
+ "Hoggins!" on asterisk-dev list.
+
+2009-07-18 01:31 +0000 [r207285] Richard Mudgett <rmudgett@digium.com>
+
+ * /: Recorded merge of revisions 145293,158010 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r145293 | rmudgett | 2008-09-30 18:55:24 -0500
+ (Tue, 30 Sep 2008) | 54 lines channels/chan_misdn.c
+ channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk
+ to make merging easier later. ........ r145200 | rmudgett |
+ 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines *
+ Miscellaneous formatting changes to make v1.4 and trunk more
+ merge compatible in the mISDN area. channels/chan_misdn.c *
+ Eliminated redundant code in cb_events() EVENT_SETUP ........
+ r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008)
+ | 9 lines improved helptext of misdn_set_opt. ........ r142181 |
+ rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line
+ Cleaned up comment ........ r138738 | rmudgett | 2008-08-18
+ 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines
+ channels/chan_misdn.c * Made bearer2str() use
+ allowed_bearers_array[] * Made use the causes.h defines instead
+ of hardcoded numbers. * Made use Asterisk presentation indicator
+ values if either of the mISDN presentation or screen options are
+ negative. * Updated the misdn_set_opt application option
+ descriptions. * Renamed the awkward Caller ID presentation
+ misdn_set_opt application option value not_screened to
+ restricted. Deprecated the not_screened option value.
+ channels/misdn/isdn_lib.c * Made use the causes.h defines instead
+ of hardcoded numbers. * Fixed some spelling errors and typos. *
+ Added all defined facility code strings to fac2str().
+ channels/misdn/isdn_lib.h * Added doxygen comments to struct
+ misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen
+ comments to struct misdn_stack. channels/misdn_config.c
+ configs/misdn.conf.sample * Updated the mISDN presentation and
+ screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex)
+ * Updated the misdn_set_opt application option descriptions. *
+ Fixed some spelling errors and typos. ................ r158010 |
+ rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines
+ Merged revision 157977 from
+ https://origsvn.digium.com/svn/asterisk/team/group/issue8824
+ ........ Fixes JIRA ABE-1726 The dial extension could be empty if
+ you are using MISDN_KEYPAD to control ISDN provider features.
+ ................
+
+2009-07-17 22:29 +0000 [r207255] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/voicemail_odbc_postgresql.txt: Add flag here, too (as
+ requested by jsmith)
+
+2009-07-17 22:07 +0000 [r207225] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: fixes an error in r203638 CEL commit
+ (closes issue #15525) Reported by: elguero Patches:
+ iax2-double-unlock.patch uploaded by elguero (license 37)
+ 15525.diff uploaded by dvossel (license 671) Tested by: dvossel
+
+2009-07-17 22:04 +0000 [r207224] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/tex/odbcstorage.tex, UPGRADE.txt: Document the "flag" field
+ in the voicemessages table.
+
+2009-07-17 19:37 +0000 [r207095-207156] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 207155 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r207155 | jpeeler | 2009-07-17 14:36:19 -0500 (Fri, 17
+ Jul 2009) | 7 lines Fix format specifier to print out an unsigned
+ long long. Yep, it's even ifdefed out code. But it made it to the
+ RR list... (closes issue #14726) Reported by: lmadsen ........
+
+ * configs/chan_dahdi.conf.sample: Update some missing allowed
+ options for overlapdial
+
+2009-07-17 17:51 +0000 [r207029] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: sip option flags handled incorrectly (closes
+ issue #15376) Reported by: Takehiko Ooshima Tested by: dvossel,
+ Takehiko_Ooshima
+
+2009-07-17 17:02 +0000 [r206998] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c: Fix segfault in
+ sig_analog when using callwaiting, respect callwaiting options
+ Sig_analog handles allocating the sub channel for callwaiting, so
+ no longer try to do it in chan_dahdi. Modified analog_alloc_sub
+ to only mark the sub as allocated upon success of the alloc_sub
+ callback, which was responsible for the segfault. Also, the
+ callwaiting and callwaitingcallerid options were being
+ unconditionally set to true. Now, the options are properly set
+ from chan_dahdi.conf. (closes issue #15508) Reported by: elguero
+ Tested by: elguero
+
+2009-07-17 16:13 +0000 [r206868-206939] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 206938 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009)
+ | 14 lines SIP incorrect From: header information when callpres
+ is prohib Some ITSP make use of the "Anonymous" display name to
+ detect a requirement to withhold caller id across the PSTN. This
+ does not work if the display name is "Unknown". (closes issue
+ #14465) Reported by: Nick_Lewis Patches:
+ chan_sip.c-callerpres.patch uploaded by Nick (license 657)
+ chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license
+ 671) Tested by: Nick_Lewis, dvossel ........
+
+ * funcs/func_timeout.c: TIMEOUT(absolute) returned negative value.
+ (closes issue #15513) Reported by: ys
+
+ * configs/iax.conf.sample, /: Merged revisions 206872 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16
+ Jul 2009) | 6 lines error in iax.conf related IP-based access
+ control (closes issue #15518) Reported by: pkempgen ........
+
+ * /, main/callerid.c: Merged revisions 206867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206867 | dvossel | 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009)
+ | 8 lines avoid segfault caused by user error If the CALLERPRES()
+ dialplan function is set to nothing, a segfault occurs. This is
+ user error to begin with, but I'd rather see a cli warning
+ message than have Asterisk crash on me. ........
+
+2009-07-16 16:51 +0000 [r206808] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_realtime.c: Merged revisions 206807 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r206807 | tilghman | 2009-07-16 11:27:35 -0500 (Thu, 16
+ Jul 2009) | 6 lines Fix a memory leak. (closes issue #15517)
+ Reported by: adomjan Patches:
+ func_realtime.c-ast_variable_destroy.diff uploaded by adomjan
+ (license 487) ........
+
+2009-07-15 22:04 +0000 [r206768] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Session timer were not activated if
+ Supported header field in INVITE had both "timer" and other
+ options. (closes issue #15403) Reported by: makoto Patches:
+ sip-session-timer.patch uploaded by makoto (license 38)
+
+2009-07-15 22:02 +0000 [r206767] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h, channels/sig_pri.c: The dialing flag was
+ mistakingly removed from sig_pri. This readds the proper setting
+ of the flag and is really a continuation of r205731. The flag was
+ being set properly in sig_analog, but use of the newly added
+ set_dialing callback allowed for some simplification in
+ chan_dahdi. (closes issue #15486) Reported by: rmudgett
+
+2009-07-15 21:14 +0000 [r206707] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib_intern.h, /, channels/misdn/isdn_lib.c:
+ Merged revisions 206706 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r206706 | rmudgett | 2009-07-15 15:44:55 -0500
+ (Wed, 15 Jul 2009) | 26 lines Merged revision 206700 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
+ .......... Fixed chan_misdn crash because mISDNuser library is
+ not thread safe. With Asterisk the mISDNuser library is driven by
+ two threads concurrently: 1.
+ channels/misdn/isdn_lib.c::manager_event_handler() 2.
+ channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher() Calls
+ into the library are done concurrently and recursively from
+ isdn_lib.c. Both threads can fiddle with the master/child
+ layer3_proc_t lists. One thread may traverse the list when the
+ other interrupts it and then removes the list element which the
+ first thread was currently handling. This is exactly what caused
+ the crash. About 60 calls were needed to a Gigaset CX475 before
+ it occurred once. This patch adds locking when calling into the
+ mISDNuser library. This also fixes some cb_log calls with wrong
+ port parameter. JIRA ABE-1913 Patches: misdn-locking.patch
+ (Modified with mostly cosmetic changes) ..........
+ ................
+
+2009-07-15 20:20 +0000 [r206702] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: callerid(num) is wrong when username is
+ missing A domain only sip uri <sip:123.123.123.123> would return
+ 123.123.123.123 as callid num. Now, if the username is missing
+ from a uri, the callerid num field is left empty. (closes issue
+ #15476) Reported by: viraptor
+
+2009-07-15 16:00 +0000 [r206636] Sean Bright <sean@malleable.com>
+
+ * /, codecs/codec_dahdi.c: Merged revisions 206635 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed,
+ 15 Jul 2009) | 1 line Only print debug info in codec_dahdi if we
+ are asking for it. ........
+
+2009-07-14 20:38 +0000 [r206603] Jeff Peeler <jpeeler@digium.com>
+
+ * configs/chan_dahdi.conf.sample: fix a typo in sample config file
+ for option change
+
+2009-07-14 20:14 +0000 [r206567] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c, contrib/scripts/meetme.sql: Document all
+ meetme realtime fields, and in the process, make some field
+ lengths more consistent. (closes issue #15493) Reported by: lasko
+ Patches: meetme.diff uploaded by lasko (license 833)
+
+2009-07-14 20:01 +0000 [r206566] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h: Restore some missing functionality to
+ sig_analog. The main purpose of this commit is to restore missing
+ functionality present in the ss_thread before all the sig related
+ work was done. Two of the biggest missing things were distinctive
+ ring detection and cid handling for V23. fxsoffhookstate and
+ associated mwi variables have been moved inside sig_analog as
+ they were not being set properly as well.
+
+2009-07-14 17:03 +0000 [r206490] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c: I AM A TERRIBLE PERSON
+
+2009-07-14 17:01 +0000 [r206489] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
+ channels/misdn/isdn_lib.c: Merged revisions 206487 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14
+ Jul 2009) | 28 lines Fixes several call transfer issues with
+ chan_misdn. * issue #14355 - Crash if attempt to transfer a call
+ to an application. Masquerade the other pair of the four asterisk
+ channels involved in the two calls. The held call already must be
+ a bridged call (not an applicaton) or it would have been
+ rejected. * issue #14692 - Held calls are not automatically
+ cleared after transfer. Allow the core to initate disconnect of
+ held calls to the ISDN port. This also fixes a similar case where
+ the party on hold hangs up before being transferred or taken off
+ hold. * JIRA ABE-1903 - Orphaned held calls left in
+ music-on-hold. Do not simply block passing the hangup event on
+ held calls to asterisk core. * Fixed to allow held calls to be
+ transferred to ringing calls. Previously, held calls could only
+ be transferred to connected calls. * Eliminated unused call
+ states to simplify hangup code. * Eliminated most uses of
+ "holded" because it is not a word. (closes issue #14355) (closes
+ issue #14692) Reported by: sodom Patches:
+ misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664)
+ Tested by: rmudgett ........
+
+2009-07-14 16:09 +0000 [r206455] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c: Reset the sentringing indication when redirects
+ occur. If a redirecting control frame is processed or a call
+ forward occurs, we need to reset the sentringing flag so that we
+ can send another ringing indication to the phone that may contain
+ a connected line update. AST-164
+
+2009-07-14 14:51 +0000 [r206386] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 206385 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r206385 | russell | 2009-07-14 09:48:00 -0500
+ (Tue, 14 Jul 2009) | 13 lines Merged revisions 206384 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009)
+ | 6 lines Ensure apathetic replies are sent out on the proper
+ socket. chan_iax2 supports multiple address bindings. The
+ send_apathetic_reply() function did not attempt to send its
+ response on the same socket that the incoming message came in on.
+ ........ ................
+
+2009-07-14 00:48 +0000 [r206341] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
+ revisions 206284 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009)
+ | 4 lines Fix some memory leaks in chan_misdn. JIRA ABE-1911
+ ........
+
+2009-07-13 23:26 +0000 [r206280] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: dns lookup of peername rather than peer's
+ host in transmit_register() (closes issue #15052) Reported by:
+ fsantulli Patches: chan_sip_bug_15052_[20090626204511].patch
+ uploaded by fsantulli (license 818) Tested by: fsantulli
+
+2009-07-13 18:46 +0000 [r206225] Sean Bright <sean@malleable.com>
+
+ * contrib/upstart/asterisk.upstart-0.3.9: Make sure that since we
+ are passing -c to asterisk that we have a console. Without this
+ line, Asterisk will busy-loop trying to read and write to
+ /dev/null (woops... my bad).
+
+2009-07-13 16:23 +0000 [r206185] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Remove reference to non-existent help file
+ (closes issue #15427) Reported by: brushtyler Patches:
+ app_voicemail.c.diff uploaded by brushtyler (license 821)
+
+2009-07-13 14:06 +0000 [r206092-206094] Kevin P. Fleming <kpfleming@digium.com>
+
+ * .cleancount: Bump up cleancount so that existing checkouts will
+ update themselves properly for the 'Addons' -> 'ADDONS' change.
+
+ * addons/Makefile: Make the menuselect category for Add-Ons
+ consistent with the other directories (uppercase).
+
+2009-07-11 19:30 +0000 [r206021-206049] Russell Bryant <russell@digium.com>
+
+ * CHANGES: note the security events API in CHANGES
+
+ * doc/tex/security-events.tex (added), tests/test_security_events.c
+ (added), main/manager.c, main/security_events.c (added),
+ include/asterisk/event_defs.h, main/event.c,
+ include/asterisk/security_events.h (added), doc/tex/asterisk.tex,
+ include/asterisk/security_events_defs.h (added),
+ res/res_security_log.c (added), tests/test_ami_security_events.sh
+ (added): Add an API for reporting security events, and a security
+ event logging module. This commit introduces the security events
+ API. This API is to be used by Asterisk components to report
+ events that have security implications. A simple example is when
+ a connection is made but fails authentication. These events can
+ be used by external tools manipulate firewall rules or something
+ similar after detecting unusual activity based on security
+ events. Inside of Asterisk, the events go through the ast_event
+ API. This means that they have a binary encoding, and it is easy
+ to write code to subscribe to these events and do something with
+ them. One module is provided that is a subscriber to these events
+ - res_security_log. This module turns security events into a
+ parseable text format and sends them to the "security" logger
+ level. Using logger.conf, these log entries may be sent to a
+ file, or to syslog. One service, AMI, has been fully updated for
+ reporting security events. AMI was chosen as it was a fairly
+ straight forward service to convert. The next target will be
+ chan_sip. That will be more complicated and will be done as its
+ own project as the next phase of security events work. For more
+ information on the security events framework, see the
+ documentation generated from doc/tex/. "make asterisk.pdf"
+ Review: https://reviewboard.asterisk.org/r/273/
+
+2009-07-10 21:42 +0000 [r205985] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: SIP register not using peer's outbound proxy
+ If callbackextension is defined for a peer it successfully causes
+ a registration to occur, but the registration ignores the
+ outboundproxy settings for the peer. This patch allows the peer
+ to be passed to obproxy_get() in transmit_register(). (closes
+ issue #14344) Reported by: Nick_Lewis Patches:
+ callbackextension_peer_trunk.diff uploaded by dvossel (license
+ 671) Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/294/
+
+2009-07-10 18:44 +0000 [r205939] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c: Update comments about the level of T.38 support in
+ Asterisk.
+
+2009-07-10 17:39 +0000 [r205878] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 205877 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r205877 | mmichelson | 2009-07-10 12:39:13 -0500
+ (Fri, 10 Jul 2009) | 23 lines Merged revisions 205776 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500
+ (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul
+ 2009) | 10 lines Ensure that outbound NOTIFY requests are
+ properly routed through stateful proxies. With this change, we
+ make note of Record-Route headers present in any SUBSCRIBE
+ request that we receive so that our outbound NOTIFY requests will
+ have the proper Route headers in them. (closes issue #14725)
+ Reported by: ibc ........ ................ ................
+
+2009-07-10 16:42 +0000 [r205840] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 205804 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009)
+ | 31 lines SIP registration auth loop caused by stale nonce If an
+ endpoint sends two registration requests in a very short period
+ of time with the same nonce, both receive 401 responses from
+ Asterisk, each with a different nonce (the second 401 containing
+ the current nonce and the first one being stale). If the endpoint
+ responds to the first 401, it does not match the current nonce so
+ Asterisk sends a third 401 with a newly generated nonce (which
+ updates the current nonce)... Now if the endpoint responds to the
+ second 401, it does not match the current nonce either and
+ Asterisk sends a fourth 401 with a newly generated nonce... This
+ loop goes on and on. There appears to be a simple fix for this.
+ If the nonce from the request does not match our nonce, but is a
+ good response to a previous nonce, instead of sending a 401 with
+ a newly generated nonce, use the current one instead. This breaks
+ the loop as the nonce is not updated until a response is
+ received. Additional logic has been added to make sure no nonce
+ can be responded to twice though. (closes issue #15102) Reported
+ by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license
+ 809) nonce_sip.diff uploaded by dvossel (license 671) Tested by:
+ Jamuel Review: https://reviewboard.asterisk.org/r/289/ ........
+
+2009-07-10 16:00 +0000 [r205780] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_fax.c: Eliminate extraneous LOG_DEBUG messages generated
+ by app_fax. The transmit_audio() and transmit_t38() functions in
+ app_fax have processing loops that are supposed to wait for
+ frames to arrive on the channel and then handle them, but they
+ also have short timeouts so that the loops can have watchdog
+ timers and do other required processing. This commit changes the
+ loops to not actually call ast_read() and attempt to process the
+ returned frame unless a frame actually arrived, eliminating
+ hundreds of LOG_DEBUG messages and slightly improving
+ performance.
+
+2009-07-10 15:56 +0000 [r205776] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 205775 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul
+ 2009) | 10 lines Ensure that outbound NOTIFY requests are
+ properly routed through stateful proxies. With this change, we
+ make note of Record-Route headers present in any SUBSCRIBE
+ request that we receive so that our outbound NOTIFY requests will
+ have the proper Route headers in them. (closes issue #14725)
+ Reported by: ibc ........
+
+2009-07-10 15:28 +0000 [r205770] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_fax.c: Fix some remaining T.38 negotiation problems in
+ app_fax. Revision 205696 did not quite fix all the issues with
+ the T.38 negotiation changes and app_fax; this patch corrects
+ them, along with a couple of other minor issues. (closes issue
+ #15480) Reported by: dimas Patches: test2-15480.patch uploaded by
+ dimas (license 88)
+
+2009-07-09 21:32 +0000 [r205700] Matthew Nicholson <mnicholson@digium.com>
+
+ * addons/chan_mobile.c: Fix mbl_fixup() in chan_mobile to update
+ newchan->tech_pvt instead of oldchan. (closes issue #15299)
+ Reported by: nikkk
+
+2009-07-09 21:20 +0000 [r205696] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c, apps/app_fax.c, include/asterisk/frame.h:
+ Repair ability of SendFAX/ReceiveFAX to respond to T.38
+ switchover. Recent changes in T.38 negotiation in Asterisk caused
+ these applications to not respond when the other endpoint
+ initiated a switchover to T.38; this resulted in the T.38
+ switchover failing, and the FAX attempt to be made using an audio
+ connection, instead of T.38 (which would usually cause the FAX to
+ fail completely). This patch corrects this problem, and the
+ applications will now correctly respond to the T.38 switchover
+ request. In addition, the response will include the appopriate
+ T.38 session parameters based on what the other end offered and
+ what our end is capable of. (closes issue #14849) Reported by:
+ afosorio
+
+2009-07-09 20:04 +0000 [r205666] Matthew Nicholson <mnicholson@digium.com>
+
+ * funcs/func_odbc.c: Convert func_odbc to use
+ ast_dummy_alloc_channel() Review:
+ https://reviewboard.asterisk.org/r/290/
+
+2009-07-09 16:19 +0000 [r205600] David Vossel <dvossel@digium.com>
+
+ * /, include/asterisk/time.h: Merged revisions 205599 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09
+ Jul 2009) | 2 lines Changing ast_samp2tv to not use floating
+ point. ........
+
+2009-07-09 14:10 +0000 [r205532-205562] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/cel.c: make this compile again under devmode
+
+ * main/ssl.c: pthread_self returns a pthread_t which is not an
+ unsigned int on all pthread implementations. Casting it to an
+ unsigned int fixes compiler warnings. Tested on OpenBSD and Linux
+ both 32 and 64 bit
+
+2009-07-08 23:19 +0000 [r205479] David Vossel <dvossel@digium.com>
+
+ * res/res_rtp_asterisk.c, /, channels/chan_iax2.c,
+ include/asterisk/frame.h: Merged revisions 205471 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08
+ Jul 2009) | 10 lines Fixes 8khz assumptions Many calculations
+ assume 8khz is the codec rate. This is not always the case. This
+ patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am
+ sure there are other areas that make this assumption as well.
+ Review: https://reviewboard.asterisk.org/r/306/ ........
+
+2009-07-08 23:07 +0000 [r205469] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c: Fix a CEL related regression with hints updating by
+ subscribing to AST_DEVICE_STATE instead of
+ AST_DEVICE_STATE_CHANGED. (closes issue #15440) Reported by:
+ lmsteffan
+
+2009-07-08 22:15 +0000 [r205410-205412] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/devicestate.h, main/pbx.c, /,
+ main/devicestate.c, include/asterisk/pbx.h: Merged revisions
+ 205409 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009)
+ | 6 lines moving ast_devstate_to_extenstate to pbx.c from
+ devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This
+ change fixes a compile time error with chan_vpb as well. ........
+
+ * main/devicestate.c: missing comma in devstatestring array
+
+2009-07-08 19:26 +0000 [r205350] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 205349 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul
+ 2009) | 14 lines Prevent phantom calls to queue members. If a
+ caller were to hang up while a periodic announcement or position
+ were being said, the return value for those functions would
+ incorrectly indicate that the caller was still in the queue. With
+ these changes, the problem does not occur. (closes issue #14631)
+ Reported by: latinsud Patches: queue_announce_ghost_call2.diff
+ uploaded by latinsud (license 745) (with small modification from
+ me) ........
+
+2009-07-08 18:19 +0000 [r205291] Jason Parker <jparker@digium.com>
+
+ * config.sub, /, config.guess: Merged revisions 205288 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul
+ 2009) | 1 line Update config.guess and config.sub from the
+ savannah.gnu.org git repo. ........
+
+2009-07-08 17:26 +0000 [r205254] David Brooks <dbrooks@digium.com>
+
+ * main/features.c: Fixes Park() argument handling Park() was not
+ respecting the arguments passed to it. Any
+ extension/context/priority given to it was being ignored. This
+ patch remedies this. (closes issue #15380) Reported by: DLNoah
+
+2009-07-08 16:59 +0000 [r205221] Tilghman Lesher <tlesher@digium.com>
+
+ * main/say.c: Oops, fixing build
+
+2009-07-08 16:54 +0000 [r205216] David Vossel <dvossel@digium.com>
+
+ * /, include/asterisk/time.h: Merged revisions 205215 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08
+ Jul 2009) | 10 lines ast_samp2tv needs floating point for 16khz
+ audio In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is
+ 16000. The .5 is currently stripped off because we don't
+ calculate using floating points. This causes madness with 16khz
+ audio. (issue ABE-1899) Review:
+ https://reviewboard.asterisk.org/r/305/ ........
+
+2009-07-08 16:43 +0000 [r205214] Sean Bright <sean@malleable.com>
+
+ * utils/muted.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac, main/dns.c: Fix a few compilation problems found
+ when building Asterisk against uClibc.
+
+2009-07-08 16:27 +0000 [r205196] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/say.c: Merged revisions 205188 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205188 | tilghman | 2009-07-08 11:26:15 -0500 (Wed, 08 Jul 2009)
+ | 2 lines Add redirection warnings for the invalid language codes
+ previously removed. ........
+
+2009-07-08 15:56 +0000 [r205120-205151] Russell Bryant <russell@digium.com>
+
+ * main/ssl.c: Use tabs instead of spaces for indentation.
+
+ * res/res_crypto.c, main/ssl.c (added),
+ include/asterisk/_private.h, res/res_jabber.c, main/asterisk.c:
+ Move OpenSSL initialization to a single place, make library usage
+ thread-safe. While doing some reading about OpenSSL, I noticed a
+ couple of things that needed to be improved with our usage of
+ OpenSSL. 1) We had initialization of the library done in multiple
+ modules. This has now been moved to a core function that gets
+ executed during Asterisk startup. We already link OpenSSL into
+ the core for TCP/TLS functionality, so this was the most logical
+ place to do it. 2) OpenSSL is not thread-safe by default.
+ However, making it thread safe is very easy. We just have to
+ provide a couple of callbacks. One callback returns a thread ID.
+ The other handles locking. For more information, start with the
+ "Is OpenSSL thread-safe?" question on the FAQ page of
+ openssl.org.
+
+2009-07-08 14:45 +0000 [r205118] Luigi Rizzo <rizzo@icir.org>
+
+ * bootstrap.sh: FreeBSD now has autoconf 2.62 in the ports, 2.61
+ has disappeared.
+
+2009-07-07 21:10 +0000 [r205086] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Permit setting custom headers from the peer
+ definition. (closes issue #14059) Reported by: fnordian
+
+2009-07-07 18:24 +0000 [r205014-205047] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/sig_analog.c: Fix a deadlock in sig_analog
+
+ * channels/sig_analog.c: Add CEL transfer events to analog
+ (chan_dahdi) transfers.
+
+2009-07-06 21:37 +0000 [r204986] Tilghman Lesher <tlesher@digium.com>
+
+ * addons/res_config_mysql.c: Merged revisions 981 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk-addons/branches/1.4
+ ........ r981 | tilghman | 2009-07-06 16:30:13 -0500 (Mon, 06 Jul
+ 2009) | 7 lines Don't reset reconnect time, unless a reconnect
+ really occurred. (closes issue #15375) Reported by: kowalma
+ Patches: 20090628__issue15375.diff.txt uploaded by tilghman
+ (license 14) Tested by: kowalma, jacco ........
+
+2009-07-06 13:38 +0000 [r204948] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/channel.c: Improve handling of AST_CONTROL_T38 and
+ AST_CONTROL_T38_PARAMETERS for non-T.38-capable channels. This
+ change allows applications that request T.38 negotiation on a
+ channel that does not support it to get the proper indication
+ that it is not supported, rather than thinking that negotiation
+ was started when it was not.
+
+2009-07-03 15:44 +0000 [r204893-204919] Sean Bright <sean@malleable.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ channels/sig_pri.c: Add a configure check for Reverse Charging
+ Indication support in LibPRI. Also go back and wrap all of the
+ places that use the specific reverse charge APIs with
+ preprocessor conditionals.
+
+ * include/asterisk/rtp_engine.h: Wrap rtp_engine.h header comments
+ to 80 characters.
+
+2009-07-02 22:01 +0000 [r204835] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 204834 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02
+ Jul 2009) | 10 lines Removed confusing warning message "Got Busy
+ in Connected State" If an incoming mISDN call is answered with
+ the Answer application and a subsequent Dial gets a busy endpoint
+ then it is valid for that already connected channel to get the
+ busy indication. Asterisk will play the busy tones until the
+ dialplan plays something else or hangs up the call. (closes issue
+ #11974) Reported by: fvdb ........
+
+2009-07-02 20:37 +0000 [r204807] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, main/features.c: Moved trigger for BRIDGE_END CEL
+ event so that it is more accurate.
+
+2009-07-02 17:46 +0000 [r204749] Sean Bright <sean@malleable.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, funcs/func_channel.c, CHANGES,
+ channels/sig_pri.c: Support setting and receiving Reverse
+ Charging Indication over ISDN PRI. This is a continuation of
+ revision 885 to LibPRI (Capture and expose the Reverse Charging
+ Indication IE on ISDN PRI) which added the ability to get/set
+ Reverse Charging Indication in LibPRI. This patch adds the
+ ability to specify RCI on the outbound leg of a PRI call from
+ within Asterisk, by prefixing the dialed number with a capital
+ 'C' like: ...,Dial(DAHDI/g1/C4445556666) And to read it off an
+ inbound channel: exten => s,1,Set(RCI=${CHANNEL(reversecharge)})
+ Thanks again to rmudgett for the thorough review. (closes issue
+ #13760) Reported by: mrgabu Review:
+ https://reviewboard.asterisk.org/r/303/
+
+2009-07-02 16:03 +0000 [r204710] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/devicestate.h, main/pbx.c, /,
+ main/devicestate.c: Merged revisions 204681 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009)
+ | 14 lines Improved mapping of extension states from combined
+ device states. This fixes a few issues with incorrect extension
+ states and adds a cli command, core show device2extenstate, to
+ display all possible state mappings. (closes issue #15413)
+ Reported by: legart Patches: exten_helper.diff uploaded by
+ dvossel (license 671) Tested by: dvossel, legart, amilcar Review:
+ https://reviewboard.asterisk.org/r/301/ ........
+
+2009-07-01 19:47 +0000 [r204654] Ryan Brindley <rbrindley@digium.com>
+
+ * configs/http.conf.sample: - cfgbasic.html has been replaced by
+ index.html in the GUI for some time now
+
+2009-07-01 16:06 +0000 [r204622] Sean Bright <sean@malleable.com>
+
+ * apps/app_voicemail.c: A bunch of CODING_GUIDELINES related fixes.
+ Not even close to done.
+
+2009-06-30 20:41 +0000 [r204563] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/say.c, UPGRADE.txt: Merged revisions 204556 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30
+ Jun 2009) | 6 lines More incorrect language codes, plus ensuring
+ that regionalizations use the specified language, and not English
+ for grammar. (closes issue #15022) Reported by: greenfieldtech
+ Patches: 20090519__issue15022.diff.txt uploaded by tilghman
+ (license 14) ........
+
+2009-06-30 20:39 +0000 [r204561] Sean Bright <sean@malleable.com>
+
+ * apps/app_voicemail.c: Remove an unnecessary #ifdef
+
+2009-06-30 19:59 +0000 [r204530-204532] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Move the masquerade in
+ local_attended_transfer to a point where we hold the channel
+ lock. Masquerading without the channel's lock held is a
+ *horrible* idea.
+
+ * channels/chan_sip.c: Remove some bogus deadlock avoidance code
+ from local_attended_transfer. First of all, the code was
+ unnecessary. The goal was to lock a channel which was already
+ locked. Second, the assumption of the deadlock avoidance loop was
+ that the sip_pvt was already locked and we were trying to get the
+ channel lock. The problem is that the sip_pvt was unlocked a few
+ lines above. Basically, I'm removing 5 lines of no-op.
+
+2009-06-30 18:48 +0000 [r204475] Jason Parker <jparker@digium.com>
+
+ * /, main/say.c: Merged revisions 204474 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204474 | qwell | 2009-06-30 13:47:06 -0500 (Tue, 30 Jun 2009) |
+ 1 line Fix ast_say_counted_noun to correctly handle Polish. Fix a
+ comment typo in passing. ........
+
+2009-06-30 18:36 +0000 [r204470] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/say.c, UPGRADE.txt, apps/app_voicemail.c: Recorded merge
+ of revisions 204469 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009)
+ | 11 lines "tw" is the language specification for Twi (from
+ Ghana) not Taiwanese. (closes issue #15346) Reported by: volivier
+ Patches: 20090617__issue15346__1.4.diff.txt uploaded by tilghman
+ (license 14) 20090617__issue15346__trunk.diff.txt uploaded by
+ tilghman (license 14) 20090617__issue15346__1.6.0.diff.txt
+ uploaded by tilghman (license 14)
+ 20090617__issue15346__1.6.1.diff.txt uploaded by tilghman
+ (license 14) 20090617__issue15346__1.6.2.diff.txt uploaded by
+ tilghman (license 14) Tested by: volivier ........
+
+2009-06-30 17:22 +0000 [r204417-204440] Russell Bryant <russell@digium.com>
+
+ * configs/res_config_sqlite.conf (removed),
+ configs/res_config_sqlite.conf.sample (added): Rename
+ res_config_sqlite.conf to res_config_sqlite.conf.sample (missing
+ .sample).
+
+ * addons/chan_ooh323.c, configs/chan_ooh323.conf.sample (added),
+ configs/ooh323.conf.sample (removed): Rename ooh323.conf to
+ chan_ooh323.conf, make module support both names
+
+ * configs/mobile.conf.sample (removed), addons/chan_mobile.c,
+ configs/chan_mobile.conf.sample (added): Rename mobile.conf to
+ chan_mobile.conf, make module support old name, too
+
+ * configs/res_config_mysql.conf.sample (added),
+ configs/res_mysql.conf.sample (removed),
+ addons/res_config_mysql.c: Rename res_mysql.conf to
+ res_config_mysql.conf, make module support both
+
+ * Makefile: Make addons build last - this is for Qwell.
+
+ * addons/app_mysql.c, configs/app_mysql.conf.sample (added),
+ configs/mysql.conf.sample (removed): Rename mysql.conf to
+ app_mysql.conf, make module support both names
+
+ * addons/Makefile, addons/cdr_mysql.c (added),
+ addons/cdr_addon_mysql.c (removed): Rename cdr_addon_mysql to
+ cdr_mysql
+
+ * addons/app_mysql.c (added), addons/app_addon_sql_mysql.c
+ (removed), addons/Makefile: Rename app_addon_sql_mysql to
+ app_mysql
+
+2009-06-30 17:04 +0000 [r204415] Kevin P. Fleming <kpfleming@digium.com>
+
+ * build_tools/embed_modules.xml, Makefile.moddir_rules,
+ addons/Makefile: Add-ons related build system improvements.
+ Ensure that add-on modules can be embedded, fix up
+ Makefile.moddir_rules to allow module directory Makefiles to more
+ easily specify the modules to be built, and explicitly list the
+ addons modules in its Makefile, since the module names don't
+ follow any pattern.
+
+2009-06-30 16:40 +0000 [r204413] Russell Bryant <russell@digium.com>
+
+ * autoconf/ast_ext_tool_check.m4, addons/ooh323c/src/oochannels.h,
+ addons/ooh323c/src/printHandler.h, addons/chan_ooh323.c,
+ addons/ooh323c/src/ooq931.h, include/asterisk/autoconfig.h.in,
+ addons/ooh323c/src/ootrace.h, addons/chan_ooh323.h,
+ addons/ooh323c/src/ooasn1.h, configs/res_mysql.conf.sample
+ (added), addons/ooh323c/src/ooStackCmds.c,
+ addons/ooh323c/src/errmgmt.c, addons/ooh323c/src/ooStackCmds.h,
+ addons/ooh323c/src/eventHandler.c,
+ addons/ooh323c/src/h323/H235-SECURITY-MESSAGES.h,
+ addons/mp3/huffman.h, configure,
+ addons/ooh323c/src/eventHandler.h, addons/ooh323cDriver.c,
+ include/asterisk/mod_format.h, addons/mp3/interface.c,
+ doc/tex/asterisk.tex, addons/ooh323cDriver.h,
+ addons/cdr_addon_mysql.c, addons/ooh323c/src/encode.c,
+ addons/mp3/MPGLIB_README,
+ addons/ooh323c/src/h323/H235-SECURITY-MESSAGESEnc.c,
+ configure.ac, doc/tex/chan_mobile.tex (added),
+ addons/ooh323c/src/ooports.c, addons/mp3/mpg123.h,
+ addons/mp3/mpglib.h, addons (added),
+ addons/ooh323c/src/h323/MULTIMEDIA-SYSTEM-CONTROL.c,
+ addons/ooh323c/src/ooports.h, addons/ooh323c/src/memheap.c,
+ Makefile, addons/ooh323c/src/h323/MULTIMEDIA-SYSTEM-CONTROL.h,
+ addons/ooh323c/src/ooh245.c, addons/mp3/common.c,
+ addons/ooh323c/src/memheap.h, addons/ooh323c/src/perutil.c,
+ addons/mp3/decode_i386.c, addons/ooh323c/src/ooh245.h,
+ addons/mp3/dct64_i386.c, addons/ooh323c/src/ooSocket.c,
+ addons/ooh323c/src/h323/H235-SECURITY-MESSAGESDec.c,
+ addons/mp3/layer3.c, addons/ooh323c/src/ooper.h,
+ addons/ooh323c/src/ooCmdChannel.c, addons/ooh323c/src/ooSocket.h,
+ addons/ooh323c/src/ooCommon.h, addons/ooh323c/src/ooCmdChannel.h,
+ addons/ooh323c/COPYING, addons/format_mp3.c,
+ addons/ooh323c/src/Makefile.in, configs/mobile.conf.sample
+ (added), addons/ooh323c/src/ootypes.h, addons/mp3,
+ addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooTimer.c,
+ addons/ooh323c/src/ooLogChan.h, addons/ooh323c/src/dlist.c,
+ addons/ooh323c/src/ooCapability.c, addons/ooh323c/src/oohdr.h,
+ README-addons.txt (added), addons/app_addon_sql_mysql.c,
+ addons/ooh323c/src/ooTimer.h, addons/ooh323c/src/ooCapability.h,
+ addons/ooh323c/src/dlist.h, addons/mp3/Makefile, addons/Makefile,
+ addons/ooh323c/README, addons/ooh323c, doc/tex/cdrdriver.tex,
+ addons/ooh323c/src/h323/H323-MESSAGESEnc.c, addons/chan_mobile.c,
+ configs/cdr_mysql.conf.sample (added),
+ addons/ooh323c/src/ooDateTime.c, addons/ooh323c/src/rtctype.c,
+ addons/ooh323c/src/ooCalls.c, addons/ooh323c/src/ooGkClient.c,
+ addons/ooh323c/src/h323, addons/ooh323c/src/ooUtils.c,
+ addons/ooh323c/src/ooDateTime.h,
+ addons/ooh323c/src/h323/MULTIMEDIA-SYSTEM-CONTROLEnc.c,
+ addons/ooh323c/src/rtctype.h, addons/ooh323c/src/ooCalls.h,
+ configs/mysql.conf.sample (added), addons/ooh323c/src/ooh323ep.c,
+ addons/ooh323c/src/ooGkClient.h,
+ addons/ooh323c/src/h323/H323-MESSAGES.c,
+ addons/ooh323c/src/ooUtils.h, addons/mp3/README, UPGRADE.txt,
+ addons/mp3/MPGLIB_TODO, addons/ooh323c/src/ooh323ep.h,
+ addons/ooh323c/src/h323/H323-MESSAGES.h,
+ addons/mp3/decode_ntom.c, configs/ooh323.conf.sample (added),
+ addons/ooh323c/src/ooh323.c,
+ addons/ooh323c/src/h323/H323-MESSAGESDec.c, addons/ooh323c/src,
+ build_tools/menuselect-deps.in, addons/mp3/tabinit.c,
+ addons/ooh323c/src/ooh323.h, doc/tex/Makefile,
+ addons/ooh323c/src/decode.c, addons/ooh323c/src/context.c,
+ main/file.c,
+ addons/ooh323c/src/h323/MULTIMEDIA-SYSTEM-CONTROLDec.c,
+ makeopts.in, addons/ooh323c/src/oochannels.c,
+ addons/app_saycountpl.c, addons/ooh323c/src/printHandler.c,
+ addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ootrace.c,
+ addons/res_config_mysql.c: Move Asterisk-addons modules into the
+ main Asterisk source tree. Someone asked yesterday, "is there a
+ good reason why we can't just put these modules in Asterisk?".
+ After a brief discussion, as long as the modules are clearly set
+ aside in their own directory and not enabled by default, it is
+ perfectly fine. For more information about why a module goes in
+ addons, see README-addons.txt. chan_ooh323 does not currently
+ compile as it is behind some trunk API updates. However, it will
+ not build by default, so it should be okay for now.
+
+2009-06-29 23:50 +0000 [r204355] Sean Bright <sean@malleable.com>
+
+ * apps/app_meetme.c: A few const changes in app_meetme.c that I
+ noticed while browsing the source.
+
+2009-06-29 22:50 +0000 [r204247-204301] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 204300 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun
+ 2009) | 9 lines Add error message so that it is clear why a SIP
+ peer was not processed when a DNS lookup fails on a host or
+ outboundproxy. (closes issue #13432) Reported by: p_lindheimer
+ Patches: outboundproxy.patch uploaded by p (license 558) ........
+
+ * /, channels/chan_sip.c: Merged revisions 204243,204246 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun
+ 2009) | 22 lines Fix a problem where chan_sip would ignore "old"
+ but valid responses. chan_sip has had a problem for quite a long
+ time that would manifest when Asterisk would send multiple SIP
+ responses on the same dialog before receiving a response. The
+ problem occurred because chan_sip only kept track of the highest
+ outgoing sequence number used on the dialog. If Asterisk sent two
+ requests out, and a response arrived for the first request sent,
+ then Asterisk would ignore the response. The result was that
+ Asterisk would continue retransmitting the requests and ignoring
+ the responses until the maximum number of retransmissions had
+ been reached. The fix here is to rearrange the code a bit so that
+ instead of simply comparing the sequence number of the response
+ to our latest outgoing sequence number, we walk our list of
+ outstanding packets and determine if there is a match. If there
+ is, we continue. If not, then we ignore the response. In doing
+ this, I found a few completely useless variables that I have now
+ removed. (closes issue #11231) Reported by: flefoll Review:
+ https://reviewboard.asterisk.org/r/298 ........ r204246 |
+ mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3
+ lines Fix build oops. ........
+
+2009-06-29 20:29 +0000 [r204119-204217] Sean Bright <sean@malleable.com>
+
+ * configs/cel_adaptive_odbc.conf.sample: Reorganize this adaptive
+ CEL config a bit.
+
+ * apps/app_rpt.c: Get app_rpt compiling again. I doubt seriously
+ that it actually works. Also, the code in this module is
+ horrendous and we should remove it from the tree. I'm not sure
+ who is supposed to be maintaning this thing, but they clearly are
+ not. I don't see the sense of leaving it in the main tree. If it
+ lives *anywhere* it should be in addons.
+
+ * configs/cel_sqlite3_custom.conf.sample, configs/cel.conf.sample,
+ configs/cel_adaptive_odbc.conf.sample,
+ configs/cel_pgsql.conf.sample, configs/cel_custom.conf.sample:
+ Add common headers to CEL related configs.
+
+2009-06-29 17:56 +0000 [r204069-204118] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, include/asterisk/channel.h: Allow trunk to once
+ again compile under MALLOC_DEBUG
+
+ * configs/cel_adaptive_odbc.conf.sample: Remove invalid entries in
+ the config. This might seem like a legitimate comment that merely
+ needed semicolon prefixes, but in reality, the adaptive layer is
+ designed to allow arbitrary CDR variables, without needing the
+ use of a userfield to store multiple items. It's therefore not
+ only invalid syntax but also goes against the intent of the
+ adaptive method.
+
+2009-06-27 20:26 +0000 [r203985] Sean Bright <sean@malleable.com>
+
+ * CHANGES: Another CHANGES spelling fix.
+
+2009-06-27 10:04 +0000 [r203960-203962] Russell Bryant <russell@digium.com>
+
+ * main/app.c: Only update total silence counter after a counter
+ reset. (closes issue #2264) Reported by: pfn Patches:
+ silent-vm-1.6.2-fix2.txt uploaded by pfn (license 810) Tested by:
+ pfn
+
+ * UPGRADE.txt, CHANGES: Minor tweaks and spelling fixes for CHANGES
+ and UPGRADE.txt.
+
+2009-06-27 01:07 +0000 [r203909] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 203908 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009)
+ | 16 lines The ISDN CPE side should not exclusively pick B
+ channels normally. Before this patch, Asterisk unconditionally
+ picked B channels exclusively on the CPE side and normally
+ allowed alternative B channels on the network side. Now Asterisk
+ does the opposite. Reasons for the CPE side to normally not pick
+ B channels exclusively: * For CPE point-to-multipoint mode (i.e.
+ phone side), the CPE side does not have enough information to
+ exclusively pick B channels. (There may be other devices on the
+ line.) * Q.931 gives preference to the network side picking B
+ channels. * Some telcos require the CPE side to not pick B
+ channels exclusively. (closes issue #14383) Reported by:
+ mbrancaleoni ........
+
+2009-06-26 22:11 +0000 [r203853] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 203848 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r203848 | jpeeler | 2009-06-26 17:09:19 -0500 (Fri, 26
+ Jun 2009) | 5 lines Make sure to recreate the dahdi pseudo
+ channel after dahdi restart (closes issue #14477) Reported by:
+ timking ........
+
+2009-06-26 22:08 +0000 [r203846] Sean Bright <sean@malleable.com>
+
+ * cdr/cdr_syslog.c (added), build_tools/menuselect-deps.in,
+ configure, configure.ac, configs/cdr_syslog.conf.sample (added),
+ CHANGES: Add a new module, cdr_syslog, which allows writing CDRs
+ to syslog. The original patch for this was written by Brett
+ Bryant, and I split it out into it's own module. (closes issue
+ #12876) Reported by: bbryant Patches:
+ 06162008_cdr_custom_syslog.diff uploaded by bbryant (license 36)
+ 05212009_cdr_syslog.patch uploaded by seanbright (license 71)
+ Tested by: seanbright Review:
+ https://reviewboard.asterisk.org/r/297/
+
+2009-06-26 21:48 +0000 [r203802-203842] Russell Bryant <russell@digium.com>
+
+ * CHANGES, apps/app_chanspy.c: Add 's' option to ChanSpy, which
+ makes the app exit when no channels are left to spy on. (closes
+ issue #14594) Reported by: JimDickenson Patches: chanspy.diff
+ uploaded by JimDickenson (license 710)
+
+ * /, main/file.c: Merged revisions 203785 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009)
+ | 15 lines Don't fast forward past the end of a message. This is
+ nice change for users of the voicemail application. If someone
+ gets a little carried away with fast forwarding through a
+ message, they can easily get to the end and accidentally exit the
+ voicemail application by hitting the fast forward key during the
+ following prompt. This adds some safety by not allowing a fast
+ forward past the end of a message. (closes issue #14554) Reported
+ by: lacoursj Patches: 21761.patch uploaded by lacoursj (license
+ 707) Tested by: lacoursj ........
+
+2009-06-26 20:52 +0000 [r203783] Mark Michelson <mmichelson@digium.com>
+
+ * doc/manager_1_1.txt, main/manager.c: Add timestamp to response to
+ "Ping" manager action. (closes issue #14596) Reported by:
+ JimDickenson Patches: pong2.diff uploaded by JimDickenson
+ (license 710)
+
+2009-06-26 20:45 +0000 [r203779] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Ensure the TCP read buffer is fully
+ initialized before handling each packet. (closes issue #14452)
+ Reported by: umberto71
+
+2009-06-26 20:19 +0000 [r203735] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Fix the
+ 'nat' option to actually do RFC3581 as expected and extend the
+ configurable values for finer control. (closes issue #8855)
+ Reported by: mikma Tested by: klaus3000, file
+
+2009-06-26 20:13 +0000 [r203721] David Brooks <dbrooks@digium.com>
+
+ * apps/app_voicemail.c: Fixing voicemail's error in checking max
+ silence vs min message length Max silence was represented in
+ milliseconds, yet vmminsecs (minmessage) was represented as
+ seconds. Also, the inequality was reversed. The warning, if
+ triggered, was "Max silence should be less than minmessage or you
+ may get empty messages", which should have been logged if max
+ silence was greater than minmessage, but the check was for less
+ than. Also, conforming if statement to coding guidelines. closes
+ issue #15331) Reported by: markd Review:
+ https://reviewboard.asterisk.org/r/293/
+
+2009-06-26 19:47 +0000 [r203710] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: moving debug message from level 0 to 1.
+ (closes issue #15404) Reported by: leobrown Patches:
+ iax_codec_debug.patch uploaded by leobrown (license 541)
+
+2009-06-26 19:31 +0000 [r203702] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/devicestate.h, main/pbx.c, main/devicestate.c:
+ Make invalid hints report Unavailable instead of Idle. (closes
+ issue #14413) Reported by: pj
+
+2009-06-26 19:27 +0000 [r203699] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c, main/frame.c, main/rtp_engine.c,
+ channels/chan_sip.c, apps/app_fax.c, configs/sip.conf.sample,
+ include/asterisk/frame.h: Improve T.38 negotiation by exchanging
+ session parameters between application and channel.
+
+2009-06-26 19:03 +0000 [r203672] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/sig_analog.c: Check if polarityonanswerdelay has elapsed
+ before setting a channel as answered after a polarity reversal.
+ Previously on a polarity switch event chan_dahdi would set the
+ channel immediately as answered. This would cause problems if a
+ polarity reversal occurred when the line was picked up as the
+ dial would not have yet occurred. Now if the polarity reversal
+ occurs before delay has elapsed after coming off hook or an
+ answer, it is ignored. Also, some refactoring was done in
+ _handle_event. (closes issue #13917) Reported by: alecdavis
+ Patches: chan_dahdi.bug13917.feb09.diff2.txt uploaded by
+ alecdavis (license 585) Tested by: alecdavis
+
+2009-06-26 15:42 +0000 [r203638-203640] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/doxyref.h, include/asterisk/channel.h: Note a
+ new API call, and one that changed in doxygen.
+
+ * cel/cel_pgsql.c, configs/cel_sqlite3_custom.conf.sample (added),
+ cdr/cdr_sqlite3_custom.c, configs/cel.conf.sample (added),
+ channels/chan_local.c, include/asterisk/cel.h (added),
+ main/devicestate.c, apps/app_chanisavail.c, channels/chan_iax2.c,
+ doc/tex/cel-doc.tex (added), main/loader.c, main/cli.c,
+ channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/chan_skinny.c, include/asterisk/event_defs.h,
+ main/features.c, res/ais/evt.c, channels/sig_analog.h,
+ channels/chan_alsa.c, doc/tex/asterisk.tex, cdr/cdr_manager.c,
+ apps/app_dial.c, main/pbx.c, include/asterisk/utils.h,
+ channels/chan_bridge.c, cel/cel_tds.c, channels/chan_agent.c,
+ configs/cel_adaptive_odbc.conf.sample (added),
+ include/asterisk/cdr.h, include/asterisk/channel.h, CHANGES,
+ main/cel.c (added), Makefile, channels/chan_misdn.c,
+ funcs/func_channel.c, funcs/func_cdr.c, doc/tex/celdriver.tex
+ (added), main/asterisk.c, cel/cel_adaptive_odbc.c,
+ apps/app_voicemail.c, res/res_calendar.c,
+ channels/chan_unistim.c, tests/test_substitution.c,
+ cel/cel_radius.c, channels/chan_multicast_rtp.c,
+ channels/chan_vpb.cc, apps/app_meetme.c, channels/chan_gtalk.c,
+ apps/app_followme.c, configs/cel_tds.conf.sample (added),
+ main/channel.c, main/cdr.c, channels/chan_phone.c, main/dial.c,
+ main/manager.c, include/asterisk/event.h,
+ bridges/bridge_builtin_features.c, funcs/func_odbc.c,
+ cel/cel_custom.c, cel/cel_manager.c, cdr/cdr_sqlite.c,
+ res/res_agi.c, apps/app_minivm.c, main/logger.c,
+ apps/app_confbridge.c, configs/cel_custom.conf.sample (added),
+ channels/chan_mgcp.c, apps/app_parkandannounce.c,
+ cdr/cdr_custom.c, channels/chan_sip.c, cel (added),
+ configs/cel_pgsql.conf.sample (added), channels/chan_console.c,
+ include/asterisk/_private.h, channels/sig_pri.c,
+ apps/app_queue.c, channels/chan_oss.c, channels/sig_pri.h,
+ channels/chan_usbradio.c, channels/chan_jingle.c, cel/Makefile,
+ apps/app_celgenuserevent.c (added), apps/app_directed_pickup.c,
+ channels/chan_h323.c, cel/cel_sqlite3_custom.c, main/event.c,
+ channels/chan_nbs.c: Merge the new Channel Event Logging (CEL)
+ subsystem. CEL is the new system for logging channel events. This
+ was inspired after facing many problems trying to represent what
+ is possible to happen to a call in Asterisk using CDR records.
+ For more information on CEL, see the built in HTML or PDF
+ documentation generated from the files in doc/tex/. Many thanks
+ to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
+ work developing this code. Also, thanks to Matt Nicholson
+ (mnicholson) and Sean Bright (seanbright) for their assistance in
+ the final push to get this code ready for Asterisk trunk. Review:
+ https://reviewboard.asterisk.org/r/239/
+
+2009-06-26 13:00 +0000 [r203569-203605] Sean Bright <sean@malleable.com>
+
+ * include/asterisk/syslog.h, main/syslog.c: Add functions to map
+ syslog facilities and priorities constants to strings. Also
+ change the default casing of the string contants to lowercase.
+ This really just saves us from have to lowercase them later when
+ displaying them.
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ main/syslog.c: Add checks in configure for non-POSIX syslog
+ facilities.
+
+2009-06-26 00:23 +0000 [r203525-203534] Russell Bryant <russell@digium.com>
+
+ * main/syslog.c: One more formatting nit ... use spaces for inline
+ indentation.
+
+ * main/syslog.c: Convert spaces to tabs for indentation.
+
+2009-06-25 23:54 +0000 [r203508] Sean Bright <sean@malleable.com>
+
+ * include/asterisk/syslog.h (added), main/logger.c, main/syslog.c
+ (added): Move syslog utility functions into a separate file so
+ they can be re-used. This has the pleasant side effect of
+ cleaning up the header inclusion process in logger.c.
+
+2009-06-25 22:48 +0000 [r203479] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: make sure chan_dahdi compiles with only
+ libss7 and not libpri installed
+
+2009-06-25 21:45 +0000 [r203444] David Vossel <dvossel@digium.com>
+
+ * main/ast_expr2.fl, main/ast_expr2.c: fixes a few redundant
+ conditions (issue #15269)
+
+2009-06-25 21:34 +0000 [r203443] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Picking nits
+
+2009-06-25 21:22 +0000 [r203402] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Remove
+ some unnecessary code and update sample config file with respect
+ to GR-303.
+
+2009-06-25 21:15 +0000 [r203381] Terry Wilson <twilson@digium.com>
+
+ * /, main/cli.c: Merged revisions 203380 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203380 | twilson | 2009-06-25 16:13:10 -0500 (Thu, 25 Jun 2009)
+ | 4 lines I didn't see that Mark already fixed the underlying
+ issue! Yay for removing useless code. ........
+
+2009-06-25 21:04 +0000 [r203376] Russell Bryant <russell@digium.com>
+
+ * /, main/features.c: Merged revisions 203375 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009)
+ | 9 lines Fix a case where CDR answer time could be before the
+ start time involving parking. (closes issue #13794) Reported by:
+ davidw Patches: 13794.patch uploaded by murf (license 17)
+ 13794.patch.160 uploaded by murf (license 17) Tested by: murf,
+ dbrooks ........
+
+2009-06-25 20:25 +0000 [r203338] Terry Wilson <twilson@digium.com>
+
+ * /, main/cli.c: Merged revisions 203311 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203311 | twilson | 2009-06-25 15:09:15 -0500 (Thu, 25 Jun 2009)
+ | 2 lines Don't try to free NULL ........
+
+2009-06-25 19:54 +0000 [r203304] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/sig_pri.h (added), channels/chan_dahdi.c,
+ channels/sig_analog.c, channels/sig_analog.h, channels/sig_pri.c
+ (added), channels/Makefile: New signaling module to handle
+ PRI/BRI operations in chan_dahdi This merge splits the PRI/BRI
+ signaling logic out of chan_dahdi.c into sig_pri.c. Functionality
+ in theory should not change (mostly). A few trivial changes were
+ made in sig_analog with verbose messages and commenting.
+
+2009-06-25 19:22 +0000 [r203258] Jason Parker <jparker@digium.com>
+
+ * channels/chan_dahdi.c: Unmute when we get a dtmfup (we muted on
+ dtmfdown) event. This would occasionally cause one-way audio when
+ using hardware DTMF detection. (closes issue #14761) Reported by:
+ tzafrir Patches: v1-14761.patch uploaded by dimas (license 88)
+ Tested by: tzafrir, dimas
+
+2009-06-25 18:25 +0000 [r203227] Joshua Colp <jcolp@digium.com>
+
+ * res/res_rtp_multicast.c (added), channels/chan_multicast_rtp.c
+ (added), CHANGES: Add support for multicast RTP paging. (closes
+ issue #11797) Reported by: macbrody Review:
+ https://reviewboard.asterisk.org/r/270/
+
+2009-06-25 17:01 +0000 [r203188] Sean Bright <sean@malleable.com>
+
+ * main/logger.c: Pass a logmsg to ast_log_vsyslog instead of
+ separate arguments.
+
+2009-06-25 16:18 +0000 [r203126] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_dahdi.c: Insure ring cadence is set for fxs ports
+ Moved SETCADENCE ioctl call to before call into new analog signal
+ module to insure that it gets set. (closes issue #15381) Reported
+ by: alecdavis Patches: fix15381.diff uploaded by dbailey (license
+ 819) Tested by: dbailey
+
+2009-06-25 16:04 +0000 [r203116] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 203115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009)
+ | 11 lines Resolve a crash related to a T.38 reinvite race
+ condition. This change resolves a crash observed locally during
+ some T.38 testing. A call was set up using a call file, and when
+ the T.38 reinvite came in, the channel state was still
+ AST_STATE_DOWN. The reason is explained by a comment in the code
+ that previously lived in the handling of AST_STATE_RINGING. This
+ change modifies the logic to handle the same race condition for
+ any channel state that is not UP. (closes ABE-1895) ........
+
+2009-06-24 21:08 +0000 [r203037] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 203036 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24
+ Jun 2009) | 8 lines Improved chan_dahdi.conf pritimer error
+ checking. Valid format is: pritimer=timer_name,timer_value *
+ Fixed segfault if the ',' is missing. * Completely check the
+ range returned by pri_timer2idx() to prevent possible access
+ outside array bounds. ........
+
+2009-06-24 18:29 +0000 [r202967] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 202966 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202966 | mmichelson | 2009-06-24 13:28:47 -0500 (Wed, 24 Jun
+ 2009) | 3 lines Use the handy UNLINK macro instead of hand-coding
+ the same thing in-line. ........
+
+2009-06-24 18:08 +0000 [r202925] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Ensure the default settings are applied for
+ T.38 when we set it up for a peer.
+
+2009-06-24 13:53 +0000 [r202840-202889] Sean Bright <sean@malleable.com>
+
+ * doc/tex: Ignore some files generated when asterisk.pdf is
+ created.
+
+ * configs/cdr_tds.conf.sample, cdr/cdr_tds.c: Update sample cdr_tds
+ configuration to try and eliminate some confusion. Also change
+ the preferred configuration option from 'hostname' (which was
+ misleading because it didn't actually treat the value as a
+ hostname) to 'connection' and added some verbage explaining that
+ the user would need to refer to their freetds.conf file for those
+ settings. 'hostname' was kept as a backwards compatible
+ configuration parameter.
+
+ * doc/tex/billing.tex, doc/tex/cdrdriver.tex: Change some section
+ names in the CDR tex documentation.
+
+ * doc/tex/cdrdriver.tex: Remove some trailing whitespace before
+ making content changes.
+
+2009-06-23 22:47 +0000 [r202804] Russell Bryant <russell@digium.com>
+
+ * doc/tex/cdrdriver.tex: Clean up section hierarchy for the CDR
+ chapter.
+
+2009-06-23 22:08 +0000 [r202761] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_dahdi.c: I could have sworn I committed this patch
+ ages ago, but... bug fix with setting NAI properly on linksets in
+ certain situations.
+
+2009-06-23 21:38 +0000 [r202755] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: Make outgoing_colp=2 misdn.conf port
+ parameter not send redirecting or transfer messages. If the
+ outgoing_colp parameter is set to not send COLP information, then
+ it does not make sense to send redirecting or transfer messages
+ announcing new COLP information that is blocked. The service
+ provider may supply the listed number for that line when it
+ passes the messages to the next hop. Why tell the switch that
+ these events happened when the information is otherwise
+ suppressed? Also blocked the number of previous redirects that
+ may have occurred to calls going out the port when outgoing_colp
+ is 2. Follow on to JIRA ABE-1853.
+
+2009-06-23 21:25 +0000 [r202753] Ryan Brindley <rbrindley@digium.com>
+
+ * main/config.c: If we delete the info, lets also delete the lines
+ (closes issue #14509) Reported by: timeshell Patches:
+ 20090504__bug14509.diff.txt uploaded by tilghman (license 14)
+ Tested by: awk, timeshell
+
+2009-06-23 16:31 +0000 [r202672] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 202671 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202671 | dvossel | 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009)
+ | 12 lines MWI NOTIFY contains a wrong URI if Asterisk listens to
+ non-standard port and transport (closes issue #14659) Reported
+ by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt uploaded
+ by klaus3000 (license 65) mwi_port-transport_trunk.diff uploaded
+ by dvossel (license 671) Tested by: dvossel, klaus3000 Review:
+ https://reviewboard.asterisk.org/r/288/ ........
+
+2009-06-23 14:54 +0000 [r202497-202570] Russell Bryant <russell@digium.com>
+
+ * main/app.c, CHANGES: Ignore voicemail messages that are just
+ silence. (closes issue #2264) Reported by: pfn Patches:
+ silent-vm-1.6.2.txt uploaded by pfn (license 810)
+
+ * main/channel.c, /: Merged revisions 202496 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009)
+ | 4 lines Report CallerID change during a masquerade. Reported
+ by: markster ........
+
+2009-06-22 16:09 +0000 [r202417] Sean Bright <sean@malleable.com>
+
+ * cdr/cdr_sqlite3_custom.c: Fix lock usage in cdr_sqlite3_custom to
+ avoid potential crashes during reload. Pointed out by Russell
+ while working on the CEL branch.
+
+2009-06-22 16:05 +0000 [r202415] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 202414 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202414 | russell | 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009)
+ | 2 lines Make Polycom subscription type override check more
+ explicit. ........
+
+2009-06-22 15:33 +0000 [r202410] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/module.h, main/loader.c: attempting to load
+ running modules Modules placed in the priority heap for loading
+ were not properly removed from the linked list. This resulted in
+ some modules attempting to load twice.
+
+2009-06-22 14:58 +0000 [r202337-202343] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 202341-202342 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun
+ 2009) | 26 lines Fix a situation in which Asterisk would not stop
+ retransmitting 487s. If a CANCEL were received by Asterisk, we
+ would send a 487 in response to the original INVITE and a 200 OK
+ for the CANCEL. If there were a network hiccup which caused the
+ 200 OK and the 487 to be lost, then the UA communicating with
+ Asterisk may try to retransmit its CANCEL. Asterisk's response to
+ this used to be to try sending another 487 to the canceled INVITE
+ and another 200 OK to the CANCEL. The problem here is that the
+ originally-sent 487 was sent "reliably" meaning that it will be
+ retransmitted until it is received properly. So when we receive
+ the second CANCEL it is likely that the first batch of 487s we
+ sent is still going strong and reaches the UA. The result was
+ that the second set of 487s would be retransmitted constantly
+ until the maximum number of retries had been reached. The fix for
+ this is that if we receive a second CANCEL for an INVITE, then we
+ cancel the retransmission of the first set of 487s and start a
+ second set. This causes the dialog to be terminated reasonably.
+ (closes issue #14584) Reported by: klaus3000 Patches:
+ 14584_v2.patch uploaded by mmichelson (license 60) Tested by:
+ klaus3000 ........ r202342 | mmichelson | 2009-06-22 09:44:58
+ -0500 (Mon, 22 Jun 2009) | 3 lines Remove an extra debug line
+ left from previous commit. ........
+
+ * /, channels/chan_sip.c: Merged revisions 202336 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun
+ 2009) | 25 lines Fix a possible infinite loop in SDP parsing
+ during glare situation. There was a while loop in
+ get_ip_and_port_from_sdp which was controlled by a call to
+ get_sdp_iterate. The loop would exit either if what we were
+ searching for was found or if the return was NULL. The problem is
+ that get_sdp_iterate never returns NULL. This means that if what
+ we were searching for was not present, the loop would run
+ infinitely. This modification of the loop fixes the problem.
+ (closes issue #15213) Reported by: schmidts (closes issue #15349)
+ Reported by: samy (closes issue #14464) Reported by: pj (closes
+ issue #15345) Reported by: aragon Patches: sip_inf_loop.patch
+ uploaded by mmichelson (license 60) Tested by: aragon ........
+
+2009-06-21 16:36 +0000 [r202223-202301] Russell Bryant <russell@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c: Note a bug in cdr_sqlite3_custom so I
+ don't forget about it.
+
+ * cdr/cdr_manager.c: Fix possibility of crashiness during reload in
+ custom fields handling.
+
+ * cdr/cdr_manager.c: Standardize return values of load_config() so
+ reload() doesn't report an error on success.
+
+ * cdr/cdr_manager.c: Leave a note about some unsafe code in
+ cdr_manager
+
+2009-06-20 19:09 +0000 [r202183] Sean Bright <sean@malleable.com>
+
+ * apps/app_fax.c: Fix version detection for API changes in spandsp.
+ (closes issue #15355) Reported by: deuffy
+
+2009-06-20 14:09 +0000 [r202109] Russell Bryant <russell@digium.com>
+
+ * main/cdr.c, cdr/cdr_adaptive_odbc.c, cdr/cdr_pgsql.c: Remove
+ unnecessary usleep() from a couple of module unload callbacks. In
+ passing, also tweak cdr_unregister() to hold the list lock a bit
+ less time.
+
+2009-06-19 21:25 +0000 [r202039] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Use sched_yield() instead of usleep(1)
+
+2009-06-19 20:24 +0000 [r201994] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 201993 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r201993 | dvossel | 2009-06-19 15:22:02 -0500 (Fri, 19
+ Jun 2009) | 8 lines timestamp was being converted to host order
+ as a short rather than a long (closes issue #15361) Reported by:
+ ffloimair Patches: ts_issue.diff uploaded by dvossel (license
+ 671) ........
+
+2009-06-19 17:40 +0000 [r201944] Terry Wilson <twilson@digium.com>
+
+ * CHANGES: Add note about the addition of calendar support
+
+2009-06-19 15:47 +0000 [r201904] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_odbc.c: Fix 2 typos and add support for wide
+ character types. Reported by Benny Amorsen via the asterisk-users
+ mailing list.
+ http://lists.digium.com/pipermail/asterisk-users/2009-June/233622.html
+
+2009-06-19 15:41 +0000 [r201902] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp_engine.c, channels/chan_sip.c,
+ include/asterisk/rtp_engine.h: Add support for allowing an RTP
+ engine to decide on whether it is possible for specific formats
+ to be transcoded for an RTP instance.
+
+2009-06-19 00:43 +0000 [r201745-201829] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/features.c: Merged revisions 201828 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009)
+ | 6 lines If the "h" extension fails, give it another chance in
+ main/pbx.c. If the "h" extension fails, give it another chance in
+ main/pbx.c, when it returns from the bridge code. Fixes an issue
+ where the "h" extension may occasionally not fire, when a Dial is
+ executed from a Macro. Debugged in #asterisk with user tompaw.
+ ........
+
+ * apps/Makefile: One of the changes in 1.6.1 was to allow
+ app_directory to use functionality within app_voicemail for
+ directory functions. It is therefore no longer necessary for
+ app_directory to be linked against the ODBC libraries (and it
+ never was necessary for app_directory to be linked against IMAP,
+ though it was).
+
+ * funcs/func_cut.c: Clarify CUT code, and in the process, fix a bug
+ in trunk only (closes issue #15320) Reported by: chappell
+ Patches: cut_fix.patch uploaded by chappell (license 8)
+ cut_clarify.patch uploaded by chappell (license 8)
+
+2009-06-18 17:41 +0000 [r201717] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Added deadlock protection to
+ try_suggested_sip_codec in chan_sip.c. Review:
+ https://reviewboard.asterisk.org/r/285/
+
+2009-06-18 16:37 +0000 [r201678] David Vossel <dvossel@digium.com>
+
+ * codecs/gsm/src/gsm_destroy.c, channels/h323/ast_h323.cxx,
+ main/ast_expr2f.c, res/ael/ael_lex.c, utils/ael_main.c,
+ utils/extconf.c, channels/xpmr/xpmr.c, pbx/pbx_config.c,
+ res/res_config_ldap.c, apps/app_rpt.c, channels/misdn/isdn_lib.c,
+ main/asterisk.c, utils/conf2ael.c, main/ast_expr2.c,
+ utils/stereorize.c: fixes some memory leaks and redundant
+ conditions (closes issue #15269) Reported by: contactmayankjain
+ Patches: patch.txt uploaded by contactmayankjain (license 740)
+ memory_leak_stuff.trunk.diff uploaded by dvossel (license 671)
+ Tested by: contactmayankjain, dvossel
+
+2009-06-18 15:27 +0000 [r201610] Russell Bryant <russell@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 201600 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18
+ Jun 2009) | 29 lines Fix memory corruption and leakage related
+ reloads of non files mode MoH classes. For Music on Hold classes
+ that are not files mode, meaning that we are executing an
+ application that will feed us audio data, we use a thread to
+ monitor the external application and read audio from it. This
+ thread also makes use of the MoH class object. In the MoH class
+ destructor, we used pthread_cancel() to ask the thread to exit.
+ Unfortunately, the code did not wait to ensure that the thread
+ actually went away. What needed to be done is a pthread_join() to
+ ensure that the thread fully cleans up before we proceed. By
+ adding this one line, we resolve two significant problems: 1)
+ Since the thread was never joined, it never fully goes away. So,
+ on every reload of non-files mode MoH, an unused thread was
+ sticking around. 2) There was a race condition here where the
+ application monitoring thread could still try to access the MoH
+ class, even though the thread executing the MoH reload has
+ already destroyed it. (issue #15109) Reported by: jvandal (issue
+ #15123) Reported by: axisinternet (issue #15195) Reported by:
+ amorsen (issue AST-208) ........
+
+2009-06-18 15:20 +0000 [r201583] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_rtp_asterisk.c, main/rtp_engine.c, channels/chan_sip.c,
+ include/asterisk/rtp_engine.h: Trunk implementation of setting an
+ alternate RTP source. This contains the interface by which we can
+ let an rtp instance know that it might start receiving audio from
+ a new source. This is similar in nature to revision 197588 of
+ Asterisk 1.4. Review: https://reviewboard.asterisk.org/r/276
+
+2009-06-18 15:16 +0000 [r201534-201570] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: parsing extension correctly from sip
+ register lines If a transport type was specified, but no
+ extension, parsing of the extension would return whatever was
+ after the transport rather than defaulting to 's'. (closes issue
+ #15111) Reported by: ffs Patches:
+ chan_sip.c_register-parser.patch uploaded by ffs (license 730)
+ Tested by: ffs, dvossel
+
+ * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Add
+ rtsavesysname to chan_iax chan_sip has an option to save the
+ sysname on rtupdate. This patch copies that same logic to
+ chan_iax. (closes issue #14837) Reported by: barthpbx Patches:
+ iax2-rtsavesysname.patch uploaded by barthpbx (license 744)
+ rt_iax.diff uploaded by dvossel (license 671)
+
+2009-06-17 21:31 +0000 [r201531] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Initialize additional variables, to prevent
+ a possible crash. (closes issue #15186) Reported by: ajohnson
+ Patches: 20090528__issue15186.diff.txt uploaded by tilghman
+ (license 14) Tested by: ajohnson
+
+2009-06-17 20:10 +0000 [r201458-201462] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix problem with no audio due to ignoring
+ the SDP. A recent change to our SDP version comparison made audio
+ not function on some calls. This was because of a test wherein we
+ were trying to see if an unsigned value was less than 0. This is
+ a dumb comparison and arguably the compiler should have warned
+ about it. Alas, though, it slipped past. Now it's fixed by
+ changing the variable to be a signed type. Found by several
+ developers. Tested by mnicholson and dbrooks.
+
+ * main/channel.c, /: Merged revisions 201450 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun
+ 2009) | 9 lines Change the datastore traversal in
+ ast_do_masquerade to use a safe list traversal. It is possible
+ for datastore fixup functions to remove the datastore from the
+ list and free it. In particular, the queue_transfer_fixup in
+ app_queue does this. While I don't yet know of this causing any
+ crashes, it certainly could. Found while discussing a separate
+ issue with Brian Degenhardt. ........
+
+2009-06-17 20:00 +0000 [r201445-201453] David Vossel <dvossel@digium.com>
+
+ * doc/datastores.txt: ast_channel_datastore_alloc is no longer
+ used. updating datastores.txt to reflect that.
+
+ * /, apps/app_mixmonitor.c: Merged revisions 201423 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17
+ Jun 2009) | 19 lines StopMixMonitor race condition (not giving up
+ file immediately) StopMixMonitor only indicates to the MixMonitor
+ thread to stop writing to the file. It does not guarantee that
+ the recording's file handle is available to the dialplan
+ immediately after execution. This results in a race condition. To
+ resolve this, the filestream pointer is placed in a datastore on
+ the channel. When StopMixMonitor is called, the datastore is
+ retrieved from the channel and the filestream is closed
+ immediately before returning to the dialplan. Documentation
+ indicating the use of StopMixMonitor to free files has been
+ updated as well. (closes issue #15259) Reported by: travisghansen
+ Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/283/ ........
+
+2009-06-17 19:15 +0000 [r201381] David Brooks <dbrooks@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 201380 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009)
+ | 9 lines Checks for NULL sip_pvt pointer in
+ chan_sip.c->acf_channel_read() Zombie channels could be passed,
+ and chan_sip.c wasn't checking for it. Could crash Asterisk. Now
+ checking for NULL pointer. (closes issue #15330) Reported by:
+ okrief Tested by: dbrooks ........
+
+2009-06-17 15:20 +0000 [r201331-201344] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: SIP registry ref count error During a sip
+ reload, the list of sip_registry objects are supposed to be
+ traversed, unlinked, and destroyed, but destruction never takes
+ place due to a ref counting error. This causes a memory leak when
+ registry items are removed from sip.conf and reloaded. While the
+ registries are removed from the global list, they are not removed
+ from the scheduler. Because of this, SIP register attempts
+ continue to be sent out for the item even though it may no longer
+ be in the .conf. (closes issue #15295) Reported by: amorsen
+ Review: https://reviewboard.asterisk.org/r/282/
+
+ * channels/chan_iax2.c: update chan_iax to use 64bit feature flags.
+ (closes issue #15335) Reported by: lmadsen Review:
+ https://reviewboard.asterisk.org/r/284/
+
+2009-06-17 12:04 +0000 [r201262] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, include/asterisk/linkedlists.h: Merged revisions 201261 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun
+ 2009) | 9 lines Correct AST_LIST_APPEND_LIST behavior when list
+ to be appended is empty. When the list to be appended is empty,
+ and the list to be appended to is *not*, AST_LIST_APPEND_LIST
+ would actually cause the target list to become broken, and no
+ longer have a pointer to its last entry. This patch fixes the
+ problem. (reported by Stanislaw Pitucha on the asterisk-dev
+ mailing list) ........
+
+2009-06-16 22:29 +0000 [r201223] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fix issue with build_contact introduced by
+ the "SIP trasnport type issues" commit
+
+2009-06-16 22:11 +0000 [r201190] Sean Bright <sean@malleable.com>
+
+ * CREDITS: Update my e-mail address (thanks for the props, russell
+ :))
+
+2009-06-16 21:10 +0000 [r200985-201139] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c, channels/chan_sip.c, apps/app_fax.c,
+ include/asterisk/frame.h: Enable applications to enable/disable
+ digit and tone detection. Some applications (notably app_fax) do
+ not need digit detection nor FAX tone detection while they are
+ running, and if Asterisk is using software DSPs to provide the
+ detection, this consumes extra CPU cycles that could be better
+ spent on the actual application. This patch allows applications
+ to query and control the state of digit and tone detection on a
+ channel, and modifies app_fax to disable them while the FAX
+ operations are occurring (and re-enable digit detection
+ afterwards).
+
+ * configure, configure.ac: Explicitly test for 'static weakref'
+ support. Since we use 'static' weakref symbols, and not all GCC
+ versions support them, test for that combination explicitly.
+
+ * Makefile: When compiling in an Emacs-spawned shell, always print
+ directory names. This change ensures that Emacs can find the
+ proper source files when parsing compiler error messages, since
+ it uses the 'make' output including directory names to do it.
+
+ * configure, autoconf/ast_gcc_attribute.m4, configure.ac: Another
+ minor fix to compiler attribute checking. Defaulting to 'static'
+ for the function scope was bad... so remove it.
+
+ * main/channel.c, main/autoservice.c, main/frame.c, /,
+ apps/app_meetme.c, main/slinfactory.c,
+ include/asterisk/linkedlists.h, main/file.c,
+ include/asterisk/channel.h, include/asterisk/frame.h,
+ apps/app_chanspy.c, apps/app_mixmonitor.c: Merged revisions
+ 200991 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun
+ 2009) | 11 lines Improve support for media paths that can
+ generate multiple frames at once. There are various media paths
+ in Asterisk (codec translators and UDPTL, primarily) that can
+ generate more than one frame to be generated when the application
+ calling them expects only a single frame. This patch addresses a
+ number of those cases, at least the primary ones to solve the
+ known problems. In addition it removes the broken TRACE_FRAMES
+ support, fixes a number of bugs in various frame-related API
+ functions, and cleans up various code paths affected by these
+ changes. https://reviewboard.asterisk.org/r/175/ ........
+
+ * configure, autoconf/ast_gcc_attribute.m4, configure.ac: Fix
+ problems with new compiler attribute checking in configure
+ script. The last changes to ast_gcc_attribute.m4 caused some
+ problems checking for various attributes, because the scope of
+ the symbol the attribute is applied to can be important; this
+ patch allows the scope to be specified for the check.
+
+2009-06-16 16:03 +0000 [r200946] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: SIP transport type issues What this patch
+ addresses: 1. ast_sip_ouraddrfor() by default binds to the UDP
+ address/port reguardless if the sip->pvt is of type UDP or not.
+ Now when no remapping is required, ast_sip_ouraddrfor() checks
+ the sip_pvt's transport type, attempting to set the address and
+ port to the correct TCP/TLS bindings if necessary. 2. It is not
+ necessary to send the port number in the Contact header unless
+ the port is non-standard for the transport type. This patch fixes
+ this and removes the todo note. 3. In sip_alloc(), the default
+ dialog built always uses transport type UDP. Now sip_alloc()
+ looks at the sip_request (if present) and determines what
+ transport type to use by default. 4. When changing the transport
+ type of a sip_socket, the file descriptor must be set to -1 and
+ in some cases the tcptls_session's ref count must be decremented
+ and set to NULL. I've encountered several issues associated with
+ this process and have created a function, set_socket_transport(),
+ to handle the setting of the socket type. (closes issue #13865)
+ Reported by: st Patches: dont_add_port_if_tls.patch uploaded by
+ Kristijan (license 753) 13865.patch uploaded by mmichelson
+ (license 60) tls_port_v5.patch uploaded by vrban (license 756)
+ transport_issues.diff uploaded by dvossel (license 671) Tested
+ by: mmichelson, Kristijan, vrban, jmacz, dvossel Review:
+ https://reviewboard.asterisk.org/r/278/
+
+2009-06-16 15:51 +0000 [r200943] Michiel van Baak <michiel@vanbaak.info>
+
+ * apps/app_voicemail.c: add FILE_STORAGE to Voicemail Build Options
+ Voicemail can only use one storage module at the moment. Because
+ it's unclear that selecting one of the storage modules in
+ menuselect will disable filesystem storage we now have a
+ FILE_STORAGE option that conflicts with the other modules.
+ (closes issue #15333)
+
+2009-06-16 15:26 +0000 [r200942] Russell Bryant <russell@digium.com>
+
+ * CREDITS: Add Sean Bright to CREDITS - Thanks, Sean!
+
+2009-06-16 14:12 +0000 [r200841-200878] Eliel C. Sardanons <eliels@gmail.com>
+
+ * /: Recorded merge of revisions 200875 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r200875 | eliel | 2009-06-16 09:25:51 -0400 (Tue, 16 Jun 2009) |
+ 5 lines Show the interface name on error, if it is not found. If
+ the smdiport specified is not found, show the interface name
+ instead of '(null)'. ........
+
+ * res/res_smdi.c: Show the interface name on error, if it is not
+ found. If the smdiport specified is not found, show the interface
+ name instead of '(null)'.
+
+2009-06-16 02:32 +0000 [r200805] Russell Bryant <russell@digium.com>
+
+ * main/manager.c: Don't claim a char * is a mansession object.
+ Since there was only 1 bucket, and no hash function was
+ specified, the code actually worked perfectly fine. However, in
+ theory, this was invalid use of the OBJ_POINTER flag, so remove
+ it so the code provides a better usage example.
+
+2009-06-16 02:24 +0000 [r200799] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: keep
+ backwards compatible chan_dahdi with older openr2 versions by not
+ using the new skip category feature unless supported
+
+2009-06-16 01:28 +0000 [r200764] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configure, autoconf/ast_gcc_attribute.m4: Ensure that
+ configure-script testing for compiler attributes actually works.
+ The configure script tests for compiler attributes didn't
+ actually enable enough warnings or provide a proper test harness
+ to determine whether the compiler supports the attribute in
+ question or not; this caused gcc 4.1 to report that it supports
+ 'weakref', but it doesn't actually support it in the way that is
+ needed for our optional API mechanism. The new configure script
+ test will properly distinguish between full support and partial
+ support for this attribute, among others.
+
+2009-06-16 01:26 +0000 [r200762] Russell Bryant <russell@digium.com>
+
+ * doc/tex/channelvariables.tex: Add missing closure of verbatim
+ environment.
+
+2009-06-16 01:03 +0000 [r200519-200726] Kevin P. Fleming <kpfleming@digium.com>
+
+ * CHANGES: Document the new automatic 'ignoresdpversion' behavior.
+ Asterisk will now automatically ignore incorrect incoming SDP
+ version numbers when necessary to complete a T.38 re-INVITE
+ operation.
+
+ * channels/chan_sip.c: Accept T.38 re-INVITE responses with invalid
+ SDP versions. This commit changes the 'incoming SDP version'
+ check logic a bit more; when 'ignoresdpversion' is *not* set for
+ a peer, if we initiate a re-INVITE to switch to T.38, we'll
+ always accept the peer's SDP response, even if they don't
+ properly increment the SDP version number as they should. If this
+ situation occurs, a warning message will be generated suggesting
+ that the peer's configuration be changed to include the
+ 'ignoresdpversion' configuration option (although ideally they'd
+ fix their SIP implementation to be RFC compliant). AST-221
+
+ * doc/CODING-GUIDELINES, apps/app_read.c, apps/app_page.c,
+ apps/app_fax.c, apps/app_readexten.c, apps/app_queue.c,
+ include/asterisk/app.h, apps/app_skel.c, apps/app_minivm.c,
+ apps/app_macro.c, apps/app_url.c, apps/app_sms.c,
+ apps/app_externalivr.c, apps/app_stack.c, apps/app_mixmonitor.c,
+ apps/app_voicemail.c: Last batch of 'static' qualifiers for
+ module-level global variables. Fix up modules in the 'apps'
+ directory, and also correct the bad example of enum definitions
+ in include/asterisk/app.h, which many developers followed (thanks
+ for reading the documentation!). In addition, add some basic
+ usage examples of the 'pahole' and 'pglobal' tools to the coding
+ guidelines.
+
+ * res/res_snmp.c, main/devicestate.c, funcs/func_vmcount.c,
+ res/res_calendar_caldav.c, formats/format_wav_gsm.c,
+ res/res_jabber.c, main/loader.c, main/cli.c, funcs/func_enum.c,
+ main/manager.c, res/res_smdi.c, funcs/func_odbc.c,
+ main/features.c, main/logger.c, main/http.c, pbx/pbx_realtime.c,
+ main/image.c, main/db.c, cdr/cdr_manager.c,
+ res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
+ res/res_config_pgsql.c, funcs/func_lock.c, pbx/pbx_lua.c,
+ funcs/func_cut.c, include/asterisk/calendar.h,
+ funcs/func_realtime.c, funcs/func_curl.c, funcs/func_cdr.c,
+ funcs/func_channel.c, main/file.c, main/event.c, pbx/pbx_dundi.c,
+ main/xmldoc.c, res/res_calendar.c: More 'static' qualifiers on
+ module global variables. The 'pglobal' tool is quite handy indeed
+ :-)
+
+ * channels/chan_dahdi.c, channels/chan_misdn.c,
+ channels/chan_sip.c, channels/chan_skinny.c,
+ channels/chan_agent.c, channels/chan_h323.c,
+ channels/chan_iax2.c: Convert a number of global module variables
+ to 'static'. These modules all contained variables that are
+ module-global but not system-global, but were not marked
+ 'static'.
+
+ * channels/chan_sip.c: Some minor structure size improvements in
+ sip_pvt and sip_peer. Using the 'pahole' tool, it is now quite
+ easy to see where structure fields could be organized differently
+ to keep the compiler from having to add padding to satisfy
+ alignment requirements. These changes reduced the sizes of
+ sip_pvt and sip_peer by a few bytes each (on 64-bit platforms),
+ and also fixed a spelling error in a field name.
+
+ * include/asterisk/agi.h, main/Makefile,
+ include/asterisk/autoconfig.h.in, res/res_smdi.exports,
+ configure.ac, res/res_agi.exports, include/asterisk/compiler.h,
+ apps/app_queue.c, res/res_monitor.c,
+ include/asterisk/optional_api.h, Makefile, res/res_smdi.c,
+ configure, res/res_agi.c, include/asterisk/monitor.h,
+ apps/app_stack.c, include/asterisk/smdi.h,
+ res/res_monitor.exports, apps/app_voicemail.c: Redesigned
+ 'optional API' support. This patch provides a new implementation
+ of the optional API support defined in asterisk/optional_api.h;
+ this new version provides solves compatibility issues with the
+ use of linker version scripts for suppressing global symbols. In
+ addition, there is now a functional (and tested!) implementation
+ for Mac OS/X, so module writers no longer need to use special
+ tests before calling optional API functions. All future
+ implementations must provide these same semantics, so that module
+ writers can rely on them.
+
+2009-06-15 15:22 +0000 [r200514] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 200513 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun
+ 2009) | 5 lines Add INFO to our allowed methods so that endpoints
+ know they may send it to us. AST-223 ........
+
+2009-06-14 06:13 +0000 [r200477] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
+ build_tools/menuselect-deps.in: added openr2 to
+ menuselect-deps.in, recent commit in menuselect made me realize
+ this was never done but was working anyways also added support
+ for skip category request feature of openr2 and updated
+ chan_dahdi.conf.sample
+
+2009-06-12 19:46 +0000 [r200428-200430] Sean Bright <sean@malleable.com>
+
+ * contrib/upstart/asterisk.upstart-0.3.9: Include basic
+ installation and usage instructions for upstart script.
+
+ * contrib/upstart/asterisk.upstart-0.3.9 (added), contrib/upstart
+ (added): First shot at an upstart script for asterisk on Ubuntu.
+ This works relatively well (assuming you are using
+ /var/run/asterisk) as your run directory and upstart 0.3.9. Needs
+ to be generalized and eventually added to the 'make install'
+ target for Ubuntu.
+
+2009-06-12 19:07 +0000 [r200290-200361] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 200360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun
+ 2009) | 10 lines Suppress a warning message and give a better
+ return code when generating inband ringing after a call is
+ answered. (closes issue #15158) Reported by: madkins Patches:
+ 15158.patch uploaded by mmichelson (license 60) Tested by:
+ madkins ........
+
+ * channels/chan_local.c, apps/app_queue.c: Fix some bad locking
+ stemming from trying to forward a call to a non-existent
+ extension from a queue.
+
+ * apps/app_queue.c: Fix a potential crash from trying to access a
+ NULL channel pointer.
+
+2009-06-12 02:20 +0000 [r200254] Sean Bright <sean@malleable.com>
+
+ * contrib/init.d/rc.debian.asterisk: Call chgrp instead of chown
+ when setting run directory group ownership. (issue #13153)
+ Reported by: pabelanger
+
+2009-06-11 21:17 +0000 [r200146] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix a crash due to a potentially NULL
+ p->options. Thanks to mnicholson for pointing it out.
+
+2009-06-11 15:40 +0000 [r200108] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/channel.c: Release the allocated channel decreasing the
+ reference counter. When allocating the channel use ao2_ref(-1) to
+ release it, instead of calling ast_free(). Also avoid freeing
+ structures inside that channel (on error) if they will be
+ released by the channel destructor being called if the reference
+ counter reachs 0.
+
+2009-06-11 12:15 +0000 [r200039] Leif Madsen <lmadsen@digium.com>
+
+ * build_tools/make_version_c, build_tools/make_version_h: Fix path
+ for .flavor and .version (issue #14737) Reported by: davidw
+ Patches: flavor.patch uploaded by davidw (license 780) Tested by:
+ davidw
+
+2009-06-10 20:40 +0000 [r200000] Sean Bright <sean@malleable.com>
+
+ * sample.call: Remove some trailing whitespace and steal revision
+ 200000.
+
+2009-06-10 20:15 +0000 [r199958] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Only try to use the invite_branch on
+ outgoing INVITEs with auth credentials. I have added a comment to
+ the code to help ease understanding of the logic here as well.
+
+2009-06-10 20:00 +0000 [r199957] David Brooks <dbrooks@digium.com>
+
+ * main/pbx.c: Fixes the argument order in definition of
+ new_find_extension(). In the definition of new_find_extension(),
+ the arguments 'callerid' and 'label' were swapped. The prototype
+ declaration and all calls to the function are ordered 'callerid'
+ then 'label', but the function itself was ordered 'label' then
+ 'callerid'. (closes issue #15303) Reported by: JimDickenson
+
+2009-06-10 18:58 +0000 [r199923] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c: Use ast_channel_unref to instead of ast_free on a
+ newly created channel. Also I removed an unnecessary free of a
+ cid_name. This will be freed properly in the channel destructor.
+ Reported by mnicholson in #asterisk-dev.
+
+2009-06-10 16:10 +0000 [r199857] Sean Bright <sean@malleable.com>
+
+ * include/asterisk/utils.h, /: Merged revisions 199856 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed,
+ 10 Jun 2009) | 2 lines __WORDSIZE is not available on all
+ platforms, so use sizeof(void *) instead. ........
+
+2009-06-09 20:47 +0000 [r199818] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: CLI NOTIFY sending wrong transport type.
+ SIP's cli NOTIFY command only used UDP rather than copying the
+ transport type from the peer. (closes issue #15283) Reported by:
+ jthurman Patches: sip-notify-tcp-svn199728.patch uploaded by
+ jthurman (license 614) Tested by: jthurman, dvossel
+
+2009-06-09 18:08 +0000 [r199781] Sean Bright <sean@malleable.com>
+
+ * Makefile: Fix all of the parallel build warnings issued when
+ running make -j#.
+
+2009-06-09 16:22 +0000 [r199743] David Vossel <dvossel@digium.com>
+
+ * res/res_timing_pthread.c, include/asterisk/module.h,
+ res/res_timing_dahdi.c, res/res_timing_timerfd.c, main/loader.c:
+ module load priority This patch adds the option to give a module
+ a load priority. The value represents the order in which a
+ module's load() function is initialized. The lower the value, the
+ higher the priority. The value is only checked if the
+ AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER
+ flag is not set, the value will never be read and the module will
+ be given the lowest possible priority on load. Since some modules
+ are reliant on a timing interface, the timing modules have been
+ given a high load priorty. (closes issue #15191) Reported by:
+ alecdavis Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/262/
+
+2009-06-08 22:08 +0000 [r199696] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/janitor-projects.txt: Add sigaction janitor
+
+2009-06-08 19:33 +0000 [r199630] Sean Bright <sean@malleable.com>
+
+ * include/asterisk/utils.h, /: Merged revisions 199626,199628 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun
+ 2009) | 21 lines Increase the size of our thread stack on 64 bit
+ processors. We were setting the stack size for each thread to
+ 240KB regardless of architecture, which meant that in some
+ scenarios we actually had less available stack space on 64 bit
+ processors (pointers use 8 bytes instead of 4). So now we
+ calculate the stack size we reserve based on the platform's
+ __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128
+ bit -> 1008KB (that's right, we're ready for 128 bit processors)
+ Patch typed by me but written by several members of
+ #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes
+ issue #14932) Reported by: jpiszcz Patches:
+ 06052009_issue14932.patch uploaded by seanbright (license 71)
+ Tested by: seanbright ........ r199628 | seanbright | 2009-06-08
+ 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the
+ stack size calculation just introduced. ........
+
+2009-06-08 17:32 +0000 [r199588] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix a deadlock that could occur when setting
+ rtp stats on SIP calls. (closes issue #15143) Reported by:
+ cristiandimache Patches: 15143.patch uploaded by mmichelson
+ (license 60) Tested by: cristiandimache
+
+2009-06-07 19:15 +0000 [r199514-199547] Eliel C. Sardanons <eliels@gmail.com>
+
+ * apps/app_osplookup.c: Move OSP* applications static documentation
+ to XML. Move OSP* applications static documentation to the new
+ AstXML form. (closes issue #15245) Reported by: eliel Patches:
+ app_osplookup_static_conversion.txt uploaded by lmadsen (license
+ 10)
+
+ * apps/app_externalivr.c: Move application ExternalIVR static
+ documentation to XML. Move application ExternalIVR static
+ documentation to the new AstXML form. (issue #15245) Reported by:
+ eliel Patches: app_externalivr.diff uploaded by eliel (license
+ 64)
+
+2009-06-07 14:55 +0000 [r199479] Russell Bryant <russell@digium.com>
+
+ * apps/app_dial.c, apps/app_dahdibarge.c, apps/app_dictate.c,
+ apps/app_authenticate.c, apps/app_echo.c, apps/app_fax.c,
+ apps/app_dahdiras.c, apps/app_disa.c, apps/app_alarmreceiver.c,
+ apps/app_chanisavail.c, apps/app_exec.c, apps/app_db.c,
+ apps/app_controlplayback.c, apps/app_channelredirect.c,
+ apps/app_directed_pickup.c, apps/app_dumpchan.c, apps/app_amd.c,
+ apps/app_confbridge.c, apps/app_directory.c, apps/app_chanspy.c,
+ apps/app_adsiprog.c: Global var cleanup - constification and
+ removing unused vars.
+
+2009-06-06 23:28 +0000 [r199374-199446] Eliel C. Sardanons <eliels@gmail.com>
+
+ * apps/app_stack.c: Move AGI command 'gosub' static documentation
+ to XML. Move AGI command 'gosub' statis documentation to the new
+ AstXML form. (issue #15245) Reported by: eliel Patches:
+ app_stack_static_conversion.txt uploaded by lmadsen (license 10)
+ (with minor changes by me)
+
+ * res/res_musiconhold.c: Move music on hold related applications
+ documentation to XML. Move MusicOnHold, SetMusicOnHold,
+ StartMusicOnHold, StopMusicOnHold static documentation to the new
+ AstXML form. (issue #15245) Reported by: eliel Patches:
+ res_musiconhold_static_conversion.txt uploaded by lmadsen
+ (license 10) (with some fixes and formatting by me)
+
+ * res/res_phoneprov.c: Move function PP_EACH_USER and
+ PP_EACH_EXTENSION documentation to XML. Move function
+ PP_EACH_USER and PP_EACH_EXTENSION documentation to the new
+ AstXML form. (issue #15245) Reported by: eliel Patches:
+ res_phoneprov_static_conversion.txt uploaded by lmadsen (license
+ 10) (with PP_EACH_USER add by me)
+
+ * apps/app_meetme.c: Move function MEETME_INFO documentation to
+ XML. Move function MEETME_INFO static documentation to the new
+ AstXML form. (issue #15245) Reported by: eliel Patches:
+ app_meetme_static_conversion.txt uploaded by lmadsen (license 10)
+
+ * apps/app_minivm.c: Move function MINIVMACCOUNT and MINIVMCOUNTER
+ static documentation to XML. Move function MINIVMACCOUNT and
+ MINIVMCOUNTER statis documentation to the new AstXML form. (issue
+ #15245) Reported by: eliel Patches:
+ app_minivm_static_conversion.txt uploaded by lmadsen (license 10)
+ (with minor changes by me)
+
+ * funcs/func_sysinfo.c: Move function SYSINFO documentation to XML.
+ Move function SYSINFO static documentation to the new AstXML
+ form. (issue #15245) Reported by: eliel Patches:
+ func_sysinfo_static_conversion.txt uploaded by lmadsen (license
+ 10)
+
+2009-06-06 21:42 +0000 [r199368-199372] Russell Bryant <russell@digium.com>
+
+ * apps/app_jack.c: minor tweak
+
+ * apps/app_jack.c: Constify a string and strip trailing whitespace.
+
+ * Makefile: Switch from "echo -n" to printf. On my mac, the -n was
+ just getting printed out.
+
+2009-06-05 21:21 +0000 [r199298] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/devicestate.h, /, main/devicestate.c: Merged
+ revisions 199297 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009)
+ | 14 lines Fixes issue with hints giving unexpected results.
+ Hints with two or more devices that include ONHOLD gave
+ unexpected results. (closes issue #15057) Reported by:
+ p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel
+ (license 671) pbx.c.1.4.patch uploaded by p (license 558)
+ devicestate.c.trunk.patch uploaded by p (license 671) Tested by:
+ p_lindheimer, dvossel Review:
+ https://reviewboard.asterisk.org/r/254/ ........
+
+2009-06-05 13:51 +0000 [r199227] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_dahdi.c: Correct "dahdi show channels" output when
+ specifying a group. Since a DAHDI channel may belong to multiple
+ groups, we need to use a bitwise and instead of equivalence to
+ determine whether to display the channel information. (closes
+ issue #15248) Reported by: gentian Patches: 15248.patch uploaded
+ by mmichelson (license 60) Tested by: gentian
+
+2009-06-04 19:10 +0000 [r199139] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 199138 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04
+ Jun 2009) | 3 lines Additional updates to AST-2009-001 ........
+
+2009-06-04 16:29 +0000 [r199091] Eliel C. Sardanons <eliels@gmail.com>
+
+ * res/res_smdi.c: Move static docs to the new AstXML form. Move
+ SMDI_MSG and SMDI_MSG_RETRIEVE functions statis documentation to
+ XML. (issue #15245) Reported by: eliel Patches:
+ res_smdi_static_conversion.txt uploaded by lmadsen (license 10)
+
+2009-06-04 14:31 +0000 [r199051] Sean Bright <sean@malleable.com>
+
+ * /, include/asterisk/_private.h, main/asterisk.c, main/loader.c:
+ Merged revisions 199022 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun
+ 2009) | 40 lines Safely handle AMI connections/reload requests
+ that occur during startup. During asterisk startup, a lock on the
+ list of modules is obtained by the primary thread while each
+ module is initialized. Issue 13778 pointed out a problem with
+ this approach, however. Because the AMI is loaded before other
+ modules, it is possible for a module reload to be issued by a
+ connected client (via Action: Command), causing a deadlock. The
+ resolution for 13778 was to move initialization of the manager to
+ happen after the other modules had already been lodaded. While
+ this fixed this particular issue, it caused a problem for users
+ (like FreePBX) who call AMI scripts via an #exec in a
+ configuration file (See issue 15189). The solution I have come up
+ with is to defer any reload requests that come in until after the
+ server is fully booted. When a call comes in to ast_module_reload
+ (from wherever) before we are fully booted, the request is added
+ to a queue of pending requests. Once we are done booting up, we
+ then execute these deferred requests in turn. Note that I have
+ tried to make this a bit more intelligent in that it will not
+ queue up more than 1 request for the same module to be reloaded,
+ and if a general reload request comes in ('module reload') the
+ queue is flushed and we only issue a single deferred reload for
+ the entire system. As for how this will impact existing
+ installations - Before 13778, a reload issued before module
+ initialization was completed would result in a deadlock. After
+ 13778, you simply couldn't connect to the manager during startup
+ (which causes problems with #exec-that-calls-AMI configuration
+ files). I believe this is a good general purpose solution that
+ won't negatively impact existing installations. (closes issue
+ #15189) (closes issue #13778) Reported by: p_lindheimer Patches:
+ 06032009_15189_deferred_reloads.diff uploaded by seanbright
+ (license 71) Tested by: p_lindheimer, seanbright Review:
+ https://reviewboard.asterisk.org/r/272/ ........
+
+2009-06-03 20:30 +0000 [r198824-198954] David Vossel <dvossel@digium.com>
+
+ * apps/app_dial.c, main/channel.c, apps/app_queue.c:
+ ast_call_forward() todo notes and originate flag copy.
+
+ * main/channel.c, main/features.c, include/asterisk/channel.h:
+ Generic call forward api, ast_call_forward() The function
+ ast_call_forward() forwards a call to an extension specified in
+ an ast_channel's call_forward string. After an ast_channel is
+ called, if the channel's call_forward string is set this function
+ can be used to forward the call to a new channel and terminate
+ the original one. I have included this api call in both
+ channel.c's ast_request_and_dial() and feature.c's
+ feature_request_and_dial(). App_dial and app_queue already
+ contain call forward logic specific for their application and
+ options. (closes issue #13630) Reported by: festr Review:
+ https://reviewboard.asterisk.org/r/271/
+
+ * channels/chan_iax2.c: fixes issue with channels not going down
+ after transfer Iax2 currently does not support native bridging if
+ the timeoutms value is set. We check for that in iax2_bridge, but
+ then set timeoutms to 0 by default. If the timeoutms is not
+ provided it is set to -1. By setting timeoutms to 0 it is
+ processed causing a bridging retry loop. (closes issue #15216)
+ Reported by: oxymoron Tested by: dvossel
+
+2009-06-02 13:48 +0000 [r198762-198791] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: Correct
+ documentation for the register line, specifically where the
+ domain should be specified. (closes issue #14367) Reported by:
+ Nick_Lewis
+
+ * main/rtp_engine.c: Fix a bug where we were passing in address
+ information that should remain unmodified to a function that may
+ modify it. (closes issue #15243) Reported by: pj
+
+2009-06-01 21:03 +0000 [r198729] Russell Bryant <russell@digium.com>
+
+ * channels/iax2-parser.c: Tell the IAX2 parser about more control
+ frame types.
+
+2009-06-01 20:57 +0000 [r198727] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, main/channel.c, include/asterisk/app.h,
+ main/dial.c, channels/chan_sip.c, apps/app_directed_pickup.c,
+ main/features.c, apps/app_macro.c, doc/tex/channelvariables.tex,
+ main/app.c, include/asterisk/channel.h, apps/app_queue.c: Add the
+ ability to execute connected line interception macros. When
+ connected line updates are received or generated in the middle of
+ an application call, it is now possible to execute a macro to
+ manipulate the connected line data. This way, phone numbers may
+ be manipulated to be more presentable to users, names may be
+ changed for...whatever reason, or whatever else needs to be done
+ may be. Review: https://reviewboard.asterisk.org/r/256 AST-165
+
+2009-06-01 20:33 +0000 [r198725] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_math.c: Add INCrement and DECrement functions (closes
+ issue #15025) Reported by: greenfieldtech Patches:
+ func_math.c.patch_v4 uploaded by greenfieldtech (license 369)
+ slightly modified by me Tested by: greenfieldtech, lmadsen
+
+2009-06-01 20:17 +0000 [r198670] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/frame.h: Minor whitespace fix.
+
+2009-06-01 19:37 +0000 [r198661] Eliel C. Sardanons <eliels@gmail.com>
+
+ * res/res_monitor.c: Moved more static documentation to the new
+ AstXML form. Moved more static docs to XML (pplications and
+ manager actions): Monitor, StopMonitor, ChangeMonitor,
+ PauseMonitor, UnpauseMonitor.
+
+2009-06-01 18:40 +0000 [r198626] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/meetme.sql: Add information for new meetme
+ realtime fields
+
+2009-06-01 17:53 +0000 [r198561-198597] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/Makefile: Do not add say.o in a separate line.
+
+ * res/res_jabber.c: Move JabberSend manager action from static docs
+ to the AstXML form.
+
+ * res/res_agi.c: Move static documentation of E|Dead|AGI()
+ application and manager action to XML.
+
+2009-06-01 15:23 +0000 [r198558] David Vossel <dvossel@digium.com>
+
+ * main/threadstorage.c: Fixed an issue in the threadstorage cli
+ functions resulting from the constification of struct
+ ast_cli_args in r196072.
+
+2009-06-01 14:45 +0000 [r198500-198530] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Remove extra lock from app_queue.
+
+ * channels/chan_local.c: Remove extra lock from local_indicate in
+ connected line case. Oh, and this fixes a deadlock I was seeing.
+
+ * channels/chan_local.c: Add missing unlock of local pvt.
+
+ * channels/chan_agent.c: Remove documentation for the 'exten'
+ argument to the AGENT function. Since AgentCallbackLogin has been
+ removed, this should not be documented any more.
+
+2009-06-01 13:31 +0000 [r198498] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix a bug where the Event and Content-Type
+ headers were added twice to outgoing SIP NOTIFY messages. (closes
+ issue #15239) Reported by: pj
+
+2009-05-31 17:52 +0000 [r198470] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_strings.c: Fix documentation for FIELDQTY.
+
+2009-05-31 02:09 +0000 [r198442] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/Makefile: Filter the say.o object, it is being added later.
+
+2009-05-31 01:40 +0000 [r198438] Russell Bryant <russell@digium.com>
+
+ * main/manager.c: Constification and remove some unused code.
+
+2009-05-31 01:22 +0000 [r198437] Eliel C. Sardanons <eliels@gmail.com>
+
+ * res/res_timing_dahdi.c: Avoid a crash when res_timing_dahdi is
+ unloaded but wasn't properly loaded. if dahdi_test_timer() fails,
+ timing_funcs_handle remains NULL causing a crash when calling
+ ast_unregister_timing_interface() with a NULL pointer. (closes
+ issue #15234) Reported by: eliel Patches: timing_dahdi1.diff
+ uploaded by eliel (license 64)
+
+2009-05-31 01:19 +0000 [r198434] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, include/asterisk/channel.h: Constify the
+ ast_frame arg to ast_queue_frame().
+
+2009-05-30 20:11 +0000 [r198371-198375] Sean Bright <sean@malleable.com>
+
+ * res/res_jabber.c: Properly terminate the receive buffer before
+ sending to iksemel. aji_io_recv takes the maximum number of bytes
+ to read (instead of the total buffer size), so we have to
+ subtract 1 from our buffer size. Without this, when we receive
+ packets that are larger than our buffer, iksemel will choke and
+ things get wonky. (closes issue #15232) Reported by: lp0 Patches:
+ 05302009_res_jabber.c.patch uploaded by seanbright (license 71)
+ Tested by: seanbright, lp0
+
+ * /, res/res_jabber.c: Merged revisions 198370 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May
+ 2009) | 12 lines Properly terminate AMI JabberSend response
+ messages. The response message (either Error or Success) needs an
+ extra trailing \r\n after the fields to inform the client that
+ the message is complete. (closes issue #14876) Reported by: srt
+ Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright
+ (license 71) asterisk_14876.patch uploaded by srt (license 378)
+ trunk-14876-2.diff uploaded by phsultan (license 73) ........
+
+2009-05-30 03:43 +0000 [r198312] Russell Bryant <russell@digium.com>
+
+ * res/res_smdi.c, /: Merged revisions 198311 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r198311 | russell | 2009-05-29 22:42:46 -0500 (Fri, 29 May 2009)
+ | 5 lines Fix a crash that occurred when MWI SMDI messages
+ expired. (closes issue #14561) Reported by: cmoss28 ........
+
+2009-05-30 03:26 +0000 [r198285] Sean Bright <sean@malleable.com>
+
+ * apps/app_dial.c, /: Merged revisions 198251 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May
+ 2009) | 8 lines Treat an empty FORWARD_CONTEXT the same way we
+ treat a missing one. (closes issue #15056) Reported by:
+ p_lindheimer Patches: 05292009_bug15056.diff uploaded by
+ seanbright (license 71) Tested by: p_lindheimer ........
+
+2009-05-30 02:31 +0000 [r198248] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: When removing all packets from a dialog we
+ also need to free the data if present.
+
+2009-05-30 01:04 +0000 [r198217] Eliel C. Sardanons <eliels@gmail.com>
+
+ * configs/agents.conf.sample, channels/chan_agent.c: Remove not
+ used code in the Agent channel. This code was there because of
+ the AgentCallbackLogin() application. ->loginchan[] member was
+ only used by AgentCallbackLogin(). Agent where dumped to astdb if
+ they where logged in using AgentCallbacklogin() so they are not
+ being dumper anymore. Review:
+ https://reviewboard.asterisk.org/r/267/
+
+2009-05-29 23:04 +0000 [r198183-198186] Russell Bryant <russell@digium.com>
+
+ * configs/modules.conf.sample: Suggesting that only a single timing
+ module be loaded is no longer necessary.
+
+ * res/res_timing_pthread.c: Improve handling of trying to ACK too
+ many timer expirations.
+
+2009-05-29 22:21 +0000 [r198182] Terry Wilson <twilson@digium.com>
+
+ * res/res_calendar.c: Add a couple of TODO items so I don't forget
+
+2009-05-29 20:06 +0000 [r198146] Russell Bryant <russell@digium.com>
+
+ * res/res_timing_pthread.c: Resolve issues with choppy sound when
+ using res_timing_pthread. The situation that caused this problem
+ was when continuous mode was being turned on and off while a rate
+ was set for a timing interface. A very easy way to replicate this
+ bug was to do a Playback() from behind a Local channel. In this
+ scenario, a rate gets set on the channel for doing file playback.
+ At the same time, continuous mode gets turned on and off about
+ every 20 ms as frames get queued on to the PBX side channel from
+ the other side of the Local channel. Essentially, this module
+ treated continuous mode and a set rate as mutually exclusive
+ states for the timer to be in. When I dug deep enough, I observed
+ the following pattern: 1) Set timer to tick every 20 ms. 2) Wait
+ almost 20 ms ... 3) Continuous mode gets turned on for a queued
+ up frame 4) Continuous mode gets turned off 5) The timer goes
+ back to its tick per 20 ms. state but starts counting at 0 ms. 6)
+ Goto step 2. Sometimes, res_timing_pthread would make it 20 ms
+ and produce a timer tick, but not most of the time. This is what
+ produced the choppy sound (or sometimes no sound at all). Now,
+ the module treats continuous mode and a set rate as completely
+ independent timer modes. They can be enabled and disabled
+ independently of each other and things work as expected. (closes
+ issue #14412) Reported by: dome Patches: issue14412.diff.txt
+ uploaded by russell (license 2) issue14412-1.6.1.0.diff.txt
+ uploaded by russell (license 2) Tested by: DennisD, russell
+
+2009-05-29 19:46 +0000 [r198139] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/Makefile: Simplify the Makefile and avoid needing to specify
+ each object file. Instead of specifying every object file, use
+ make's magic to generate it. This will generate less conflicts in
+ team branches when a new file is added in trunk. (closes issue
+ #15226) Reported by: eliel Patches: makefile uploaded by eliel
+ (license 64) Review: http://reviewboard.asterisk.org/r/269/
+
+2009-05-29 19:19 +0000 [r198088] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c (added),
+ channels/sig_analog.h (added), channels/Makefile: New signaling
+ module to handle analog operations in chan_dahdi This branch
+ splits all the analog signaling logic out of chan_dahdi.c into
+ sig_analog.c. Functionality in theory should not change at all.
+ As noted in the code, there is still some unused code remaining
+ that will be cleaned up in a later commit. Review:
+ https://reviewboard.asterisk.org/r/253/
+
+2009-05-29 19:18 +0000 [r198083] Eliel C. Sardanons <eliels@gmail.com>
+
+ * CREDITS: Apply anti-spam obfuscation to an email address.
+
+2009-05-29 19:04 +0000 [r198072] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/cdr.c, main/channel.c, /, include/asterisk/cdr.h: Merged
+ revisions 198068 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May
+ 2009) | 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as
+ the default CDR disposition. This change also involves the
+ addition of an AST_CDR_FLAG_ORIGINATED flag that is used on
+ originated channels to distinguish: them from dialed channels.
+ (closes issue #12946) Reported by: meral Patches: null-cdr2.diff
+ uploaded by mnicholson (license 96) Tested by: mnicholson,
+ dbrooks (closes issue #15122) Reported by: sum Tested by: sum
+ ........
+
+2009-05-29 18:39 +0000 [r198064] Joshua Colp <jcolp@digium.com>
+
+ * main/file.c: Fix a memory leak of the write buffer when writing a
+ file.
+
+2009-05-29 18:15 +0000 [r198000] Sean Bright <sean@malleable.com>
+
+ * Makefile, /: Merged revisions 197998 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r197998 | seanbright | 2009-05-29 14:14:12 -0400 (Fri, 29 May
+ 2009) | 8 lines Fix 'make config' target for Slackware. There was
+ a missing semi-colon after the echo statement in the Makefile
+ that was causing problems for some users. Fix suggested by
+ reporter. (closes issue #15225) Reported by: pdavis ........
+
+2009-05-29 17:51 +0000 [r197996] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix a bug where the default setting did not
+ perform a remote bridge when it should have.
+
+2009-05-29 16:15 +0000 [r197960] Russell Bryant <russell@digium.com>
+
+ * res/res_timing_pthread.c: Trim trailing whitespace so that I can
+ work on this bug without it bothering me. :-)
+
+2009-05-29 15:48 +0000 [r197959] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: A few fixes to SIP with regards to connected
+ line updates during transfers. * Set the invitestate to
+ INV_CALLING when we send a connected line reinvite. This prevents
+ us from potentially rapid-firing reinvites to a single peer. *
+ Use the astdb to store a peer's allowed methods. This prevents us
+ from sending an UPDATE during the interval between startup and
+ the peer's first registration if the peer does not support the
+ UPDATE method. * Handle Polycom's method of indicating allowed
+ methods in REGISTER. Instead of using an Allow header, they place
+ the allowed methods in a methods= parameter in the Contact
+ header. ABE-1873
+
+2009-05-29 05:15 +0000 [r197926] Terry Wilson <twilson@digium.com>
+
+ * doc/tex/asterisk.tex, doc/tex/calendaring.tex (added): Add some
+ TeX docs for calendaring. I still need to set up tests to make
+ sure my examples are completely correct, but I ran out of time
+ tonight and felt that they at least would give an idea as to how
+ to use calendaring. I will try to test the examples and do some
+ cleanup on the docs tomorrow night.
+
+2009-05-28 22:42 +0000 [r197861] Sean Bright <sean@malleable.com>
+
+ * include/asterisk/doxygen/releases.h, sounds/Makefile: Update
+ references to downloads.digium.com to its new URL.
+
+2009-05-28 22:04 +0000 [r197828] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_mixmonitor.c: Update documentation in MixMonitor.
+ Updated the MixMonitor documentation for the 'b' option so that
+ it is more obvious that you must not optimize away the Local
+ channel when using this option. (closes issue #14829) Reported
+ by: licedey Tested by: mmichelson, licedey, lmadsen
+
+2009-05-28 21:50 +0000 [r197824] Sean Bright <sean@malleable.com>
+
+ * doc/CODING-GUIDELINES, doc/asterisk.8, BUGS, doc/backtrace.txt,
+ doc/tex/mp3.tex, channels/h323/README, main/enum.c,
+ doc/tex/misdn.tex, include/asterisk/doxyref.h,
+ contrib/scripts/ast_grab_core, doc/tex/backtrace.tex,
+ include/asterisk/doxygen/reviewboard.h,
+ include/asterisk/doxygen/commits.h,
+ contrib/scripts/asterisk.ldif,
+ contrib/scripts/asterisk.ldap-schema,
+ configs/extensions.conf.sample, doc/asterisk.sgml: Update
+ references to bugs.digium.com and reviewboard.digium.com to the
+ new URLs.
+
+2009-05-28 20:43 +0000 [r197777] Terry Wilson <twilson@digium.com>
+
+ * configs/calendar.conf.sample: Make note of Exchange calendar
+ support limitations
+
+2009-05-28 20:36 +0000 [r197775] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/utils.c: Ensure that accidental calls to
+ ast_string_field_free_memory() on embedded stringfield pools are
+ safe. It is possible for a stringfield manager structure (and
+ pool) structure to be allocated as part of a larger structure
+ allocation (using ast_calloc_with_strinfields()); when this is
+ done, the stringfield pool cannot be separately freed, but users
+ of the tructure may not be aware (and shouldn't have to be aware)
+ of whether the pool was embedded. This patch modifies the
+ behavior so that they can always call
+ ast_string_field_free_memory() and the function will do the right
+ thing for both embedded and non-embedded situations.
+
+2009-05-28 20:17 +0000 [r197740] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Treat 405 responses the same way we would a
+ 501. This makes sure that we mark a method as being unallowed if
+ we receive a 405 response so that we don't continue to try to
+ send that same type of message.
+
+2009-05-28 19:57 +0000 [r197738] Terry Wilson <twilson@digium.com>
+
+ * res/res_calendar.exports (added), res/res_calendar_exchange.c
+ (added), res/res_calendar_icalendar.c (added),
+ build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ configs/calendar.conf.sample (added), res/res_calendar_caldav.c
+ (added), include/asterisk/calendar.h (added), makeopts.in,
+ res/res_calendar.c (added): Add Calendaring support for Asterisk
+ This commit add Calendaring support to Asterisk for iCalendar,
+ CalDAV, and MS Exchange calendars. Exchange support has only been
+ tested on Exchange Server 2k3 and does not support forms-based
+ authentication at this time (patches *very* welcome). Exchange
+ support is also currently missing the ability to return a list of
+ a meting's attendees (again, patches are very, very welcome).
+ Features include: Querying a calendar for events over a specific
+ time range Checking a calendar's busy status via the dialplan
+ Writing calendar events via the dialplan (CalDAV and Exchange
+ only) Handling calendar event notifications through the dialplan
+ (closes issue #14771) Tested by: lmadsen, twilson, Shivaprakash
+ Review: https://reviewboard.asterisk.org/r/58
+
+2009-05-28 18:48 +0000 [r197701] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_local.c: Add missing lock to local_indicate
+ function for connected line frames.
+
+2009-05-28 18:45 +0000 [r197697] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: Fix a bug where the trunkmtu setting was
+ not set to the default value of 1240 on load but was on reload.
+
+2009-05-28 16:01 +0000 [r197621] Eliel C. Sardanons <eliels@gmail.com>
+
+ * /, channels/chan_sip.c: Merged revisions 197562 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) |
+ 13 lines Use the address we already know when reloading a peer
+ with nat=yes. If we already have an address for a peer, and we
+ are reloading the sip configuration, try to use that address to
+ contact the peer, instead of getting it from the Contact. (closes
+ issue #15194) Reported by: ibc Patches: sip.patch uploaded by
+ eliel (license 64) Tested by: manwe ........
+
+2009-05-28 15:35 +0000 [r197616] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c, channels/chan_console.c, apps/app_rpt.c,
+ main/astobj2.c, main/cli.c: Eliminate several needless checks and
+ fix a few memory leaks (closes issue #14833) Reported by:
+ contactmayankjain Patches: all_changes.patch uploaded by
+ contactmayankjain (license 740) slightly modified by me
+
+2009-05-28 15:32 +0000 [r197606] Mark Michelson <mmichelson@digium.com>
+
+ * /: Recorded merge of revisions 197588 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May
+ 2009) | 16 lines Allow for media to arrive from an alternate
+ source when responding to a reinvite with 491. When we receive a
+ SIP reinvite, it is possible that we may not be able to process
+ the reinvite immediately since we have also sent a reinvite out
+ ourselves. The problem is that whoever sent us the reinvite may
+ have also sent a reinvite out to another party, and that reinvite
+ may have succeeded. As a result, even though we are not going to
+ accept the reinvite we just received, it is important for us to
+ not have problems if we suddenly start receiving RTP from a new
+ source. The fix for this is to grab the media source information
+ from the SDP of the reinvite that we receive. This information is
+ passed to the RTP layer so that it will know about the alternate
+ source for media. Review: https://reviewboard.asterisk.org/r/252
+ ........
+
+2009-05-28 15:23 +0000 [r197570] Joshua Colp <jcolp@digium.com>
+
+ * main/logger.c: Fix an incorrect call to
+ ast_string_field_free_memory which caused a crash in the logger.
+ Since the message structure is allocated using
+ ast_calloc_with_stringfields we do not need to free the memory
+ used for the stringfields as it will get freed when the message
+ structure is.
+
+2009-05-28 14:58 +0000 [r197543] Mark Michelson <mmichelson@digium.com>
+
+ * /, include/asterisk/audiohook.h, main/audiohook.c,
+ apps/app_chanspy.c: Merged revisions 197537 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May
+ 2009) | 21 lines Add flags to chanspy audiohook so that audio
+ stays in sync. There are two flags being added to the chanspy
+ audiohook here. One is the pre-existing
+ AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that
+ the read and write slinfactories on the audiohook do not skew
+ beyond a certain tolerance. In addition, there is a new audiohook
+ flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set,
+ we do not allow for a slinfactory to build up a substantial
+ amount of audio before flushing it. For this particular issue,
+ this means that the person spying on the call will hear the
+ conversations in real time with very little delay in the audio.
+ (closes issue #13745) Reported by: geoffs Patches: 13745.patch
+ uploaded by mmichelson (license 60) Tested by: snblitz ........
+
+2009-05-28 14:51 +0000 [r197538] Joshua Colp <jcolp@digium.com>
+
+ * main/utils.c: Fix a bug in stringfields where it did not actually
+ free the pools of memory. (closes issue #15074) Reported by: pj
+
+2009-05-28 14:39 +0000 [r197528-197535] Sean Bright <sean@malleable.com>
+
+ * configs/amd.conf.sample, configs/users.conf.sample,
+ configs/gtalk.conf.sample, configs/rpt.conf.sample,
+ configs/rtp.conf.sample, configs/cli_aliases.conf.sample,
+ configs/modules.conf.sample, configs/phone.conf.sample,
+ configs/extensions.ael.sample, configs/skinny.conf.sample,
+ configs/ais.conf.sample, configs/meetme.conf.sample,
+ configs/extensions_minivm.conf.sample, configs/telcordia-1.adsi,
+ configs/alsa.conf.sample, configs/iax.conf.sample,
+ configs/followme.conf.sample, configs/mgcp.conf.sample,
+ configs/sip.conf.sample, configs/extensions.lua.sample,
+ configs/say.conf.sample, configs/queuerules.conf.sample,
+ configs/minivm.conf.sample, configs/osp.conf.sample,
+ configs/chan_dahdi.conf.sample,
+ configs/cli_permissions.conf.sample, configs/console.conf.sample,
+ configs/dundi.conf.sample, configs/indications.conf.sample,
+ configs/oss.conf.sample, configs/queues.conf.sample,
+ configs/voicemail.conf.sample, configs/usbradio.conf.sample,
+ configs/cdr.conf.sample, configs/jingle.conf.sample,
+ configs/misdn.conf.sample, configs/manager.conf.sample,
+ configs/festival.conf.sample, configs/features.conf.sample,
+ configs/logger.conf.sample, configs/http.conf.sample,
+ configs/h323.conf.sample, configs/sla.conf.sample,
+ configs/phoneprov.conf.sample, configs/res_odbc.conf.sample,
+ configs/agents.conf.sample, configs/alarmreceiver.conf.sample,
+ configs/func_odbc.conf.sample, configs/musiconhold.conf.sample,
+ configs/jabber.conf.sample, configs/extconfig.conf.sample,
+ configs/res_snmp.conf.sample, configs/iaxprov.conf.sample,
+ configs/unistim.conf.sample, configs/dnsmgr.conf.sample,
+ configs/extensions.conf.sample, configs/asterisk.adsi: Remove a
+ bunch of trailing whitespace in preparation for
+ reformatting/cleanup. Let's try that again, this time removing
+ trailing whitespace and not leading whitespace. I can't believe
+ no one noticed.
+
+ * configs/amd.conf.sample, configs/gtalk.conf.sample,
+ configs/rtp.conf.sample, configs/rpt.conf.sample,
+ configs/cli_aliases.conf.sample, configs/extensions.ael.sample,
+ configs/skinny.conf.sample, configs/meetme.conf.sample,
+ configs/telcordia-1.adsi, configs/alsa.conf.sample,
+ configs/iax.conf.sample, configs/mgcp.conf.sample,
+ configs/extensions.lua.sample, configs/sip.conf.sample,
+ configs/say.conf.sample, configs/minivm.conf.sample,
+ configs/console.conf.sample, configs/cli_permissions.conf.sample,
+ configs/chan_dahdi.conf.sample, configs/oss.conf.sample,
+ configs/queues.conf.sample, configs/jingle.conf.sample,
+ configs/usbradio.conf.sample, configs/voicemail.conf.sample,
+ configs/misdn.conf.sample, configs/manager.conf.sample,
+ configs/features.conf.sample, configs/h323.conf.sample,
+ configs/sla.conf.sample, configs/res_odbc.conf.sample,
+ configs/phoneprov.conf.sample, configs/alarmreceiver.conf.sample,
+ configs/func_odbc.conf.sample, configs/musiconhold.conf.sample,
+ configs/jabber.conf.sample, configs/unistim.conf.sample,
+ configs/dnsmgr.conf.sample, configs/extensions.conf.sample,
+ configs/asterisk.adsi: Remove a bunch of trailing whitespace in
+ preparation for reformatting/cleanup.
+
+2009-05-28 13:47 +0000 [r197467] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 197466 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8
+ lines Fix a bug where the flag indicating the presence of rport
+ would get overwritten by the nat setting. The presence of rport
+ is now stored as a separate flag. Once the dialog is setup and
+ authenticated (or it passes through unauthenticated) the proper
+ nat flag is set. (closes issue #13823) Reported by: dimas
+ ........
+
+2009-05-28 11:25 +0000 [r197406-197431] Gavin Henry <ghenry@suretecsystems.com>
+
+ * contrib/scripts/asterisk.ldap-schema,
+ contrib/scripts/asterisk.ldif: Added AstVoicemailContext Added
+ AstVoicemailContext (closes issue #15155) Reported by: scramatte
+ Tested by: suretec
+
+ * contrib/scripts/asterisk.ldap-schema,
+ contrib/scripts/asterisk.ldif: New objectclass AsteriskVoiceMail
+ and AstAccountCallLimit attribute Added new ObjectClass
+ AsteriskVoiceMail, and AstAccountCallLimit attribute and cleaned
+ up formatting and tested with OpenLDAP (closes issue #15155)
+ Reported by: scramatte Patches: asterisk.schema uploaded by
+ scramatte (license 796) Tested by: suretec Review: [full review
+ board URL with trailing slash]
+
+ * doc/ldap.txt, configs/res_ldap.conf.sample,
+ contrib/scripts/asterisk.ldap-schema,
+ contrib/scripts/asterisk.ldif: closes issue #15156
+
+2009-05-27 23:48 +0000 [r197374] Tilghman Lesher <tlesher@digium.com>
+
+ * main/xml.c: Revert commit 192032. This define is needed on Mac OS
+ X.
+
+2009-05-27 22:42 +0000 [r197338] Russell Bryant <russell@digium.com>
+
+ * main/rtp_engine.c: Don't do a pointer comparison before setting
+ the remote address.
+
+2009-05-27 22:21 +0000 [r197335] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/agi.h: Ensure that this header includes
+ xmldoc.h, since it depends on it.
+
+2009-05-27 20:14 +0000 [r197266] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Adding some generic handling of error codes
+ sent to us in replys to requests. Previously they always set
+ hangupcause 0, which is generally wrong. With this change, we're
+ setting some generic hangup causes. For 5xx errors, which
+ indicate some sort of problem with the remote server, we're now
+ setting CONGESTION. EDVX002
+
+2009-05-27 20:08 +0000 [r197260] Sean Bright <sean@malleable.com>
+
+ * Makefile: Use bash explicitly when calling
+ build_tools/mkpkgconfig from the Makefile. Since we use bashisms
+ in build_tools/mkpkgconfig, we should call on bash explicitly
+ when running from the Makefile, otherwise we get errors during a
+ 'make install.' (closes issue #15209) Reported by: seandarcy
+
+2009-05-27 19:20 +0000 [r197209] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_cut.c: Recorded merge of revisions 197194 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r197194 | tilghman | 2009-05-27 14:09:42 -0500 (Wed, 27 May 2009)
+ | 5 lines Use a different determinator on whether to print the
+ delimiter, since leading fields may be blank. (closes issue
+ #15208) Reported by: ramonpeek Patch by me, though inspired in
+ part by a patch from ramonpeek ........
+
+2009-05-27 18:25 +0000 [r196948-197189] Sean Bright <sean@malleable.com>
+
+ * configs/adtranvofr.conf.sample (removed): Remove a file sample
+ configuration file that is no longer used.
+
+ * configs/chan_dahdi.conf.sample, configs/vpb.conf.sample,
+ configs/smdi.conf.sample, configs/extensions.conf.sample,
+ configs/sla.conf.sample: Fix references to /etc/dahdi/system.conf
+ and /etc/asterisk/chan_dahdi.conf in the sample configuration
+ files. (closes issue #15207) Reported by: seandarcy
+
+ * channels/chan_alsa.c: Display an error message when chan_alsa
+ fails to load due to a missing or inaccessible configuration
+ file. Before this change, when chan_alsa failed to load due to a
+ missing or inaccessible configuration file, no message would be
+ displayed. With this change, when chan_alsa fails to load due to
+ a missing or inaccessible configuration file, a message will be
+ displayed. (closes issue #14760) Reported by: Nick_Lewis Patches:
+ chan_alsa.c-confload.patch uploaded by Nick (license 657)
+
+ * main/xmldoc.c: Reset the terminal to the correct fg/bg after XML
+ documenation is rendered. (closes issue #15200) Reported by:
+ ajohnson Patches: 05262009_xmldoc.patch uploaded by seanbright
+ (license 71) Tested by: ajohnson
+
+2009-05-26 22:40 +0000 [r196946] Russell Bryant <russell@digium.com>
+
+ * autoconf/ast_check_osptk.m4 (added), configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Update configure
+ script to check for OSP toolkit 3.5.0. (closes issue #14988)
+ Reported by: tzafrir Patches: configure.ac.diff uploaded by
+ homesick (license 91) new_ast_check_osptk.m4 uploaded by homesick
+ (license 91)
+
+2009-05-26 22:38 +0000 [r196907-196945] Sean Bright <sean@malleable.com>
+
+ * main/manager.c: Add ActionID to CoreShowChannel event. There is
+ inconsistency in how we handle manager responses that are lists
+ of items and, unfortunately, third parties have come to rely on
+ ActionID being on every event within those lists instead of just
+ keeping track of the ActionID for the current response. This
+ change makes CoreShowChannels include the ActionID with each
+ CoreShowChannel event generated as a result of it being called.
+ (closes issue #15001) Reported by: sum Patches:
+ patchactionid2.patch uploaded by sum (license 766)
+
+ * main/manager.c: Include startup and reload date in the CoreStatus
+ manager message. The CoreStartupTime and CoreReloadTime
+ name/value pairs in the CoreStatus response message only included
+ the time and not the date. This patch, inspired by the reporter's
+ patch, adds 2 new fields - CoreStartupDate and CoreReloadDate -
+ which contain the date portion of these values. (closes issue
+ #15000) Reported by: sum
+
+2009-05-26 19:50 +0000 [r196893] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, apps/app_directed_pickup.c: Remove some
+ redundant or unnecessary connected line-related function calls.
+
+2009-05-26 18:20 +0000 [r196843] Russell Bryant <russell@digium.com>
+
+ * /, res/res_convert.c: Merged revisions 196826 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r196826 | russell | 2009-05-26 13:14:36 -0500 (Tue, 26 May 2009)
+ | 9 lines Resolve a file handle leak. The frames here should have
+ always been freed. However, out of luck, there was never any
+ memory leaked. However, after file streams became reference
+ counted, this code would leak the file stream for the file being
+ read. (closes issue #15181) Reported by: jkroon ........
+
+2009-05-26 16:38 +0000 [r196725-196792] Sean Bright <sean@malleable.com>
+
+ * apps/app_queue.c: Add a missing unref for queues in
+ handle_statechange.
+
+ * main/pbx.c, include/asterisk/pbx.h, res/res_clioriginate.c: Add
+ new ast_complete_applications function so that we can use it with
+ the 'channel originate ... application <app>' CLI command. (And
+ yeah, I cleaned up some whitespace in res_clioriginate.c... big
+ whoop, wanna fight about it!?)
+
+ * cdr/cdr_sqlite3_custom.c: Use a properly allocated channel for
+ substitution in cdr_sqlite3_custom.
+
+2009-05-26 13:43 +0000 [r196658-196721] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix a bug where the sip unregister CLI
+ command did not completely unregister the peer. (closes issue
+ #15118) Reported by: alecdavis Patches:
+ chan_sip_unregister.diff2.txt uploaded by alecdavis (license 585)
+
+ * /, contrib/scripts/safe_asterisk: Merged revisions 196657 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r196657 | file | 2009-05-26 10:06:09 -0300 (Tue, 26 May 2009) | 7
+ lines Remove some bash specific stuff from safe_asterisk. (closes
+ issue #10812) Reported by: paravoid Patches:
+ safe_asterisk_bashism.diff uploaded by tzafrir (license 46)
+ ........
+
+2009-05-26 12:14 +0000 [r196622] Sean Bright <sean@malleable.com>
+
+ * cdr/cdr_manager.c: Use a properly allocated channel for
+ substitution in cdr_manager.
+
+2009-05-24 16:17 +0000 [r196554-196585] Eliel C. Sardanons <eliels@gmail.com>
+
+ * res/res_agi.c: Move AGI static documentation to the new AstXML
+ form. Move AGI commands documentation to XML docs: 'set priority'
+ 'set variable' 'stream file' 'control stream file' 'tdd mode'
+ 'verbose' 'wait for digit' 'speech create' 'speech set' 'speech
+ destroy' 'speech load grammar' 'speech unload grammar' 'speech
+ activate grammar' 'speech deactivate grammar' 'speech recognize'
+
+ * res/res_agi.c: Move static AGI commands documentation to XML.
+ Move AGI commands ('say datetime', 'send image', 'send text',
+ 'set autohangup', 'set callerid', 'set context', 'set extension')
+ documentation to the AstXML form.
+
+2009-05-23 15:16 +0000 [r196520] Sean Bright <sean@malleable.com>
+
+ * cdr/cdr_custom.c: Fix errors in cdr_custom that cause reference
+ errors when non-CDR variable substitution is done. cdr_custom was
+ creating a ast_channel struct directly and passing it into the
+ core for variable substition. This was fine as long as the format
+ string contained only calls to the CDR() function. Doing
+ something like ${EPOCH} on the other hand tried to lock the
+ channel, which would fail and throw an error because the passed
+ channel hadn't been allocated as an ao2 object. So now we create
+ the dummy channel with ast_channel_alloc, and everything works as
+ expected.
+
+2009-05-23 13:31 +0000 [r196488] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/cli.h: Correct example for CLI autocompletion
+ (generation) Reported by Atis on #asterisk-dev
+
+2009-05-23 04:27 +0000 [r196456] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c: set MFCR2_CATEGORY just when starting the
+ pbx
+
+2009-05-22 21:11 +0000 [r196417] Sean Bright <sean@malleable.com>
+
+ * main/asterisk.c: Call ast_stun_init() when we're initializing to
+ get the 'stun debug set' commands.
+
+2009-05-22 21:09 +0000 [r196416] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: SIP set outbound
+ transport type from Registration In sip.conf the transport option
+ allows for the configuration of what transport types (udp, tcp,
+ and tls) a peer will accept, but only the first type listed was
+ used for outbound connections. This patch changes this. Now the
+ default transport type is only used until the peer registers.
+ When registration takes place the transport type is parsed out of
+ the Contact header. If the Contact header's transport type is
+ equal to one that the peer supports, the peer's default transport
+ type for outbound connections is set to match the Contact
+ header's type. If the Contact header's transport type is not
+ present, then the peer's default transport type is set to match
+ the one the peer registered with. When a peer unregisters or the
+ registration expires, the default transport type for that peer is
+ reset. (closes issue #12282) Reported by: rjain Patches:
+ reg_patch_1.diff uploaded by dvossel (license 671) Tested by:
+ dvossel (closes issue #14727) Reported by: pj Patches:
+ reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj,
+ dvossel Review: https://reviewboard.asterisk.org/r/249/
+
+2009-05-22 20:01 +0000 [r196381] Sean Bright <sean@malleable.com>
+
+ * channels/chan_gtalk.c: Don't crash if an RTP instance can't be
+ created. This could occur when an invalid bindaddr was specified
+ in gtalk.conf.
+
+2009-05-22 19:38 +0000 [r196308-196377] Eliel C. Sardanons <eliels@gmail.com>
+
+ * apps/app_minivm.c: Unregister every registered application by
+ MiniVM. The MinivmMWI application was not being unregistered on
+ unload and we were not able to load again the module or reload
+ it. (closes issue #15174) Reported by: junky Patches:
+ unregister_minivm_mwi.diff uploaded by junky (license 177)
+
+ * res/res_agi.c: Moved static documentation to the AstXML form.
+ Moved AGI commands static documentation to XML docs ('say alpha',
+ 'say digits', 'say number', 'say phonetic', 'say date' and 'say
+ time').
+
+ * main/pbx.c, channels/chan_sip.c, apps/app_meetme.c,
+ channels/chan_agent.c, apps/app_queue.c, channels/chan_iax2.c,
+ include/asterisk/manager.h, channels/chan_dahdi.c,
+ main/manager.c, channels/chan_skinny.c, main/features.c,
+ res/res_agi.c, include/asterisk/xmldoc.h, include/asterisk/pbx.h,
+ apps/app_senddtmf.c, doc/appdocsxml.dtd, main/db.c,
+ main/xmldoc.c, apps/app_voicemail.c: Implement a new element in
+ AstXML for AMI actions documentation. A new xml element was
+ created to manage the AMI actions documentation, using AstXML. To
+ register a manager action using XML documentation it is now
+ possible using ast_manager_register_xml(). The CLI command
+ 'manager show command' can be used to show the parsed
+ documentation. Example manager xml documentation: <manager
+ name="ami action name" language="en_US"> <synopsis> AMI action
+ synopsis. </synopsis> <syntax> <xi:include
+ xpointer="xpointer(...)" /> <-- for ActionID <parameter
+ name="header1" required="true"> <para>Description</para>
+ </parameter> ... </syntax> <description> <para>AMI action
+ description</para> </description> <see-also> ... </see-also>
+ </manager>
+
+2009-05-22 16:53 +0000 [r196272] Tilghman Lesher <tlesher@digium.com>
+
+ * main/astmm.c: Two more minor fixes due to constification
+
+2009-05-22 16:51 +0000 [r196270] Sean Bright <sean@malleable.com>
+
+ * res/res_agi.c: Fix res_agi compilation after the const-ify the
+ world merge. Since we are dealing with a 'const char * const'
+ now, we have to create a temporary copy of the string to work on
+ rather than the original. Fix inspired by reporter. Reviewed by
+ everyone-and-their-mother in #asterisk-dev. (closes issue #15184)
+ Reported by: andrew
+
+2009-05-22 16:50 +0000 [r196268] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: s/it's/its/
+
+2009-05-22 16:20 +0000 [r196246] Russell Bryant <russell@digium.com>
+
+ * channels/chan_dahdi.c: resolve compiler warning
+
+2009-05-22 16:10 +0000 [r196227] Sean Bright <sean@malleable.com>
+
+ * channels/chan_dahdi.c, main/pbx.c, res/res_jabber.c,
+ res/res_monitor.c: Fix build under dev mode and remove some casts
+ that are no longer necessary as a result of the const-ify the
+ world patch.
+
+2009-05-22 15:07 +0000 [r196187-196188] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_mp3.c: Fix constify the world compile problem.
+
+ * channels/chan_misdn.c: Make chan_misdn compile.
+
+2009-05-22 13:56 +0000 [r196117] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 196116 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r196116 | file | 2009-05-22 10:54:17 -0300 (Fri, 22 May
+ 2009) | 5 lines Fix a bug where using immediate with mISDN caused
+ a cause code of 16 to get sent back instead of 1 if the 's'
+ extension did not exist. (closes issue #12286) Reported by:
+ lmamane ........
+
+2009-05-22 13:34 +0000 [r196114] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/pbx.c: Avoid using prototypes when not necessary (it is
+ already defined in the header file). Make log_match_char_tree()
+ static to main/pbx.c (only used there).
+
+2009-05-21 21:13 +0000 [r196072] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_dahdibarge.c, main/frame.c, apps/app_record.c,
+ apps/app_playtones.c, funcs/func_strings.c,
+ include/asterisk/extconf.h, apps/app_alarmreceiver.c,
+ apps/app_chanisavail.c, apps/app_ices.c, apps/app_exec.c,
+ channels/chan_iax2.c, main/astobj2.c, channels/chan_dahdi.c,
+ channels/chan_skinny.c, apps/app_dumpchan.c, pbx/pbx_ael.c,
+ main/pbx.c, channels/vcodecs.c, apps/app_softhangup.c,
+ apps/app_morsecode.c, apps/app_talkdetect.c,
+ channels/iax2-parser.c, apps/app_db.c, apps/app_speech_utils.c,
+ apps/app_sendtext.c, pbx/pbx_config.c, apps/app_mixmonitor.c,
+ main/asterisk.c, res/res_odbc.c, apps/app_voicemail.c,
+ apps/app_dictate.c, apps/app_authenticate.c,
+ apps/app_readexten.c, apps/app_userevent.c, res/res_jabber.c,
+ include/asterisk/abstract_jb.h, main/channel.c,
+ apps/app_setcallerid.c, apps/app_osplookup.c, funcs/func_odbc.c,
+ apps/app_mp3.c, apps/app_minivm.c, apps/app_directory.c,
+ apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c,
+ apps/app_read.c, channels/chan_sip.c,
+ include/asterisk/taskprocessor.h, include/asterisk/cli.h,
+ apps/app_originate.c, utils/conf2ael.c,
+ apps/app_channelredirect.c, apps/app_forkcdr.c,
+ main/abstract_jb.c, channels/misdn/chan_misdn_config.h,
+ apps/app_sms.c, utils/extconf.c, funcs/func_devstate.c,
+ apps/app_stack.c, apps/app_verbose.c, main/dsp.c, main/udptl.c,
+ include/asterisk/agi.h, cdr/cdr_sqlite3_custom.c,
+ apps/app_readfile.c, apps/app_sayunixtime.c, apps/app_test.c,
+ include/asterisk/speech.h, cdr/cdr_adaptive_odbc.c,
+ apps/app_image.c, main/taskprocessor.c, main/loader.c,
+ main/cli.c, apps/app_skel.c, include/asterisk/module.h,
+ main/features.c, apps/app_amd.c, channels/chan_alsa.c,
+ apps/app_url.c, apps/app_externalivr.c, formats/format_gsm.c,
+ apps/app_milliwatt.c, res/res_speech.c, main/ast_expr2.fl,
+ apps/app_dial.c, include/asterisk/utils.h, apps/app_page.c,
+ apps/app_privacy.c, apps/app_fax.c, apps/app_echo.c,
+ channels/chan_agent.c, apps/app_dahdiras.c, apps/app_disa.c,
+ pbx/dundi-parser.c, apps/app_transfer.c, res/res_monitor.c,
+ apps/app_playback.c, include/asterisk/app.h,
+ channels/chan_misdn.c, apps/app_waitforring.c,
+ include/asterisk/image.h, apps/app_macro.c,
+ apps/app_zapateller.c, apps/app_chanspy.c, apps/app_cdr.c,
+ channels/chan_unistim.c, apps/app_meetme.c, main/utils.c,
+ res/res_musiconhold.c, apps/app_followme.c,
+ channels/misdn_config.c, apps/app_controlplayback.c, main/ulaw.c,
+ main/cdr.c, main/manager.c, channels/console_gui.c,
+ cdr/cdr_sqlite.c, res/res_agi.c, main/app.c,
+ apps/app_confbridge.c, main/image.c, apps/app_ivrdemo.c,
+ apps/app_parkandannounce.c, res/res_clioriginate.c,
+ apps/app_jack.c, apps/app_while.c, res/res_rtp_asterisk.c,
+ apps/app_nbscat.c, apps/app_festival.c, res/res_limit.c,
+ apps/app_waitforsilence.c, apps/app_waituntil.c,
+ channels/chan_console.c, apps/app_queue.c, apps/app_system.c,
+ apps/app_getcpeid.c, channels/chan_oss.c,
+ include/asterisk/features.h, apps/app_flash.c,
+ apps/app_directed_pickup.c, channels/chan_nbs.c,
+ include/asterisk/strings.h, include/asterisk/pbx.h,
+ apps/app_senddtmf.c: Const-ify the world (or at least a good part
+ of it) This patch adds 'const' tags to a number of Asterisk APIs
+ where they are appropriate (where the API already demanded that
+ the function argument not be modified, but the compiler was not
+ informed of that fact). The list includes: - CLI command handlers
+ - CLI command handler arguments - AGI command handlers - AGI
+ command handler arguments - Dialplan application handler
+ arguments - Speech engine API function arguments In addition,
+ various file-scope and function-scope constant arrays got 'const'
+ and/or 'static' qualifiers where they were missing. Review:
+ https://reviewboard.asterisk.org/r/251/
+
+2009-05-21 19:11 +0000 [r195995] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 195991 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21
+ May 2009) | 14 lines Sign problem calculating timestamp for iax
+ frame leads to no audio on the receiving peer. There are rare
+ cases in which a frame's delivery timestamp is slightly less than
+ the iax2_pvt's offset. This causes the pvt's timestamp to be a
+ small negative number, but since the timestamp value is unsigned
+ it looks like a huge positive number. This patch checks for this
+ negative case and sets the ms to zero. A similar check is already
+ done right below this one in the 'else' statement. (closes issue
+ #15032) Reported by: guillecabeza Patches:
+ chan_iax2.c.patch_timestamp uploaded by guillecabeza (license
+ 380) Tested by: guillecabeza (closes issue #14216) Reported by:
+ Andrey Sofronov ........
+
+2009-05-21 19:06 +0000 [r195992] Mark Michelson <mmichelson@digium.com>
+
+ * main/features.c: Pass connected line updates along during a
+ bridge.
+
+2009-05-21 17:15 +0000 [r195949] Sean Bright <sean@malleable.com>
+
+ * configs/cdr_custom.conf.sample: Rework the cdr_custom.conf.sample
+ header a bit to reflect the changes in functionality (allowing
+ multiple mappings).
+
+2009-05-21 15:33 +0000 [r195882] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/cdr.c, /, include/asterisk/cdr.h: Merged revisions 195881
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May
+ 2009) | 13 lines This commit prevents cdr records with
+ AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated
+ in certain cases. This is accomplished by adding two functions to
+ update the answer time and disposition of calls that checks for
+ the proper lock flags. These functions are used in the
+ ast_bridge_call() function so that ForkCDR(A) calls are
+ respected. This patch also modifies the way ast_bridge_call()
+ chooses the cdr record to base the bridged_cdr on. Previously the
+ first unlocked cdr record would be chosen, now instead the first
+ cdr record is chosen and forked cdr records are moved to the
+ bridge_cdr. This allows the original cdr record and any forked
+ cdr records to be properly updated with answer and end times.
+ (closes issue #13797) Reported by: sh0t Tested by: sh0t (closes
+ issue #14744) Reported by: deepesh ........
+
+2009-05-20 23:30 +0000 [r195839] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_stack.c: If a variable had a blank value upon the
+ initial setting, then it would do nothing. Identified by Dmitry
+ Andrianov via private email, fixed by me.
+
+2009-05-20 20:45 +0000 [r195763-195798] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Get rid of some duplicated code and correct
+ a connected line error. When receiving a 200 OK response to an
+ INVITE, it was possible to transmit two connected line updates
+ instead of a single one. Furthermore, the second did not have the
+ proper information present. Now the two have been combined into a
+ single update and the correct information is presented.
+
+ * apps/app_dial.c: Plug a memory leak in app_dial. Since we may
+ have copied connected line info into the chanlist struct prior to
+ placing an outbound call, we need to be sure to free the
+ allocated data when we hang the call up.
+
+2009-05-20 17:33 +0000 [r195636-195698] Joshua Colp <jcolp@digium.com>
+
+ * /, main/features.c: Merged revisions 195688 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195688 | file | 2009-05-20 14:30:25 -0300 (Wed, 20 May 2009) | 5
+ lines Fix some code that wrongly assumed a pointer would always
+ be non-NULL when dealing with CDRs after a bridge. (closes issue
+ #15079) Reported by: barryf ........
+
+ * /, apps/app_meetme.c: Merged revisions 195635 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195635 | file | 2009-05-20 14:14:00 -0300 (Wed, 20 May 2009) | 5
+ lines Fix a bug where the MeetMe option 'D' did not actually
+ prompt for the pin. (closes issue #15050) Reported by: pmhaddad
+ ........
+
+2009-05-19 20:59 +0000 [r195589] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: Add basic support
+ for handling connected line-related UPDATE requests. SIP purists
+ may want to look the other way... When COLP/CONP support for SIP
+ was committed, there was a condition under which Asterisk may
+ transmit a SIP UPDATE in order to communicate the change in
+ connected line information. The issue here is that while we could
+ send a SIP UPDATE message, we were not prepared to receive such
+ an UPDATE and would always responde with a 501 when we received
+ an UPDATE. The situation was a bit rough. We really want to be
+ able to receive UPDATEs having to do with connected line changes,
+ but the amount of effort involved in properly supporting RFC 3311
+ was staggering. This commit represents a compromise. First, it
+ was decided that it is important to only send a SIP UPDATE to an
+ endpoint that is able to handle one. So, now we have added
+ parsing of the Allow header into SIP. We store the allowed
+ methods on SIP peers so that when we communicate with them, we
+ already will know what we can and cannot send to them. We will
+ parse the peer's allowed methods when he registers with us. If
+ the peer is not the type to register with us, but the qualify
+ option is enabled, then we will use the response to the OPTIONS
+ request we send the peer to determine the peer's allowed methods.
+ When the peer's registration expires, or when qualify deems the
+ peer to be unreachable, we clear the allowed methods from the
+ peer. For an actual call, we will copy the peer's allowed methods
+ to the sip_pvt representing the call leg. If we are communicating
+ with an endpoint which is not a peer, then we will just parse the
+ Allow header from the first message we receive during the call
+ and store the information in the sip_pvt. If, during
+ communication with a peer, we receive a 501 response, then we
+ will make sure to save the fact that we cannot use that method
+ when communicating with that peer. Now, with all that
+ infrastructure in place, the only actual place we use this
+ information currently is when attempting to send a connected line
+ change using an UPDATE request. If we cannot send the change
+ immediately using an UPDATE, we will set the SIP_NEEDREINVITE
+ flag so that we can send a REINVITE as soon as it is allowed. The
+ second part of the changes here is for Asterisk to accept UPDATE
+ requests that have connected line changes. Since we are not fully
+ supporting RFC 3311, Asterisk will NOT place the UPDATE method in
+ Allow headers it sends. Instead, if you are communicating with
+ what you know to be another Asterisk box, you may set the
+ rpid_update parameter in sip.conf so that we will send UPDATEs to
+ that Asterisk box. When we send a connected line update, we set a
+ custom header called "X-Asterisk-rpid-update." On the receiving
+ end, if Asterisk receives an UPDATE that does not have the
+ "X-Asterisk-rpid-update" header present, then Asterisk will
+ respond with a 501 since media-changing UPDATEs are not
+ supported. We should never get such UPDATEs, since as was stated
+ earlier, Asterisk does not put UPDATE in its Allow header. If the
+ custom header is present in the received UPDATE, though, then we
+ will check the incoming request for connected line updates and
+ queue the update on the channel where the change occurred.
+ ABE-1840 ABE-1822
+
+2009-05-19 20:16 +0000 [r195521] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 195520 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r195520 | tilghman | 2009-05-19 15:12:20 -0500 (Tue, 19
+ May 2009) | 7 lines Ensure thread keys are initialized before
+ attempting to access them. (closes issue #14889) Reported by:
+ jaroth Patches: app_voicemail.c.patch uploaded by msirota
+ (license 758) Tested by: msirota, BlargMaN ........
+
+2009-05-19 14:43 +0000 [r195449] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 195448 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195448 | file | 2009-05-19 11:41:45 -0300 (Tue, 19 May 2009) | 7
+ lines Fix a bug where direct RTP setup would partially occur even
+ when disabled if the calling channel was answered. (issue #13545)
+ Reported by: davidw (issue #14244) Reported by: mbnwa ........
+
+2009-05-18 20:52 +0000 [r195370] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_smdi.c, /, include/asterisk/monitor.h, apps/app_queue.c,
+ include/asterisk/smdi.h, res/res_monitor.c, apps/app_voicemail.c:
+ Recorded merge of revisions 195366 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195366 | tilghman | 2009-05-18 15:24:13 -0500 (Mon, 18 May 2009)
+ | 8 lines Add a similar dependency on SMDI for voicemail as
+ already exists for ADSI. (closes issue #14846) Reported by: pj
+ Patches: 20090413__bug14846__1.4.diff.txt uploaded by tilghman
+ (license 14) 20090507__issue14846__1.6.0.diff.txt uploaded by
+ tilghman (license 14) 20090507__issue14846__1.6.1.diff.txt
+ uploaded by tilghman (license 14) ........
+
+2009-05-18 20:49 +0000 [r195365-195369] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/manager.c: Fix the CLI command 'manager show command'
+ documentation and functionality. The CLI command 'manager show
+ command' supports passing multiple action names in the same line,
+ but it was not allowing that because of a incorrect check in the
+ argumentes counter. Also the documentation was updated to show
+ that this usage of the command is possible.
+
+ * main/manager.c: Rollback commit 195367. The CLI command 'manager
+ show command' supports passing multiple AMI actions at a time.
+ The issue with this command was in another place.
+
+ * main/manager.c: Avoid autocompleting passed the action name
+ argument in the CLI command. When running the autocomplete of the
+ CLI command 'manager show command <action>' it was autocompleting
+ everything else after the <action> argument, giving an error,
+ because this command doesn't support multiple AMI action names at
+ a time.
+
+ * res/res_agi.c: Move AGI documentation from static to the XML
+ form. Move the AGI commands 'receive text', 'receive char' and
+ 'record' static documentation to XML docs.
+
+2009-05-18 19:17 +0000 [r195320] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c: Move the spawn of astcanary down, until after
+ the call to daemon(3). This avoids possible conflicts with the
+ internal implementation of daemon(3). (closes issue #15093)
+ Reported by: tzafrir Patches: 20090513__issue15093__2.diff.txt
+ uploaded by tilghman (license 14) Tested by: tzafrir
+
+2009-05-18 18:58 +0000 [r195316] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_externalivr.c: Fix externalivr's setvariable command so
+ that it properly sets multiple variables. The command had a for
+ loop that was guaranteed to only execute once since the
+ continuation operation of the loop would set the input buffer
+ NULL. I rewrote the loop so that its operation was more obvious,
+ and it would set multiple variables correctly. I also reduced
+ stack space required for the function, constified the input
+ string, and modified the function so that it would not modify the
+ input string while I was at it. (closes issue #15114) Reported
+ by: chris-mac Patches: 15114.patch uploaded by mmichelson
+ (license 60) Tested by: chris-mac
+
+2009-05-18 17:08 +0000 [r195279] Sean Bright <sean@malleable.com>
+
+ * cdr/cdr_custom.c: Remove some unused code.
+
+2009-05-18 16:29 +0000 [r195266] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: The facilityenable parameter does not have
+ anything to do with pritimer parameters.
+
+2009-05-18 15:55 +0000 [r195210] Sean Bright <sean@malleable.com>
+
+ * cdr/cdr_custom.c: Const-ify a string, fix a log message, and use
+ the correct signature for the load_module function.
+
+2009-05-18 15:53 +0000 [r195207] Joshua Colp <jcolp@digium.com>
+
+ * main/frame.c, /: Merged revisions 195206 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195206 | file | 2009-05-18 12:51:22 -0300 (Mon, 18 May 2009) | 7
+ lines Fix a typo which caused loss of audio when using G729 in
+ some scenarios with a smoother present. (closes issue #15105)
+ Reported by: bamby Patches: process-vad-correctly.diff uploaded
+ by bamby (license 430) ........
+
+2009-05-18 14:54 +0000 [r195165] Sean Bright <sean@malleable.com>
+
+ * configs/cdr_custom.conf.sample, CHANGES, cdr/cdr_custom.c: Allow
+ cdr_custom to write to multiple files instead of just one. Up to
+ now, cdr_custom would only accept a single filename/format from
+ cdr_custom.conf. This change allows you to specify multiple
+ filename & format directives.
+
+2009-05-18 14:45 +0000 [r195162] Eliel C. Sardanons <eliels@gmail.com>
+
+ * apps/app_dial.c, main/pbx.c, apps/app_macro.c: Warn about the use
+ of the application WaitExten() within a Macro(). Update
+ applications documentation to warn the user about the use of the
+ WaitExten() application within a Macro(). Recommend the use of
+ Read() instead. (closes issue #14444) Reported by: ewieling
+
+2009-05-18 13:56 +0000 [r195089-195096] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp_engine.c, /: Merged revisions 195095 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195095 | file | 2009-05-18 10:53:39 -0300 (Mon, 18 May 2009) | 5
+ lines Fix a bug where the codecs of the called party leg were not
+ properly sent back to the caller call leg when reinvited. (closes
+ issue #13569) Reported by: bkw918 ........
+
+ * channels/chan_sip.c: Fix a bug where specifying an empty
+ outboundproxy would cause packets to get sent to ourself. (closes
+ issue #15106) Reported by: timeshell
+
+2009-05-18 13:30 +0000 [r195075] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/xml.c: Do not avoid loading the XML documentation if not
+ XInclude substitution is done.
+
+2009-05-18 12:59 +0000 [r195021] Russell Bryant <russell@digium.com>
+
+ * /: Recorded merge of revisions 195020 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195020 | russell | 2009-05-18 07:57:46 -0500 (Mon, 18 May 2009)
+ | 5 lines Don't try to unlock a bogus channel. (closes issue
+ #15144) Reported by: cristiandimache ........
+
+2009-05-16 20:01 +0000 [r194945-194982] Eliel C. Sardanons <eliels@gmail.com>
+
+ * Makefile, main/xml.c, doc/appdocsxml.dtd: Allow to include
+ sections of other parts of the xml documentation. Avoid
+ duplicating xml documentation by allowing to include other parts
+ of the xml documentation using XInclude. Example: <xi:include
+ xpointer="xpointer(/docs/function[@name='CHANNEL']/synopsis)" />
+ (Insert this line to include the synopsis of the CHANNEL function
+ xml documentation). It is also possible to include documentation
+ from other files in the 'documentation/' directory using the
+ href="" attribute inside a xinclude element. (closes issue
+ #15107) Reported by: lmadsen (issue #14444) Reported by: ewieling
+
+ * main/pbx.c: Fix a missing unlock in case of error, and a missing
+ free(). Always free the allocated memory for a string field,
+ because we are always using it (not only when xmldocs are
+ enabled). Also if there is an error allocating memory for the
+ string field remember to unlock the list of registered
+ applications, before returning.
+
+2009-05-15 22:44 +0000 [r194833-194874] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 194873 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r194873 | dvossel | 2009-05-15 17:43:13 -0500 (Fri, 15
+ May 2009) | 17 lines IAX2 REGAUTH loop IAX was not sending REGREJ
+ to terminate invalid registrations. Instead it sent another
+ REGAUTH if the authentication challenge failed. This caused a
+ loop of REGREQ and REGAUTH frames. (Related to Security fix
+ AST-2009-001) (closes issue #14867) Reported by: aragon Tested
+ by: dvossel (closes issue #14717) Reported by: mobeck Patches:
+ regauth_loop_update_patch.diff uploaded by dvossel (license 671)
+ Tested by: dvossel ........
+
+ * channels/iax2-parser.h, /, channels/iax2.h, channels/chan_iax2.c,
+ channels/iax2-parser.c: Merged revisions 194557,194685 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r194557 | dvossel | 2009-05-14 17:59:43 -0500 (Thu, 14 May 2009)
+ | 10 lines IAX2 "Ghost" Channels There is a bug tracker issue
+ where people are reporting "Ghost" channels in their 'iax2 show
+ channels' output. The confusion is caused by channels being
+ listed as "(NONE)" with format "unknown". These are not channels
+ of coarse. They are usually just pending registration or poke
+ requests, but it is confusing output. To help make sense of this
+ I have added two columns to 'iax2 show channels'. One shows the
+ first message which started the transaction, and the second shows
+ the last message sent by either side of the call. This helps
+ diagnose why the entry exists and why it may not go away. (closes
+ issue #14207) Reported by: clive18 Review:
+ https://reviewboard.asterisk.org/r/246/ ........ r194685 |
+ dvossel | 2009-05-15 10:40:37 -0500 (Fri, 15 May 2009) | 6 lines
+ Update to previous IAX2 "Ghost" Channels patch. Fixed some
+ comments made on reviewboard for the previous patch. (issue
+ #14207) ........
+
+2009-05-15 18:43 +0000 [r194714-194765] Russell Bryant <russell@digium.com>
+
+ * /, configs/logger.conf.sample: Merged revisions 194764 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r194764 | russell | 2009-05-15 13:43:18 -0500 (Fri, 15 May 2009)
+ | 2 lines Fix some spelling fail. ........
+
+ * codecs/g722/g722_encode.c, codecs/g722/g722_decode.c: Shuttle
+ some bits around to address some gain issues with G.722. (closes
+ AST-209)
+
+ * codecs/Makefile, codecs/g722/Makefile (removed): Further simplify
+ codec_g722 build.
+
+ * codecs/Makefile: Actually force running make for g722.
+
+2009-05-15 13:43 +0000 [r194649] Michiel van Baak <michiel@vanbaak.info>
+
+ * CREDITS: add eliel
+
+2009-05-15 13:23 +0000 [r194635] Eliel C. Sardanons <eliels@gmail.com>
+
+ * doc/appdocsxml.dtd, main/xmldoc.c: Allow to specify an enumlist
+ inside an enum. It was not possible to use an enumlist inside an
+ enum: <enumlist> <enum name="aa"> <enumlist> ... </enumlist>
+ </enum> </enumlist> Now we will be able to insert as many levels
+ as we want. (closes issue #15112) Reported by: lmadsen
+
+2009-05-15 13:13 +0000 [r194520-194610] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/logger.h, tests/test_logger.c (added),
+ main/logger.c: Add ability for modules to dynamically register
+ logger levels This patch adds the ability for modules to
+ dynamically create logger levels for their own use; these are
+ named levels just like the built-in levels, and can be directed
+ to any destination that the logger can send any level to, by
+ including their names in logger.conf. Review:
+ https://reviewboard.asterisk.org/r/244/
+
+ * /: Merged revisions 194509 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r194509 | kpfleming | 2009-05-14 17:23:49 -0500 (Thu, 14 May
+ 2009) | 1 line Update URL to Reviewboard ........
+
+2009-05-14 22:20 +0000 [r194496] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 194484 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r194484 | mmichelson | 2009-05-14 17:17:55 -0500 (Thu, 14 May
+ 2009) | 24 lines Fix a race condition where a reinvite could
+ trigger a 482 response. The loop detection/spiral detection code
+ in chan_sip used the owner channel's state as a criterion for
+ determining if the incoming INVITE is a looped request. The
+ problem with this is that the INVITE-handling code happens in a
+ different thread than the thread that marks the owner channel as
+ being up. As a result, if a reinvite were to come in very
+ quickly, say from another Asterisk on the same LAN, it was
+ possible for the reinvite to arrive before the owner channel had
+ been set to the up state. This patch corrects the problem by
+ using the invitestate of the sip_pvt instead, since that can be
+ guaranteed to be set correctly by the time the reinvite arrives.
+ Since there is a switch statement further in the INVITE-handling
+ code, the AST_STATE_RINGING state also checks the invitestate of
+ the sip_pvt in case we should actually be treating the channel as
+ if it were up already. (closes issue #12215) Reported by: jpyle
+ Patches: 12215_confirmed.patch uploaded by mmichelson (license
+ 60) Tested by: lmadsen ........
+
+2009-05-14 22:03 +0000 [r194479] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.h, channels/chan_misdn.c,
+ channels/misdn/chan_misdn_config.h,
+ channels/misdn/isdn_msg_parser.c, configs/misdn.conf.sample,
+ CHANGES, channels/misdn/isdn_lib.c, channels/misdn_config.c: Add
+ outgoing_colp misdn.conf port parameter. Select what to do with
+ outgoing COLP information on this port. 0 - Send out COLP
+ information unaltered. (default) 1 - Force COLP to restricted on
+ all outgoing COLP information. 2 - Do not send COLP information.
+ outgoing_colp=0 Also fixed sending the EctInform message so it
+ always has the required redirectionNumber parameter when the
+ status is active. JIRA ABE-1853
+
+2009-05-14 21:24 +0000 [r194477] Russell Bryant <russell@digium.com>
+
+ * main/features.c: Fix a typo where an equality check should be an
+ assignment. (closes issue #15103) Reported by: lmsteffan Patches:
+ transfer_crash.patch uploaded by lmsteffan (license 779)
+
+2009-05-14 17:05 +0000 [r194434] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_meetme.c: Fix a bug where the 'T' option to Meetme did
+ not work. (closes issue #15031) Reported by: Stochastic (closes
+ issue #13801) Reported by: justdave
+
+2009-05-14 16:22 +0000 [r194430] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: If the timing ended on a zero, then we would loop
+ forever. (closes issue #14983) Reported by: teox Patches:
+ 20090513__issue14983.diff.txt uploaded by tilghman (license 14)
+ Tested by: teox
+
+2009-05-13 15:02 +0000 [r194283] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/manager.c: Do not lock the 'sessions' container, lock the
+ allocated 'session'. There was a typo in the structure being
+ locked, and we were locking the 'sessions' container instead of
+ the 'session' structure thar we are modifying. Reported by
+ seanbright on #asterisk-dev, thanks!
+
+2009-05-13 13:39 +0000 [r194209] Joshua Colp <jcolp@digium.com>
+
+ * res/res_rtp_asterisk.c, /: Merged revisions 194208 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r194208 | file | 2009-05-13 10:38:01 -0300 (Wed, 13 May
+ 2009) | 11 lines Fix RFC2833 issues with DTMF getting duplicated
+ and with duration wrapping over. (closes issue #14815) Reported
+ by: geoff2010 Patches: v1-14815.patch uploaded by dimas (license
+ 88) Tested by: geoff2010, file, dimas, ZX81, moliveras (closes
+ issue #14460) Reported by: moliveras Tested by: moliveras
+ ........
+
+2009-05-13 00:52 +0000 [r194101-194138] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 194137 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r194137 | tilghman | 2009-05-12 19:52:03 -0500 (Tue, 12 May 2009)
+ | 7 lines Fix logic for how to proceed with a single digit
+ extension. (closes issue #15091) Reported by: andrew Patches:
+ 20090512__issue15091.diff.txt uploaded by tilghman (license 14)
+ Tested by: andrew ........
+
+ * main/pbx.c, main/logger.c: Two fixes found while debugging with
+ ast_backtrace(): 1) If MALLOC_DEBUG is used when concurrently
+ using ast_backtrace, the free() used in that routine will trigger
+ an error, because the memory was allocated internally to libc,
+ where we could not intercept that call to wrap it. Therefore,
+ it's not memory we knew about, and the free is reported as an
+ error. 2) Now that channels are objects, the old hack of
+ initializing a channel to all zeroes no longer works, since we
+ may try to call something like ast_channel_lock() within a
+ function on that reference. In that case, it's reported as an
+ error, because the pointer isn't an object reference.
+
+2009-05-12 22:49 +0000 [r194060] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/manager.c: Fix a crash when logging out from the AMI and
+ avoid astobj2 warning messages. When the user logout the session
+ was being destroyed twice and the file descriptor was being
+ closed twice. The sessions reference counter wasn't used in a
+ proper way. The 'mansession' structure was being treated as an
+ astobj2 and we were calling ao2_lock/ao2_unlock causing astobj2
+ report a warning message and not locking the structure. Also we
+ were using an ugly naming convention 'destroy_session',
+ 'session_destroy', 'free_session', ... all this "duplicated" code
+ was merged. (closes issue #14974) Reported by: pj Patches:
+ manager.diff2 uploaded by eliel (license 64) Tested by: dhubbard,
+ eliel, mnicholson (closes issue #15088) Reported by: eliel
+ Review: http://reviewboard.asterisk.org/r/248/
+
+2009-05-12 22:32 +0000 [r194057] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 194028 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r194028 | mnicholson | 2009-05-12 17:15:45 -0500 (Tue, 12 May
+ 2009) | 16 lines This change modifies app_queue to properly
+ generate CDR records in failure situations. This involves setting
+ a proper cdr disposition coresponding to the given failure
+ condition and ensuring the proper information is stored in the
+ cdr record. (closes issue #13691) Reported by: dferrer Tested by:
+ mnicholson (closes issue #13637) Reported by: atis Tested by:
+ atis ........
+
+2009-05-12 20:40 +0000 [r193956] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 193955 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r193955 | tilghman | 2009-05-12 15:39:21 -0500 (Tue, 12
+ May 2009) | 6 lines Avoid initializing routines if the
+ authentication fails. Fixes a crash (RR) issue. (closes issue
+ #14508) Reported by: tiziano Patches:
+ 20090221_2_wrongmailbox.diff.txt uploaded by tiziano (license
+ 377) ........
+
+2009-05-12 20:28 +0000 [r193954] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Update spiral support in trunk and 1.6.X to
+ match what is in 1.4. In 1.4, a SIP spiral is treated the same
+ way as a call forward. This works much better than what is
+ currently in trunk and 1.6.X. The code in trunk and 1.6.X did not
+ create a new call to the recipient of the spiral, instead trying
+ to continue the same call. In addition to just being plain wrong,
+ this also had the side effect of only being able to spiral calls
+ to other SIP channels. With this in place, as long as call
+ forwards are honored, SIP spirals will work properly. This means
+ that it will work for outbound calls made by the Queue, Dial, and
+ Page applications. For originated calls and spool calls, however,
+ the spiral will not work properly until a generic call forward
+ mechanism is introduced into Asterisk. (relates to issue #13630)
+
+2009-05-12 17:29 +0000 [r193870] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Convert a THREADSTORAGE object into a
+ simple malloc'd object (as suggested by Russell on -dev)
+
+2009-05-12 13:59 +0000 [r193832] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_dial.c, main/pbx.c, apps/app_meetme.c, apps/app_page.c,
+ main/devicestate.c, apps/app_queue.c, apps/app_transfer.c,
+ apps/app_playback.c, apps/app_controlplayback.c, main/term.c,
+ channels/chan_dahdi.c, channels/chan_misdn.c, funcs/func_curl.c,
+ apps/app_sendtext.c, apps/app_directed_pickup.c,
+ channels/console_gui.c, main/features.c, apps/app_confbridge.c,
+ apps/app_externalivr.c, apps/app_chanspy.c,
+ apps/app_mixmonitor.c, apps/app_stack.c, res/res_odbc.c,
+ apps/app_voicemail.c: add 'const' qualifiers in various places
+ where they should have been
+
+2009-05-11 23:04 +0000 [r193756-193757] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Found and fixed a memory leak
+
+ * /: Recorded merge of revisions 193755 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193755 | tilghman | 2009-05-11 17:48:20 -0500 (Mon, 11 May 2009)
+ | 18 lines Move 300 bytes around on the stack, to make more room
+ for an extension buffer. This allows more concurrent extensions
+ to be copied for a single voicemail, without creating a
+ possibility of upsetting existing users, where a dialplan could
+ run out of stack space where it had run fine before.
+ Alternatively, we could have allocated off the heap, but that is
+ a larger change and would have increased the chance for
+ instability introduced by this change. This is really solved
+ starting in 1.6.0.11, as the use of an ast_str buffer allows an
+ unlimited number of extensions (up to available memory). We
+ additionally create a new warning message when the buffer length
+ is exceeded, permitting administrators to see an issue after the
+ fact, whereas previously the list was silently truncated. (closes
+ issue #14739) Reported by: p_lindheimer Patches:
+ 20090417__bug14739.diff.txt uploaded by tilghman (license 14)
+ Tested by: p_lindheimer ........
+
+2009-05-11 22:04 +0000 [r193718] Russell Bryant <russell@digium.com>
+
+ * res/res_timing_timerfd.c: Fix some timer state corruption. In
+ res_timer_timerfd, handle the case that set_rate gets called
+ while a timer is still in continuous mode. In this case, we want
+ to remember the configured rate, but not actually set it until
+ continuous mode has been disabled. Thanks to dvossel for finding
+ and helping to debug the problem. (closes issue #15080) Reported
+ by: dvossel Tested by: dvossel
+
+2009-05-11 19:32 +0000 [r193678] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Don't nullify an ast_str pointer. (closes
+ issue #15061) Reported by: alecdavis
+
+2009-05-11 19:11 +0000 [r193614] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 193613 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11
+ May 2009) | 12 lines Sent wrong message to clear a call we
+ started if the other end has not responed yet. In the state
+ MISDN_CALLING (i.e. SETUP was sent but no answer has arrived
+ yet), it is not allowed to clear the call with RELEASE_COMPLETE.
+ It must be cleared with DISCONNECT. A RELEASE_COMPLETE is only
+ allowed as an answer to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a,
+ 5.3.2.b) Patches: chan-misdn-ccstate7.patch uploaded by customer.
+ JIRA ABE-1862 ........
+
+2009-05-11 18:01 +0000 [r193545] Leif Madsen <lmadsen@digium.com>
+
+ * /, funcs/func_channel.c: Recorded merge of revisions 193544 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193544 | lmadsen | 2009-05-11 13:35:17 -0400 (Mon, 11 May 2009)
+ | 7 lines Document CHANNEL(transfercapability) in CLI
+ documentation. (issue #15073) Reported by: pkempgen Patches:
+ 20090511__issue15073.diff.txt uploaded by tilghman (license 14)
+ ........
+
+2009-05-10 17:07 +0000 [r193502] Joshua Colp <jcolp@digium.com>
+
+ * main/bridging.c: Fix a bug where receiving a control frame of
+ subclass -1 would cause certain channels to get hung up.
+
+2009-05-09 11:33 +0000 [r193459-193461] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/event.h: Minor documentation update for
+ ast_event_queue().
+
+ * main/channel.c: Declare private data as static.
+
+2009-05-08 20:32 +0000 [r193387] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: TCP not matching valid peer. find_peer()
+ does not find a valid peer when using pvt->recv as the
+ sockaddr_in argument. Because of the way TCP works, the port
+ number in pvt->recv is not what we're looking for at all. There
+ is currently only one place that find_peer searches for a peer
+ using the sockaddr_in argument. If the peer is not found after
+ using pvt->recv (works for UDP since the port number will be
+ correct), a temp sockaddr_in struct is made using the Contact
+ header in the sip_request. This has the correct port number in
+ it. Review: http://reviewboard.digium.com/r/236/
+
+2009-05-08 19:50 +0000 [r193349] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Reset the members' call counts when resetting
+ queue statistics. This helps to prevent odd scenarios where a
+ queue will claim to have taken 0 calls, but the members appear to
+ have taken a non-zero amount. (closes issue #15068) Reported by:
+ sum Patches: patchreset.patch uploaded by sum (license 766)
+ Tested by: sum
+
+2009-05-08 15:18 +0000 [r193274] Sean Bright <sean@malleable.com>
+
+ * funcs/func_devstate.c: Fix the spelling of UNAVAILABLE in
+ func_devstate CLI completion.
+
+2009-05-08 14:52 +0000 [r193263] David Vossel <dvossel@digium.com>
+
+ * /, channels/misdn_config.c: Merged revisions 193262 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r193262 | dvossel | 2009-05-08 09:51:09 -0500 (Fri, 08
+ May 2009) | 9 lines "misdn show config" segfaults asterisk, if no
+ MSN lists (closes issue #14976) Reported by: alecdavis Patches:
+ misdn_config.diff.txt uploaded by alecdavis (license 585) Tested
+ by: alecdavis, FabienToune ........
+
+2009-05-08 14:06 +0000 [r193194] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, main/logger.c, configs/logger.conf.sample: Merged revisions
+ 193193 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193193 | kpfleming | 2009-05-08 09:03:28 -0500 (Fri, 08 May
+ 2009) | 7 lines Make absolute paths for logger channels work
+ properly (Note: This is not a new feature, it was previously
+ undocumented and broken.) The Asterisk logger has a feature to
+ support absolute pathnames for logger channels, but the code
+ implementing the feature was broken. This has been fixed, and the
+ absolute path feature is now documented in the sample
+ logger.conf. ........
+
+2009-05-07 23:42 +0000 [r193120] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 193119 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193119 | tilghman | 2009-05-07 18:41:11 -0500 (Thu, 07 May 2009)
+ | 19 lines Fix Background within a Macro for FreePBX. If the
+ single digit DTMF is an extension in the specified context, then
+ go there and signal no DTMF. Otherwise, we should exit with that
+ DTMF. If we're in Macro, we'll exit and seek that DTMF as the
+ beginning of an extension in the Macro's calling context. If
+ we're not in Macro, then we'll simply seek that extension in the
+ calling context. Previously, someone complained about the
+ behavior as it related to the interior of a Gosub routine, and
+ the fix (#14011) inadvertently broke FreePBX (#14940). This
+ change should fix both of these situations, but with the possible
+ incompatibility that if a single digit extension does not exist
+ (but a longer extension COULD have matched), it would have
+ previously gone immediately to the "i" extension, but will now
+ need to wait for a timeout. (closes issue #14940) Reported by:
+ p_lindheimer Patches: 20090420__bug14940.diff.txt uploaded by
+ tilghman (license 14) Tested by: p_lindheimer ........
+
+2009-05-07 22:24 +0000 [r193077] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 193050 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r193050 | rmudgett | 2009-05-07 17:17:06 -0500 (Thu, 07
+ May 2009) | 5 lines Give a more helpful message when an incoming
+ call's dialed extension does not match. Added the dialed
+ extension and context to the chan_misdn messages warning that the
+ dialed number cannot be matched in the dialplan. ........
+
+2009-05-07 17:51 +0000 [r192933-193006] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_odbc.c: Second result should not contain data from the
+ first result. (closes issue #15039) Reported by: jims Patches:
+ 20090506__issue15039.diff.txt uploaded by tilghman (license 14)
+ Tested by: jims
+
+ * channels/chan_unistim.c: Send DTMF frame before playing back
+ audio. (closes issue #14858) Reported by: barryf Patches:
+ 20090507__bug14858.diff.txt uploaded by tilghman (license 14)
+
+ * /, channels/chan_sip.c: Merged revisions 192932 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r192932 | tilghman | 2009-05-07 11:29:08 -0500 (Thu, 07 May 2009)
+ | 10 lines Eliminate repetition of fullcontact during
+ reconstruction. If the fullcontact field appears in both the
+ sippeers and the sipregs table, then during reconstruction of the
+ field, it will otherwise be doubled. (closes issue #14754)
+ Reported by: Alexei Gradinari Patches:
+ 20090506__bug14754.diff.txt uploaded by tilghman (license 14)
+ Tested by: lmadsen ........
+
+2009-05-06 22:17 +0000 [r192853-192861] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 192858 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r192858 | jpeeler | 2009-05-06 17:15:19 -0500 (Wed, 06 May 2009)
+ | 10 lines Make ParkedCall application stop execution of the
+ dialplan after hang up Just changed park_exec to always return
+ non-zero. I really wasn't entirely sure at first if this was a
+ bug. Decided it was since it would be surprising when not using
+ ParkedCall in the dialplan to hang up and have dialplan execution
+ continue. (closes issue #14555) Reported by: francesco_r ........
+
+ * main/pbx.c: If no extension was found in the pattern tree, don't
+ crash.
+
+2009-05-06 17:38 +0000 [r192808] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: Fix a bug where a timer would be created
+ but not acknowledged. This scenario crept up if chan_iax2 was
+ loaded with no configuration file present. It would create a
+ timer and tell it to go at an interval but the thread that
+ normally acknowledges it would not be created because no
+ configuration file was present. The timer will now be closed if
+ no configuration file is present. (closes issue #15014) Reported
+ by: madkins
+
+2009-05-06 16:28 +0000 [r192772] Tilghman Lesher <tlesher@digium.com>
+
+ * main/say.c, doc/lang/urdu.ods (added): Add numbers in Urdu, the
+ national language of Pakistan (closes issue #15034) Reported by:
+ nasirq Patches: ast_say_number_full_ur-patch.c uploaded by nasirq
+ (license 772) urdu.ods uploaded by nasirq (license 772)
+
+2009-05-06 16:09 +0000 [r192634-192736] Joshua Colp <jcolp@digium.com>
+
+ * res/res_clialiases.c: Make the code that prevents an infinite
+ loop from happening into a case insensitive check. (thanks eliel)
+
+ * res/res_clialiases.c: Fix an infinite loop with tab completion of
+ CLI aliases that reference themselves. (closes issue #15020)
+ Reported by: junky
+
+ * /, channels/chan_sip.c: Merged revisions 192633 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r192633 | file | 2009-05-06 10:30:51 -0300 (Wed, 06 May 2009) | 7
+ lines Update some old logic to stop both begin and end DTMF
+ frames from reaching the core if rfc2833 is not enabled. (closes
+ issue #15036) Reported by: dimas Patches: v1-15036.patch uploaded
+ by dimas (license 88) ........
+
+2009-05-05 20:54 +0000 [r192590] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, channels/chan_sip.c, apps/app_directed_pickup.c,
+ main/features.c, apps/app_queue.c: Fixed crashes from issue8824
+ review board channel locking changes. The local struct
+ ast_party_connected_line connected_caller variable was
+ uninitialized when the copy function was called.
+
+2009-05-05 19:57 +0000 [r192525] Sean Bright <sean@malleable.com>
+
+ * /, static-http/astman.js: Merged revisions 192524 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r192524 | seanbright | 2009-05-05 15:56:11 -0400 (Tue,
+ 05 May 2009) | 11 lines Fix Javascript error when using astman.js
+ in Internet Explorer. Internet Explorer (tested with 7.0) does
+ not like trailing commas on constructs like object initializers,
+ so get rid of them to avoid some errors. (closes issue #15026)
+ Reported by: rajnishgiri Patches: bug15026.patch uploaded by
+ seanbright (license 71) Tested by: seanbright ........
+
+2009-05-05 18:23 +0000 [r192430-192462] Joshua Colp <jcolp@digium.com>
+
+ * /, main/features.c: Merged revisions 192454 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r192454 | file | 2009-05-05 15:22:27 -0300 (Tue, 05 May 2009) | 8
+ lines Fix an incorrect assumption that certain values on the
+ channel will always exist when they may not. The CDR code
+ involved with bridges wrongly assumed that the currently
+ executing application and data values will always exist. It is
+ possible for this to be false when call forwarding is involved.
+ (closes issue #14984) Reported by: gincantalupo ........
+
+ * /, apps/app_followme.c: Merged revisions 192429 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r192429 | file | 2009-05-05 14:43:30 -0300 (Tue, 05 May 2009) | 5
+ lines Fix a bug where the followme application would continue
+ trying numbers after the caller hung up. (closes issue #13624)
+ Reported by: sgenyuk ........
+
+2009-05-05 17:33 +0000 [r192427] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_dahdi.c: Revert CPC patch for now, until I decide
+ whether or not it all should be merged into libss7/1.0 (It's
+ still in the bug13495 branch and in libss7/trunk)
+
+2009-05-05 14:22 +0000 [r192387] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix a bug with setting t38pt_udptl at the
+ user or peer level. If an incoming call authenticated as a user
+ or peer and t38pt_udptl was not set to yes in general then no
+ UDPTL session would be present and any T38 related things would
+ fail. This commit changes it so that if after authenticating T38
+ is enabled but no UDPTL session is present one will be created.
+ (issue AST-215)
+
+2009-05-05 14:17 +0000 [r192279-192362] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/utils.c, include/asterisk/stringfields.h: Add a more
+ efficient way of allocating structures that use stringfields This
+ commit adds an API call that can be used to allocate a structure
+ along with this stringfield storage in a single allocation.
+
+ * main/utils.c, main/astobj2.c, include/asterisk/stringfields.h:
+ Correct some flaws in the memory accounting code for stringfields
+ and ao2 objects Under some conditions, the memory allocation for
+ stringfields and ao2 objects would not have supplied valid
+ file/function names for MALLOC_DEBUG tracking, so this commit
+ corrects that.
+
+ * main/channel.c, include/asterisk/astobj2.h,
+ include/asterisk/datastore.h, include/asterisk/channel.h,
+ main/astobj2.c, main/datastore.c: Properly account for memory
+ allocated for channels and datastores As in previous commits,
+ when channels are allocated (with ast_channel_alloc) or
+ datastores are allocated (with ast_datastore_alloc) properly
+ account for the memory being owned by the caller, instead of the
+ allocator function itself.
+
+ * main/utils.c, include/asterisk/stringfields.h: Ensure that string
+ pools allocated to hold stringfields are properly accounted in
+ MALLOC_DEBUG mode This commit modifies the stringfield pool
+ allocator to remember the 'owner' of the stringfield manager the
+ pool is being allocated for, and ensures that pools allocated in
+ the future when fields are populated are owned by that
+ file/function.
+
+2009-05-04 22:44 +0000 [r192214] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 192213 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r192213 | dvossel | 2009-05-04 17:37:31 -0500 (Mon, 04
+ May 2009) | 11 lines global mohinterpret setting is ignored
+ mohinterpret and mohsuggest global variables were not copied over
+ during build_users and build_peers. (closes issue #14728)
+ Reported by: dimas Patches: v1-14728.patch uploaded by dimas
+ (license 88) Tested by: dimas, dvossel ........
+
+2009-05-04 19:29 +0000 [r192132-192171] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/autoconfig.h.in, res/res_agi.c: Restore
+ 'asyncagi break' command to 1.6.1 and higher. (closes issue
+ #14985) Reported by: nikkk Patches: 20090428__bug14985.diff.txt
+ uploaded by tilghman (license 14)
+ 20090429__bug14985__1.6.1.diff.txt uploaded by tilghman (license
+ 14) Tested by: nikkk
+
+ * autoconf/ast_ext_tool_check.m4: Pass libraries in LIBS, not
+ LDFLAGS. (closes issue #14671) Reported by: Chainsaw Patches:
+ asterisk-1.6.0.6-toolcheck-libs-not-ldflags.patch uploaded by
+ Chainsaw (license 723)
+
+2009-05-04 17:42 +0000 [r192096] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_forkcdr.c: Commit documentation changes related to issue
+ #14801. (issue #14801)
+
+2009-05-04 16:24 +0000 [r192059] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/astobj2.h, main/astobj2.c: Ensure that astobj2
+ memory allocations are properly accounted for when MALLOC_DEBUG
+ is used This commit ensures that all astobj2 allocated objects
+ are properly accounted for in MALLOC_DEBUG mode by passing down
+ the file/function/line information from the module/function that
+ actually called the astobj2 allocation function.
+
+2009-05-04 15:35 +0000 [r192032] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/xml.c: Do not re-define _POSIX_C_SOURCE if it was already
+ defined.
+
+2009-05-04 12:52 +0000 [r191919-191997] Kevin P. Fleming <kpfleming@digium.com>
+
+ * tests/test_skel.c, tests/test_sched.c: Minor changes in test
+ modules Correct command description in test_sched.c and include
+ asterisk/cli.h in test_skel.c, since it's highly unlikely that a
+ test module will *not* want to provide CLI commands to execute
+ the tests
+
+ * configs/modules.conf.sample: Ensure that by default only one
+ console channel driver is loaded This configuration file was
+ changed to ensure that only one console channel driver (chan_oss)
+ is loaded by default, but the change would only work if
+ chan_console was not built. Now it will work as expected; if
+ chan_alsa or chan_console are built and installed, they will not
+ be loaded unless explicity requested.
+
+ * include/asterisk/event.h, include/asterisk/event_defs.h,
+ main/event.c: Add 'bitflags'-style information elements to event
+ framework This patch add a new payload type for information
+ elements, a set of bit flags. The payload is transported as a
+ 32-bit unsigned integer but when matching is performed between
+ events and subscribers, the matching is done by using a bitwise
+ AND instead of numeric value comparison. Review:
+ http://reviewboard.asterisk.org/r/242/
+
+2009-05-03 14:05 +0000 [r191848-191884] Russell Bryant <russell@digium.com>
+
+ * Makefile: Remove unnecessary compiler flag
+
+ * main/event.c: Do a bit of code cleanup. - convert handling of IE
+ PLTYPEs to switch statements - add braces to various small blocks
+ - remove a bit of trailing whitespace - remove a couple of
+ unnecessary ast_strdupa() uses
+
+2009-05-02 19:02 +0000 [r191775-191785] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/logger.h, main/manager.c, pbx/pbx_spool.c,
+ main/logger.c, apps/app_sms.c, CHANGES, apps/app_verbose.c,
+ configs/logger.conf.sample: Remove rarely-used
+ event_log/LOG_EVENT support In discussions today at the Europe
+ Asterisk Developer Meet-Up, we determined that the event_log was
+ used in only 9 places in the entire tree, and really was not
+ needed at all. The users have been converted to use LOG_NOTICE,
+ or the messages have been removed since other messages were
+ already in place that provided the same information.
+
+ * main/logger.c: Fix an error in queue_log file rotation
+ optimization code This code was copy-and-pasted without properly
+ changing references to event_rotate into queue_rotate, so under
+ some conditions the log rotation would rotate queue_log even
+ though it was not necessary.
+
+2009-05-02 16:43 +0000 [r191700-191739] Sean Bright <sean@malleable.com>
+
+ * channels/chan_dahdi.c: Conditional include ioctl's to change EC
+ policy based on DAHDI caps. This feels like a sane change
+ (wouldn't compile without this addition), but I'm not intimately
+ familiar with this code.
+
+ * main/asterisk.c: Update copyright year to 2009
+
+2009-05-01 20:01 +0000 [r191494-191560] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 191559 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r191559 | tilghman | 2009-05-01 15:00:23 -0500 (Fri, 01 May 2009)
+ | 6 lines SIP Response 410 maps to cause code 22 (or 23), not 1.
+ (closes issue #14993) Reported by: BigJimmy Patches: causepatch
+ uploaded by BigJimmy (license 371) ........
+
+ * channels/chan_iax2.c: Set debug message back to DEBUG level.
+ (closes issue #15007) Reported by: hulber
+
+2009-05-01 18:09 +0000 [r191489] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 191488 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r191488 | jpeeler | 2009-05-01 12:40:46 -0500 (Fri, 01 May 2009)
+ | 9 lines Fix DTMF not being sent to other side after a partial
+ feature match This fixes a regression from commit 176701. The
+ issue was that ast_generic_bridge never exited after the feature
+ digit timeout had elapsed, which prevented the queued DTMF from
+ being sent to the other side. This issue was reported to me
+ directly. ........
+
+2009-05-01 14:58 +0000 [r191419] Joshua Colp <jcolp@digium.com>
+
+ * main/audiohook.c: Drop my IRC nickname.
+
+2009-05-01 09:50 +0000 [r191418] TransNexus OSP Development <support@transnexus.com>
+
+ * configs/osp.conf.sample, apps/app_osplookup.c: Made security
+ features optional.
+
+2009-04-30 21:42 +0000 [r191411] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, CHANGES: Add
+ buffer and echo canceller control to CHANNEL() dialplan function
+ for DAHDI channels Adds ability for CHANNEL() dialplan function,
+ when used on DAHDI channels, to temporarily change the number of
+ buffers and/or the buffer policy, and also to enable, disable, or
+ switch the echo canceller between FAX/data and voice modes.
+
+2009-04-30 17:40 +0000 [r191367] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ main/asterisk.c: Detect eaccess (or euidaccess) before using it.
+ Reported by Andrew Lindh via the -dev list.
+
+2009-04-30 09:11 +0000 [r191300-191332] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c: Added routing number support.
+
+ * apps/app_osplookup.c: Fixed not report source network ID and not
+ export destination network ID issues.
+
+2009-04-30 06:47 +0000 [r191219-191283] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c: Change working directory to / under certain
+ conditions. If backgrounding and no core will be produced, then
+ changing the directory won't break anything; likewise, if the CWD
+ isn't accessible by the current user, then a core wasn't possible
+ anyway. (closes issue #14831) Reported by: chris-mac Patches:
+ 20090428__bug14831.diff.txt uploaded by tilghman (license 14)
+ 20090430__bug14831.diff.txt uploaded by tilghman (license 14)
+ Tested by: chris-mac
+
+ * /: Recorded merge of revisions 191220 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r191220 | tilghman | 2009-04-29 18:10:54 -0500 (Wed, 29 Apr 2009)
+ | 2 lines Allow H.323 to compile with FDLEAK checking enabled.
+ ........
+
+ * channels/h323/ast_h323.cxx, channels/chan_h323.c: Make H.323
+ compile with FDLEAK detection code enabled
+
+2009-04-29 22:56 +0000 [r191213] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_phoneprov.c: fix typos
+
+2009-04-29 22:23 +0000 [r191211] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Part of the merge did not happen correctly, which
+ resulted in a compile error
+
+2009-04-29 21:13 +0000 [r191177] David Vossel <dvossel@digium.com>
+
+ * main/tcptls.c, configs/sip.conf.sample,
+ include/asterisk/tcptls.h, CHANGES: SIP option to specify
+ outbound TLS/SSL client protocol. chan_sip allows for outbound
+ TLS connections, but does not allow the user to specify what
+ protocol to use (default was SSLv2, and still is if this new
+ option is not specified). This patch lets the user pick the
+ SSL/TLS client method for outbound connections in sip. (closes
+ issue #14770) Reported by: TheOldSaint (closes issue #14768)
+ Reported by: TheOldSaint Review:
+ http://reviewboard.digium.com/r/240/
+
+2009-04-29 21:07 +0000 [r191175] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, CHANGES: Outgoing PTP redirected calls did
+ not wait for the COLR from the redirected-to party. For outgoing
+ PTP redirected calls, you now need to use the inhibit(i) option
+ on all of the REDIRECTING statements before dialing the
+ redirected-to party. You still have to set the
+ REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The
+ PTP call will update the redirecting-to presentation when it
+ becomes available and queue the redirecting update to the calling
+ channel.
+
+2009-04-29 18:53 +0000 [r191140] Tilghman Lesher <tlesher@digium.com>
+
+ * tests/test_substitution.c (added), funcs/func_base64.c,
+ funcs/func_rand.c, funcs/func_speex.c, funcs/func_md5.c,
+ funcs/func_module.c, include/asterisk/autoconfig.h.in,
+ funcs/func_env.c, funcs/func_strings.c, res/res_phoneprov.c,
+ funcs/func_sysinfo.c, funcs/func_vmcount.c, funcs/func_sha1.c,
+ funcs/func_logic.c, apps/app_exec.c, funcs/func_groupcount.c,
+ configure, funcs/func_aes.c, main/ast_expr2f.c, res/res_agi.c,
+ apps/app_minivm.c, include/asterisk/ast_expr.h, cdr/cdr_custom.c,
+ main/strings.c, main/pbx.c, funcs/func_dialplan.c,
+ funcs/func_db.c, funcs/func_timeout.c, funcs/func_lock.c,
+ funcs/func_cut.c, funcs/func_extstate.c, res/res_config_curl.c,
+ funcs/func_curl.c, funcs/func_blacklist.c, apps/app_macro.c,
+ include/asterisk/pbx.h, funcs/func_callerid.c,
+ apps/app_voicemail.c: Merge str_substitution branch. This branch
+ adds additional methods to dialplan functions, whereby the result
+ buffers are now dynamic buffers, which can be expanded to the
+ size of any result. No longer are variable substitutions limited
+ to 4095 bytes of data. In addition, the common case of needing
+ buffers much smaller than that will enable substitution to only
+ take up the amount of memory actually needed. The existing
+ variable substitution routines are still available, but users of
+ those API calls should transition to using the dynamic-buffer
+ APIs. Reviewboard: http://reviewboard.digium.com/r/174/
+
+2009-04-29 18:32 +0000 [r191136] David Brooks <dbrooks@digium.com>
+
+ * pbx/pbx_config.c: Removing crufty code that is no longer
+ necessary. Code cleanup.
+
+2009-04-29 14:39 +0000 [r191028] David Vossel <dvossel@digium.com>
+
+ * main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c,
+ configs/manager.conf.sample, include/asterisk/tcptls.h, CHANGES,
+ configs/http.conf.sample: Consistent SSL/TLS options across conf
+ files ast_tls_read_conf() is a new api call for handling SSL/TLS
+ options across all conf files. Before this change, SSL/TLS
+ options were not consistent. http.conf and manager.conf required
+ the 'ssl' prefix while sip.conf used options with the 'tls'
+ prefix. While the options had different names in different conf
+ files, they all did the exact same thing. Now, instead of mixing
+ 'ssl' or 'tls' prefixes to do the same thing depending on what
+ conf file you're in, all SSL/TLS options use the 'tls' prefix.
+ For example. 'sslenable' in http.conf and manager.conf is now
+ 'tlsenable' which matches what already existed in sip.conf. Since
+ this has the potential to break backwards compatibility, previous
+ options containing the 'ssl' prefix still work, but they are no
+ longer documented in the sample.conf files. The change is noted
+ in the CHANGES file though. Review:
+ http://reviewboard.digium.com/r/237/
+
+2009-04-29 08:58 +0000 [r190989-190993] Russell Bryant <russell@digium.com>
+
+ * main/indications.c: Log an error message if indications.conf is
+ not found. (closes issue #14990) Reported by: tzafrir Patches:
+ indications_err.diff uploaded by tzafrir (license 46)
+
+ * apps/app_queue.c: Fix app_queue XML documentation. I think it
+ would behoove us to force "make validate-docs" to be run after
+ the XML documentation has been generated if dev-mode is enabled.
+ (closes issue #14989) Reported by: tzafrir Patches:
+ app_queue_xml.diff uploaded by tzafrir (license 46)
+
+ * main/rtp_engine.c, include/asterisk/channel.h: Resolve Solaris
+ build issues and add some API documentation. (issue #14981)
+ Reported by: snuffy
+
+2009-04-28 22:07 +0000 [r190946-190947] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_dahdi.c: Add support setting CPC from channel
+ variable
+
+ * channels/chan_dahdi.c: Make sure that we do not clear the down
+ flag on the BRI during PTMP link transients
+
+2009-04-28 17:31 +0000 [r190904] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/tex/cdrdriver.tex: UniqueID column has a maximum size of 150
+
+2009-04-28 14:15 +0000 [r190861-190865] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile: Build XML documention from *only* the source files that
+ have docs in them Change the build process so that
+ doc/core-en_US.xml is dependent solely on the source files that
+ have documentation in them, not on all source files.
+
+ * Makefile.rules: Remove Makefile rules for bison and flex sources
+ We never, ever want these files to processed automatically,
+ because we store the output files in Subversion and users should
+ never need to rebuild them.
+
+2009-04-28 09:10 +0000 [r190830] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c: Updated for OSP Toolkit 3.5.
+
+2009-04-27 21:22 +0000 [r190735-190797] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Fix a small memory leak on error in
+ ast_channel_alloc().
+
+ * channels/misdn/isdn_lib.h, channels/chan_misdn.c, CHANGES,
+ channels/misdn/isdn_lib.c, funcs/func_redirecting.c: Make PTP
+ DivertingLegInformation3 message behavior closer to the
+ specifications. * Wait for a DivertingLegInformation3 message
+ after receiving a DivertingLegInformation1 message to complete
+ the redirecting-to information before queuing a redirecting
+ update to the other channel. * A DivertingLegInformation2 message
+ should be responded to with a DivertingLegInformation3 when the
+ COLR is determined. If the call could or does experience another
+ redirection, you should manually determine the COLR to send to
+ the switch by setting REDIRECTING(to-pres) to the COLR and
+ setting REDIRECTING(to-num) = ${EXTEN}. * A
+ DivertingLegInformation2 message must have an original called
+ number if the redirection count is greater than one. Since
+ Asterisk does not keep track of this information, we can only
+ indicate that the number is not available due to interworking.
+
+2009-04-27 19:34 +0000 [r190726] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Don't warn on pipe in the System call. (closes issue
+ #14979) Reported by: pj
+
+2009-04-27 19:30 +0000 [r190725] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in: Merged revisions
+ 190721 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r190721 | kpfleming | 2009-04-27 14:29:46 -0500 (Mon, 27 Apr
+ 2009) | 7 lines Fix 'inconsistent line endings' when autoconf
+ 2.63 is used Attempt to make configure script regeneration 'safe'
+ using autoconf 2.63, which embeds a bare CR into the script, thus
+ making Subversion complain about inconsistent line endings This
+ commit changes the MIME type of the configure script to be
+ 'binary' thus making Subversion no longer inspect line endings,
+ and as a bonus 'svn diff' will no longer try to generate diff
+ output for it, which is not generally useful anyway. ........
+
+2009-04-27 19:08 +0000 [r190663] Russell Bryant <russell@digium.com>
+
+ * res/res_smdi.c, /: Merged revisions 190661-190662 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r190661 | russell | 2009-04-27 14:00:54 -0500 (Mon, 27
+ Apr 2009) | 9 lines Resolve a crash in res_smdi when used with
+ chan_dahdi. When chan_dahdi goes to get an SMDI message, it
+ provides no search criteria. It just grabs the next message that
+ arrives. This code was written with the SMDI dialplan functions
+ in mind, since that is now the preferred method of using SMDI.
+ However, this broke support of it being used from chan_dahdi.
+ (closes AST-212) ........ r190662 | russell | 2009-04-27 14:03:59
+ -0500 (Mon, 27 Apr 2009) | 2 lines Fix a typo from 190661.
+ ........
+
+2009-04-27 16:37 +0000 [r190622-190626] Mark Michelson <mmichelson@digium.com>
+
+ * doc/tex/channelvariables.tex, apps/app_queue.c: Allow for a
+ position to be specified when entering a queue. This would allow
+ for one to add a caller to a specific place in the queue instead
+ of just placing the caller in the back every time. To help
+ facilitate some interesting manipulations, a new channel variable
+ called QUEUEPOSITION has been added. When a caller is removed
+ from a queue, his position in that queue is stored in the
+ QUEUEPOSITION variable. One such strategy an administrator can
+ employ is to allow for the removal of a caller from one queue
+ followed by the insertion of the same caller into a separate
+ queue in the same position. Review:
+ http://reviewboard.digium.com/r/189
+
+ * apps/app_queue.c: Update warning message to not have pipes and
+ contain all options.
+
+2009-04-27 15:18 +0000 [r190586] Joshua Colp <jcolp@digium.com>
+
+ * main/manager.c: Fix a bug where we tried to send events out when
+ no sessions container was present. This commit stops a warning
+ message (user_data is NULL) from getting output when manager
+ events get sent before manager is initialized. This happens
+ because manager is initialized *after* modules are loaded and the
+ act of loading modules triggers manager events. (issue #14974)
+ Reported by: pj
+
+2009-04-27 14:46 +0000 [r190577] Mark Michelson <mmichelson@digium.com>
+
+ * configs/sip.conf.sample: Remove nonexistent option from
+ sip.conf.sample. The option to choose which connected line header
+ to use is not 'rpid_header' but 'sendrpid'
+
+2009-04-24 21:22 +0000 [r190545] David Vossel <dvossel@digium.com>
+
+ * main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c,
+ configs/manager.conf.sample, configs/sip.conf.sample,
+ include/asterisk/tcptls.h, CHANGES, configs/http.conf.sample:
+ TLS/SSL private key option Adds option to specify a private key
+ .pem file when configuring TLS or SSL in AMI, HTTP, and SIP.
+ Before this, the certificate file was used for both the public
+ and private key. It is possible for this file to hold both, but
+ most configurations allow for a separate private key file to be
+ specified. Clarified in .conf files how these options are to be
+ used. The current conf files do not explain how the private key
+ is handled at all, so without knowledge of Asterisk's TLS
+ implementation, it would be hard to know for sure what was going
+ on or how to set it up. Review:
+ http://reviewboard.digium.com/r/234/
+
+2009-04-24 17:59 +0000 [r190516-190517] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, funcs/func_connectedline.c: There is no
+ need to use the struct ast_party_connected_line.source update
+ values. The messages sent by a technology when a connected line
+ update is received are best determined by the current call state
+ of the channel. The struct ast_party_connected_line.source value
+ is really only useful as a possible tracing aid.
+
+ * include/asterisk/channel.h: Update comment.
+
+2009-04-24 15:26 +0000 [r190423-190484] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/channel.h: Add \since tag for new API calls.
+
+ * channels/chan_misdn.c: Fix a build error.
+
+ * channels/chan_unistim.c, channels/chan_local.c,
+ apps/app_dahdiscan.c (removed), main/devicestate.c,
+ main/autochan.c (added), funcs/func_logic.c,
+ channels/chan_gtalk.c, channels/chan_iax2.c, main/cli.c,
+ main/channel.c, build_tools/cflags.xml, channels/chan_dahdi.c,
+ main/manager.c, funcs/func_odbc.c, apps/app_minivm.c,
+ main/features.c, res/res_agi.c, main/logger.c,
+ channels/chan_mgcp.c, res/res_clioriginate.c, main/pbx.c,
+ channels/chan_sip.c, include/asterisk/autochan.h (added),
+ channels/chan_bridge.c, main/Makefile, apps/app_softhangup.c,
+ channels/chan_agent.c, UPGRADE.txt, include/asterisk/channel.h,
+ CHANGES, funcs/func_global.c, res/res_monitor.c,
+ apps/app_channelredirect.c, channels/chan_misdn.c,
+ apps/app_directed_pickup.c, funcs/func_channel.c,
+ res/snmp/agent.c, include/asterisk/lock.h, apps/app_senddtmf.c,
+ apps/app_mixmonitor.c, apps/app_chanspy.c, apps/app_voicemail.c:
+ Convert the ast_channel data structure over to the astobj2
+ framework. There is a lot that could be said about this, but the
+ patch is a big improvement for performance, stability, code
+ maintainability, and ease of future code development. The channel
+ list is no longer an unsorted linked list. The main container for
+ channels is an astobj2 hash table. All of the code related to
+ searching for channels or iterating active channels has been
+ rewritten. Let n be the number of active channels. Iterating the
+ channel list has gone from O(n^2) to O(n). Searching for a
+ channel by name went from O(n) to O(1). Searching for a channel
+ by extension is still O(n), but uses a new method for doing so,
+ which is more efficient. The ast_channel object is now a
+ reference counted object. The benefits here are plentiful. Some
+ benefits directly related to issues in the previous code include:
+ 1) When threads other than the channel thread owning a channel
+ wanted access to a channel, it had to hold the lock on it to
+ ensure that it didn't go away. This is no longer a requirement.
+ Holding a reference is sufficient. 2) There are places that now
+ require less dealing with channel locks. 3) There are places
+ where channel locks are held for much shorter periods of time. 4)
+ There are places where dealing with more than one channel at a
+ time becomes _MUCH_ easier. ChanSpy is a great example of this.
+ Writing code in the future that deals with multiple channels will
+ be much easier. Some additional information regarding channel
+ locking and reference count handling can be found in channel.h,
+ where a new section has been added that discusses some of the
+ rules associated with it. Mark Michelson also assisted with the
+ development of this patch. He did the conversion of ChanSpy and
+ introduced a new API, ast_autochan, which makes it much easier to
+ deal with holding on to a channel pointer for an extended period
+ of time and having it get automatically updated if the channel
+ gets masqueraded. Mark was also a huge help in the code review
+ process. Thanks to David Vossel for his assistance with this
+ branch, as well. David did the conversion of the DAHDIScan
+ application by making it become a wrapper for ChanSpy internally.
+ The changes come from the
+ svn/asterisk/team/russell/ast_channel_ao2 branch. Review:
+ http://reviewboard.digium.com/r/203/
+
+2009-04-24 13:49 +0000 [r190421] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix nat setting on RTP instances. (closes
+ issue #14827) Reported by: pj
+
+2009-04-23 21:13 +0000 [r190357] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 190356 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r190356 | russell | 2009-04-23 16:07:07 -0500 (Thu, 23 Apr 2009)
+ | 2 lines Remove a bogus ast_channel_unlock(). ........
+
+2009-04-23 20:42 +0000 [r190349-190352] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Labels are sometimes (most of the time?) NULL for
+ extensions. (closes issue #14895) Reported by: chris-mac Patches:
+ 20090423__bug14895__2.diff.txt uploaded by tilghman (license 14)
+ Tested by: lmadsen
+
+ * include/asterisk/http.h, include/asterisk/utils.h,
+ main/manager.c, res/res_phoneprov.c, main/http.c, main/utils.c,
+ res/res_http_post.c, main/astobj2.c: Support HTTP digest
+ authentication for the http manager interface. (closes issue
+ #10961) Reported by: ys Patches: digest_auth_r148468_v5.diff
+ uploaded by ys (license 281) SVN branch
+ http://svn.digium.com/svn/asterisk/team/group/manager_http_auth
+ Tested by: ys, twilson, tilghman Review:
+ http://reviewboard.digium.com/r/223/ Reviewed by:
+ tilghman,russellb,mmichelson
+
+2009-04-23 19:15 +0000 [r190287] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 190286 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r190286 | file | 2009-04-23 16:13:18 -0300 (Thu, 23 Apr
+ 2009) | 6 lines Fix a bug in chan_local glare hangup detection.
+ If both sides of a Local channel were hung up at around the same
+ time it was possible for one thread to destroy the local private
+ structure and have the other thread immediately try to remove the
+ already freed structure from the local channel list. ........
+
+2009-04-23 17:45 +0000 [r190250] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix reversed behavior of leavewhenempty option
+ in queues.conf. (closes issue #14650) Reported by: alecdavis
+ Patches: 14650.patch uploaded by mmichelson (license 60) Tested
+ by: mmichelson, lmadsen
+
+2009-04-23 16:55 +0000 [r190217] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_directed_pickup.c: Fix a double free issue with the
+ Pickup dialplan application. As part of the pickup process the
+ connected line information is updated. Part of this process does
+ a shallow copy of the target channel's connected line information
+ to a local structure. Once complete the structure contents are
+ freed. As a result any information in the target channel's
+ connected line information structure is no longer valid. This
+ change will now set the contents back to a clean state so that
+ the freeing of the target channel's connected line information
+ structure when the channel is destroyed will no longer try to
+ double free things. (closes issue #14839) Reported by: lmsteffan
+
+2009-04-23 00:44 +0000 [r190154] Terry Wilson <twilson@digium.com>
+
+ * funcs/func_strings.c: Fix example that could fail in certain
+ circumstances
+
+2009-04-22 21:38 +0000 [r190093] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/lock.h: Merged revisions 190092 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r190092 | tilghman | 2009-04-22 16:35:03 -0500 (Wed, 22
+ Apr 2009) | 7 lines Detect availability of
+ pthread_rwlock_timedwrlock() before using it. (closes issue
+ #14930) Reported by: tilghman Patches:
+ 20090420__bug14930.diff.txt uploaded by tilghman (license 14)
+ Tested by: mvanbaak, tilghman ........
+
+2009-04-22 21:15 +0000 [r190057] Jeff Peeler <jpeeler@digium.com>
+
+ * funcs/func_groupcount.c, main/app.c, include/asterisk/channel.h,
+ main/cli.c: Fix building of chan_h323 with gcc-3.3 There seems to
+ be a bug with old versions of g++ that doesn't allow a structure
+ member to use the name list. Rename list member to group_list in
+ ast_group_info and change the few places it is used. (closes
+ issue #14790) Reported by: stuarth
+
+2009-04-22 20:07 +0000 [r190000] Terry Wilson <twilson@digium.com>
+
+ * funcs/func_strings.c: Add funcs for manipulating delimited lists
+ in the dialplan Adds PUSH and POP for appending to and
+ retrieving/removing from the end of a list and UNSHIFT and SHIFT
+ for insert to and retrieiving/ removing from the beginning of a
+ list. Review: http://reviewboard.digium.com/r/230
+
+2009-04-22 19:23 +0000 [r189993] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/h323/ast_h323.cxx, channels/chan_h323.c,
+ channels/h323/chan_h323.h: Make chan_h323 respect packetization
+ settings and fix small reload issue. Previously, packetization
+ settings were ignored and now they are not. A new config option
+ 'autoframing' has been added to mirror the way chan_sip handles
+ it. Turning on the autoframing option (available both as a global
+ option or per peer) overrides the local settings with the remote
+ packetization settings. Testing was performed with varying
+ packetization levels with the following codecs: ulaw, alaw, gsm,
+ and g729. Also, an unrelated config reload issue has been fixed
+ in the case of the config file not changing. (closes issue
+ #12415) Reported by: pj Patches:
+ 2009012200_h323packetization.diff.txt uploaded by mvanbaak
+ (license 7), modified by me
+
+2009-04-22 16:56 +0000 [r189951] Russell Bryant <russell@digium.com>
+
+ * main/features.c: Fix call parking callback. Pipes -> Commas.
+
+2009-04-22 16:01 +0000 [r189911] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_unistim.c: Do not continue to receive DTMF, when
+ the channel is hungup and about to be destroyed. (closes issue
+ #14858) Reported by: barryf Patches: 20090421__bug14858.diff.txt
+ uploaded by tilghman (license 14) Tested by: barryf
+
+2009-04-22 14:30 +0000 [r189850] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 189849
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189849 | mvanbaak | 2009-04-22 16:29:28 +0200 (Wed, 22 Apr 2009)
+ | 12 lines replace sed with tr to remove \r from downloaded file
+ On some systems, sed does not recognize \r in the pattern the way
+ it was used here. Use tr instead because this works the same
+ across systems. (closes issue #14936) Reported by: leobrown
+ Patches: 2009042201_14936.diff.txt uploaded by mvanbaak (license
+ 7) Tested by: leobrown, mvanbaak ........
+
+2009-04-22 06:33 +0000 [r189813] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, configure.ac: Detect liblua on SuSE, and add libm for
+ linking for Fedora. (Reported via the -dev list, Subject:
+ Compiling Asterisk with LUA)
+
+2009-04-21 20:28 +0000 [r189771] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Fixes segfault when switching UDP to TCP in
+ sip.conf after reload. If transport in sip.conf is switched from
+ UDP to TCP, Asterisk segfaults right after issuing a sip reload.
+ The problem is the socket type is changed to TCP but the fd may
+ still be present for UDP. Later, when the TCP session should be
+ created or set using an existing one, it isn't because the old
+ file descriptor is still present. Now every time transport is
+ changed during a sip.conf reload, the file descriptor is set to
+ -1, signifying it must be created or found. (closes issue #14727)
+ Reported by: pj Tested by: dvossel Review:
+ http://reviewboard.digium.com/r/229/
+
+2009-04-21 17:44 +0000 [r189735] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h,
+ channels/chan_misdn.c, channels/misdn/chan_misdn_config.h,
+ channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c,
+ configs/misdn.conf.sample, CHANGES, channels/misdn/isdn_lib.c,
+ channels/misdn_config.c: Added CCBS/CCNR Party A support and
+ enhanced COLP support. This change adds the following features to
+ chan_misdn: * CCBS/CCNR Party A support for PTMP and PTP modes. *
+ Enhances COLP support for call diversion and explicit call
+ transfer. These enhanced features require a modified version of
+ mISDN. The latest modified mISDN v1.1.x based version is
+ available at: http://svn.digium.com/svn/thirdparty/mISDN/trunk
+ http://svn.digium.com/svn/thirdparty/mISDNuser/trunk Taged
+ versions of the modified mISDN code are available under:
+ http://svn.digium.com/svn/thirdparty/mISDN/tags
+ http://svn.digium.com/svn/thirdparty/mISDNuser/tags Review:
+ http://reviewboard.digium.com/r/218/ Merged from
+ team/rmudgett/misdn_facility branch.
+
+2009-04-21 15:54 +0000 [r189629-189665] Doug Bailey <dbailey@digium.com>
+
+ * utils/muted.c, /: Merged revisions 189664 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189664 | dbailey | 2009-04-21 10:52:13 -0500 (Tue, 21 Apr 2009)
+ | 2 lines Remove daemon call on systems that do not support
+ forking. ........
+
+ * /, configure, include/asterisk/autoconfig.h.in,
+ include/asterisk/compat.h, configure.ac: Merged revisions 189601
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189601 | dbailey | 2009-04-21 09:00:55 -0500 (Tue, 21 Apr 2009)
+ | 3 lines Add check in configure script to check for GLOB_NOMAGIC
+ and GLOB_BRACE in glob.h This allows config.c to compile when
+ linked against uclibc that does not support these parameters
+ ........
+
+2009-04-20 22:10 +0000 [r189539] Tilghman Lesher <tlesher@digium.com>
+
+ * main/stdtime/localtime.c: Use nanosleep instead of poll. This is
+ not just because mmichelson suggested it, but also because Mac OS
+ X puked on my poll().
+
+2009-04-20 21:29 +0000 [r189495-189516] Terry Wilson <twilson@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 189465 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189465 | twilson | 2009-04-20 16:10:27 -0500 (Mon, 20 Apr 2009)
+ | 2 lines Update CDR appropriately when AST_CAUSE_NO_ANSWER is
+ set ........
+
+ * apps/app_dial.c, /: Merged revisions 189463 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189463 | twilson | 2009-04-20 16:00:52 -0500 (Mon, 20 Apr 2009)
+ | 2 lines Don't treat a NOANSWER like a CHANUNAVAIL ........
+
+2009-04-20 21:09 +0000 [r189464] Sean Bright <sean@malleable.com>
+
+ * /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 189462 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189462 | seanbright | 2009-04-20 16:58:39 -0400 (Mon, 20 Apr
+ 2009) | 13 lines Properly handle @s within hints in AEL. AEL was
+ not handling the case of a device hint containing an @ symbol,
+ which caused parking hints (e.g. hint(park:exten@context)) to
+ error out the parser. This patch makes AEL treat the @ the same
+ way it treats colon and ampersand now, meaning the characters are
+ included in verbatim. (closes issue #14941) Reported by: bpgoldsb
+ Patches: bug14941.patch uploaded by seanbright (license 71)
+ Tested by: bpgoldsb ........
+
+2009-04-20 19:28 +0000 [r189419] Doug Bailey <dbailey@digium.com>
+
+ * main/manager.c, /, main/db1-ast/recno/rec_open.c,
+ channels/chan_iax2.c: Merged revisions 189391 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189391 | dbailey | 2009-04-20 14:10:56 -0500 (Mon, 20 Apr 2009)
+ | 4 lines Clean up problem with manager implementation of mmap
+ where it was not testing against MAP_FAILED response. Got rid of
+ shadowed variable used in processign the mmap results. Change
+ test of mmap results to compare against MAP_FAILED ........
+
+2009-04-20 17:05 +0000 [r189350] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix a bug with non-UDP connections that
+ caused dialogs to not get freed. This issue crept up because of a
+ reference count issue on non-UDP based dialogs. The dialog
+ reference count was increased when transmitting a packet reliably
+ but never decreased. This caused the dialog structure to hang
+ around despite being unlinked from the dialogs container. (closes
+ issue #14919) Reported by: vrban
+
+2009-04-20 14:05 +0000 [r189278] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 189277 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189277 | mmichelson | 2009-04-20 09:04:41 -0500 (Mon, 20 Apr
+ 2009) | 12 lines Move the check for chan->fdno == -1 to after the
+ zombie/hangup check. Many users were finding that their hung up
+ channels were staying up and causing 100% CPU usage. (issue
+ #14723) Reported by: seadweller Patches: 14723_1-4-tip.patch
+ uploaded by mmichelson (license 60) Tested by: falves11, bamby
+ ........
+
+2009-04-18 01:28 +0000 [r189204] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_agent.c: Merged revisions 189203 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17
+ Apr 2009) | 12 lines Fixed autologoff in agents.conf not working
+ when agent logs in via AgentLogin app An agent logs in by calling
+ an extension that calls the AgentLogin app. In agents.conf
+ ackcall=always is set, so when they get a call they have the
+ choice to either acknowledge it or ignore it. autologoff=10 is
+ set as well, so if the agent ignores the call over 10sec one may
+ assume that the agent should be logged out (and in this case
+ hungup on as well), but this was not happening. (closes issue
+ #14091) Reported by: evandro Patches: autologoff.diff uploaded by
+ dvossel (license 671) Review:
+ http://reviewboard.digium.com/r/225/ ........
+
+2009-04-17 21:48 +0000 [r189137] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
+ revisions 188833,189134 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009)
+ | 4 lines Only disable mISDN DSP if Asterisk DSP is enabled.
+ Leave jitter setting alone. JIRA ABE-1835 ........ r189134 |
+ rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines
+ Modifed/added some debug messages. JIRA ABE-1835 ........
+
+2009-04-17 20:20 +0000 [r189097] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Prevent a crash when SIP blonde transferring
+ an unbridged call. If one attempts to use the attended transfer
+ button on a SIP phone to transfer an unbridged call (such as a
+ call to an IVR) but hangs up while the target of the transfer is
+ still ringing, we need to not crash. The problem was that
+ ast_hangup was called from outside the channel thread. AST-211
+
+2009-04-17 19:36 +0000 [r189077] Sean Bright <sean@malleable.com>
+
+ * main/asterisk.c: Fix copy/paste error with 'transmit silence'
+ flag.
+
+2009-04-17 15:44 +0000 [r189010] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 189009 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189009 | mnicholson | 2009-04-17 10:43:09 -0500 (Fri, 17 Apr
+ 2009) | 5 lines Make Busy() application set the CDR disposition
+ to BUSY. (closes issue #14306) Reported by: cristiandimache
+ ........
+
+2009-04-17 14:44 +0000 [r188947] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 188946 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) |
+ 15 lines Fix a bug where a value used to create the channel name
+ was bogus. This commit fixes the scenario where an incoming call
+ is authenticated using a peer entry. Previously the channel name
+ was created using either the username setting from the sip.conf
+ entry or the IP address that the call came from. Now the channel
+ name will be created using the peer name itself. This commit will
+ not change the way the channel name is generated for users or
+ friends. (closes issue #14256) Reported by: Nick_Lewis Patches:
+ chan_sip.c-chname.patch uploaded by Nick (license 657) Tested by:
+ Nick_Lewis, file ........
+
+2009-04-17 14:33 +0000 [r188942] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c: Fix a spacing issue that I claimed I would when I
+ committed this code. Nothing major though.
+
+2009-04-17 14:26 +0000 [r188938] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 188937 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r188937 | file | 2009-04-17 11:25:57 -0300 (Fri, 17 Apr
+ 2009) | 4 lines Fix a situation where the DAHDI channel private
+ structure lock was not unlocked when it should have been. (issue
+ AST-210) ........
+
+2009-04-17 13:29 +0000 [r188901] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c: Several fixes to the extenpatternmatchnew logic. 1.
+ Differentiate between literal characters in an extension and
+ characters that should be treated as a pattern match. Prior to
+ these fixes, an extension such as NNN would be treated as a
+ pattern, rather than a literal string of N's. 2. Fixed the logic
+ used when matching an extension with a bracketed expression, such
+ as 2[5-7]6. 3. Removed all areas of code that were executed when
+ NOT_NOW was #defined. The code in these areas had the potential
+ to crash, for one thing, and the actual intent of these blocks
+ seemed counterproductive. 4. Fixed many many coding guidelines
+ problems I encountered while looking through the corresponding
+ code. 5. Added failure cases and warning messages for when
+ duplicate extensions are encountered. 6. Miscellaneous fixes to
+ incorrect or redundant statements. (closes issue #14615) Reported
+ by: steinwej Tested by: mmichelson Review:
+ http://reviewboard.digium.com/r/194/
+
+2009-04-16 21:57 +0000 [r188774-188836] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 188835 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188835 | tilghman | 2009-04-16 16:41:13 -0500 (Thu, 16 Apr 2009)
+ | 7 lines Only update realtime, if global option rtupdate !=
+ false (closes issue #14885) Reported by: deepesh Patches:
+ 20090413__bug14885.diff.txt uploaded by tilghman (license 14)
+ Tested by: deepesh ........
+
+ * /, apps/app_voicemail.c: Merged revisions 188773 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r188773 | tilghman | 2009-04-16 16:02:29 -0500 (Thu, 16
+ Apr 2009) | 4 lines Umask should not be exported into global
+ namespace. (closes issue #14912) Reported by: jcapp ........
+
+2009-04-16 19:30 +0000 [r188742] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: SIP state notify reorganization What I've
+ done here is simply break up how a state NOTIFY is built.
+ Originally both the XML and sip header information were built
+ within the same function. While this does work, it does not allow
+ for the creation of multipart/related message bodies that can
+ contain multiple XML entries with only one sip header. Now a
+ separate function builds the XML for each notify. This patch also
+ makes maintaining and modifying state notifications in the future
+ much less of a pain. Review: http://reviewboard.digium.com/r/224/
+
+2009-04-16 13:42 +0000 [r188705] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_dahdi.c: Fix a bug with the dahdi_setoption
+ callback in chan_dahdi. This function incorrectly reported
+ success even if the option was unsupported. This was exposed by
+ the options to change the underlying channel format. The function
+ now returns a failure if the option is unsupported.
+
+2009-04-15 22:10 +0000 [r188647] David Vossel <dvossel@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 188646 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15
+ Apr 2009) | 12 lines National prefix inserted even when caller ID
+ not available When the caller ID is restricted, the expected
+ behavior is for the caller id to be blank. In chan_dahdi, the
+ national prefix is placed onto the callers number even if its
+ restricted (empty) causing the caller id to be the national
+ prefix rather than blank. (closes issue #13207) Reported by:
+ shawkris Patches: national_prefix.diff uploaded by dvossel
+ (license 671) Review: http://reviewboard.digium.com/r/220/
+ ........
+
+2009-04-15 20:17 +0000 [r188544-188585] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/file.c: Merged revisions 188582 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188582 | mmichelson | 2009-04-15 15:04:20 -0500 (Wed, 15 Apr
+ 2009) | 7 lines Update ast_readvideo_callback to match
+ ast_readaudio_callback. This fixes potential refcount errors that
+ may occur on ast_filestreams. AST-208 ........
+
+ * apps/app_dial.c: Make the cancellation of the dial timeout on a
+ call forward optional. This introduces the 'z' option to
+ app_dial. With it set, a call forward will cancel any timeout
+ originally set for this instance of the Dial application. AST-207
+
+2009-04-15 14:57 +0000 [r188515] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Don't try to do anything in
+ pri_check_restart with service messages unless libpri supports
+ it.
+
+2009-04-14 23:28 +0000 [r188470] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix a couple of queue member reference leaks.
+
+2009-04-14 17:40 +0000 [r188413] Joshua Colp <jcolp@digium.com>
+
+ * res/res_rtp_asterisk.c: Fix an incorrect clock rate when sending
+ T140 text. (closes issue #14029) Reported by: epicac
+
+2009-04-14 16:49 +0000 [r188342-188378] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, CHANGES: change some capitalization
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, CHANGES: Add
+ service maintenance message support This is the companion commit
+ to libpri r732. Service messages are now supported for switch
+ types 4ess/5ess. A new option service_message_support has been
+ added to chan_dahdi.conf and is noted in the sample config file.
+ The service message support is turned off by default. The current
+ implementation relies on AstDB to keep track of channel state,
+ which allows the statuses to be preserved across Asterisk
+ restarts. Below is a description of the storage format. The state
+ and reason for the service state are in the form
+ <state>:<reason>, where: <state> ::= { 'O' } // 'O' – Out Of
+ Service <reason> ::= { '0' | '1' | '2' | '3' }, where: '0' – No
+ reason (backwards compatibility) '1' – NEAR END '2' – FAR END '3'
+ – both NEAR and FAR END The new CLI commands to handle channel
+ service state are: pri service disable channel <chan> pri service
+ enable channel <chan> Many people contributed to the development
+ of this functionality. Because I entered at the very end I do not
+ know the exact history. Special thanks to all who moved the bug
+ forward one way or another: cmaj, PCadach, markster, mattf,
+ drmac, MikeJ, serge-v, murf, kanelbullar, Seb7, tilghman,
+ lmadsen, and especially dhubbard (he answered lots of my
+ questions and did a large portion of the work) (closes issue
+ #3450) Reported by: cmaj
+
+2009-04-14 14:22 +0000 [r188283-188284] Olle Johansson <oej@edvina.net>
+
+ * doc/manager_1_1.txt: New actions should go under "New Actions",
+ not "new events"
+
+ * main/xmldoc.c, apps/app_jack.c: Making sure we have references to
+ external libraries. Note: Update h.323 with the recent changes
+ too
+
+2009-04-14 13:14 +0000 [r188247] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix a bug with the change I made yesterday
+ to outbound proxy support. Per discussion with oej on IRC we need
+ the actual IP address, not the outbound proxy IP address, in the
+ sa field. This change matches the already existing code for all
+ other uses of the outbound proxy setting.
+
+2009-04-14 05:45 +0000 [r188206-188210] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: As suggested by Russell, warn users when their
+ dialplan arguments contain pipes, but not commas.
+
+ * utils/smsq.c: Application delimiter is ',', not '|'. (closes
+ issue #14881) Reported by: stegro Patches: smsq.patch uploaded by
+ stegro (license 752)
+
+2009-04-13 19:31 +0000 [r188102] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_musiconhold.c: Fix another crash related to cached
+ realtime music on hold. This was another off-by-one problem
+ caused by moh_register.
+
+2009-04-13 16:28 +0000 [r188067] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix a bug where using an outbound proxy
+ would cause the local address to be 127.0.0.1. Copy the outbound
+ proxy IP address into the SIP dialog structure as the IP address
+ we will be sending to. This has to be done because the logic that
+ determines what local IP address to use in the SIP messages is
+ not aware of an outbound proxy being in place. It only knows what
+ IP address we are sending to. (closes issue #12006) Reported by:
+ mnicholson
+
+2009-04-13 14:17 +0000 [r188032] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Set all queue variables on both the caller and
+ member channels. This allows for the variables to be accessed if
+ a member macro is run. Thanks to Grigoriy Puzankin for bringing
+ this up on the -dev list.
+
+2009-04-10 20:26 +0000 [r187906] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/Makefile: Fix module embedding for chan_h323. Include
+ libchanh323.a in the modules.link file so that all the symbols
+ can be resolved at link time. (closes issue #11966) Reported by:
+ dome Patches: issue_11966.patch uploaded by kpfleming (license
+ 421) Tested by: jpeeler
+
+2009-04-10 18:56 +0000 [r187830] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_local.c: Indicating connected line or redirecting
+ updates were missing a call to lock the local_pvt.
+
+2009-04-10 18:14 +0000 [r187772-187773] Joshua Colp <jcolp@digium.com>
+
+ * res/res_rtp_asterisk.c, main/rtp_engine.c: Change how we set the
+ local and remote address. The code will now only change the
+ address and port. It will not overwrite any other values.
+
+ * channels/chan_jingle.c, channels/chan_unistim.c,
+ res/res_rtp_asterisk.c, main/rtp_engine.c, channels/chan_sip.c,
+ channels/chan_skinny.c, channels/chan_h323.c,
+ channels/chan_gtalk.c, channels/chan_mgcp.c: Fix some
+ uninitialized memory notices that appeared under valgrind.
+
+2009-04-10 17:32 +0000 [r187770] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c: Make sure tc is unlocked before calling ast_call
+ since calling a Local channel could result in a deadlock.
+
+2009-04-10 17:29 +0000 [r187764] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/realtime_pgsql.sql, /,
+ contrib/scripts/sip-friends.sql: Merged revisions 187763 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r187763 | tilghman | 2009-04-10 12:28:46 -0500 (Fri, 10 Apr 2009)
+ | 2 lines Add lastms column to the contributed table designs
+ ........
+
+2009-04-10 16:51 +0000 [r187721] Kevin P. Fleming <kpfleming@digium.com>
+
+ * build_tools/embed_modules.xml: clean up some patterns for files
+ to remove add embedding support for bridge and test modules
+
+2009-04-10 16:26 +0000 [r187680-187714] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_local.c: ast_strdup failures aren't really failures
+ if the original value was NULL.
+
+ * main/channel.c: Don't let ast_channel_alloc fail if explicitly
+ passed NULL cid_name or cid_number. This also fixes a small
+ memory leak.
+
+2009-04-10 16:00 +0000 [r187675] Russell Bryant <russell@digium.com>
+
+ * tests/test_heap.c, tests/test_sched.c: Disable test modules by
+ default.
+
+2009-04-10 15:59 +0000 [r187674] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Ensure pvt is not NULL before dereferencing
+ it. (closes issue #14784) Reported by: pj
+
+2009-04-10 15:49 +0000 [r187673] David Vossel <dvossel@digium.com>
+
+ * apps/app_dial.c: Even more changes concerning r187426. Revised
+ where locks are placed yet once again. ast_call() should not be
+ called with a channel locked. could cause deadlock issues with
+ local channels.
+
+2009-04-10 15:11 +0000 [r187636] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/logger.h, main/logger.c, apps/app_verbose.c,
+ configs/logger.conf.sample: revert addition of LOG_SECURITY log
+ channel; after further discussion, a much better solution will be
+ used
+
+2009-04-10 14:53 +0000 [r187634-187635] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.h, channels/chan_misdn.c,
+ channels/misdn/isdn_lib.c: Miscellaneous minor changes to
+ chan_misdn. * Miscellaneous spacing and comment changes. * Minor
+ code rearangements. * Miscellaneous doxygen comments.
+
+ * channels/chan_misdn.c: Make chan_misdn_log() avoid generating the
+ log message if logging is disabled.
+
+2009-04-10 03:55 +0000 [r187599] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, main/pbx.c, main/manager.c,
+ include/asterisk/linkedlists.h, main/features.c, main/http.c,
+ main/app.c, include/asterisk/lock.h, main/audiohook.c,
+ main/bridging.c: Modify headers and macros, according to
+ Russell's suggestions on the -dev list
+
+2009-04-09 21:06 +0000 [r187560] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: Add a new option,
+ mwi_from, to sip.conf. This allows for you to change the From
+ header for outgoing MWI NOTIFY requests. Prior to this, the best
+ you could do was to set a callerid in the general section of
+ sip.conf. The problem was that this was used for all outbound
+ requests, not just MWI NOTIFY requests. AST-201
+
+2009-04-09 20:40 +0000 [r187556] David Vossel <dvossel@digium.com>
+
+ * apps/app_dial.c: More changes concerning r187426. Revised where
+ locks are placed.
+
+2009-04-09 19:10 +0000 [r187491] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h, CHANGES: Add
+ ability for dialplan execution to continue when caller hangs up.
+ The F option to app_dial has been modified to accept no
+ parameters and perform the above functionality. I don't see
+ anywhere else that is doing function overloading, but this really
+ is the best place for this operation because: - It makes it close
+ to the 'g' option in the argument list which provides similar
+ functionality. - The existing code to support the current F
+ option provides a very convienient location to add this new
+ feature. (closes issue #12381) Reported by: michael-fig
+
+2009-04-09 18:58 +0000 [r187488] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 187484 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r187484 | mmichelson | 2009-04-09 13:51:20 -0500 (Thu, 09 Apr
+ 2009) | 18 lines Handle a SIP race condition (reinvite before an
+ ACK) properly. RFC 5047 explains the proper course of action to
+ take if a reINVITE is received before the ACK from a previous
+ invite transaction. What we are to do is to treat the reINVITE as
+ if it were both an ACK and a reINVITE and process it normally.
+ Later, when we receive the ACK we had been expecting, we will
+ ignore it since its CSeq is less than the current iseqno of the
+ sip_pvt representing this dialog. (closes issue #13849) Reported
+ by: klaus3000 Patches: 13849_v2.patch uploaded by mmichelson
+ (license 60) Tested by: mmichelson, klaus3000 ........
+
+2009-04-09 18:40 +0000 [r187483] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, /, include/asterisk/linkedlists.h,
+ include/asterisk/lock.h: Merged revisions 187428 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09
+ Apr 2009) | 8 lines Race condition between ast_cli_command() and
+ 'module unload' could cause a deadlock. Add lock timeouts to
+ avoid this potential deadlock. (closes issue #14705) Reported by:
+ jamessan Patches: 20090320__bug14705.diff.txt uploaded by
+ tilghman (license 14) Tested by: jamessan ........
+
+2009-04-09 17:39 +0000 [r187426] David Vossel <dvossel@digium.com>
+
+ * apps/app_dial.c: Fixes deadlock caused by calling get_cid_name
+ with chan locked. get_cid_name should not be called with a
+ channel lock. get_cid_name calls ast_get_hint which eventually
+ calls pbx_find_extension. pbx_find_extension starts and stops
+ autoservice which should not be done with a channel lock, so
+ get_cid_name should not be called with one.
+
+2009-04-09 17:34 +0000 [r187421-187424] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_musiconhold.c: Use safe macro practices even though they
+ really aren't necessary.
+
+ * res/res_musiconhold.c: Fix a crash in res_musiconhold when using
+ cached realtime moh. The moh_register function links an mohclass
+ and then immediately unrefs the class since the container now has
+ a reference. The problem with using realtime music on hold is
+ that the class is allocated, registered, and started in one fell
+ swoop. The refcounting logic resulted in the count being off by
+ one. The same problem did not happen when using a static config
+ because the allocation and registration of an mohclass is a
+ separate operation from starting moh. This also did not affect
+ non-cached realtime moh because the classes are not registered at
+ all. I also have modified res_musiconhold to use the _t_ variants
+ of the ao2_ functions so that more info can be gleaned when
+ attempting to trace the refcounts. I found this to be incredibly
+ helpful for debugging this issue and there's no good reason to
+ remove it. (closes issue #14661) Reported by: sum
+
+2009-04-09 17:20 +0000 [r187363-187381] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Allow '/' in username portion of register;
+ this is a regression. (closes issue #14668) Reported by: Netview
+
+ * /, channels/chan_sip.c, apps/app_sendtext.c: Merged revisions
+ 187362 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r187362 | tilghman | 2009-04-09 11:38:37 -0500 (Thu, 09 Apr 2009)
+ | 3 lines Permit zero-length text messages in SIP. (Related to an
+ issue posted to the -users list, subject "AEL2, BASE64_DECODE and
+ hexadecimal") ........
+
+2009-04-09 16:27 +0000 [r187360-187361] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: Do not try to send the format read/format
+ write/make compatible options over IAX2.
+
+ * main/channel.c, channels/chan_sip.c, include/asterisk/frame.h:
+ Add support for allowing the channel driver to handle
+ transcoding. This was accomplished using a set of options and the
+ setoption channel callback. The core calls into the channel
+ driver using these options and the channel driver either returns
+ success or failure.
+
+2009-04-09 04:59 +0000 [r187302] Tilghman Lesher <tlesher@digium.com>
+
+ * agi/Makefile, build_tools/cflags.xml, utils/Makefile,
+ include/asterisk.h, /, main/Makefile, main/file.c, main/astfd.c
+ (added), main/asterisk.c: Merged revisions 187300-187301 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009)
+ | 3 lines Add debugging mode for diagnosing file descriptor
+ leaks. (Related to issue #14625) ........ r187301 | tilghman |
+ 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines Oops,
+ missed this file in the last commit. ........
+
+2009-04-09 02:44 +0000 [r187269] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/logger.h, main/logger.c, apps/app_verbose.c,
+ configs/logger.conf.sample: add a dedicated log channel for
+ modules to be able report security-related events, so that they
+ can be fed into external processes for analysis and possible
+ mitigation efforts (inspired by this evening's Toronto Asterisk
+ Users Group meeting and previous dicussions amongst various
+ community members)
+
+2009-04-08 21:00 +0000 [r187211] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, main/features.c, include/asterisk/channel.h: Add
+ timer for features so that backup bridge config can go away The
+ biggest change done here was elimination of the backup_config for
+ use with features. Previously, the bridging code upon detecting a
+ feature would set the start time of the bridge to the start time
+ of the feature. Then after the feature had either expired or
+ timed out the start time would be reset to the true bridge start
+ time from the backup_config. Now, the time differences are
+ calculated with respect to the newly added feature_start_time
+ timeval instead. There should be no behavior changes from the
+ previous functionality aside from the bridge timing being
+ unaffected by either valid or partial feature matches. Previously
+ the timing would be increased by the length of time configured
+ for featuredigittimeout, which was probably never noticed.
+ (closes issue #14503) Reported by: KNK Tested by: jpeeler Review:
+ http://reviewboard.digium.com/r/179/
+
+2009-04-08 20:39 +0000 [r187210] Tilghman Lesher <tlesher@digium.com>
+
+ * /: Recorded merge of revisions 187209 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r187209 | tilghman | 2009-04-08 15:39:13 -0500 (Wed, 08 Apr 2009)
+ | 4 lines Backport resolution for file descriptor leak in 1.6.0
+ to 1.4. This fixes short reads in http manager sessions, such as
+ those done by the ast-gui branch. (Fixes AST-198) ........
+
+2009-04-08 19:59 +0000 [r187179] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/doxyref.h,
+ include/asterisk/doxygen/reviewboard.h (added): Add documentation
+ for reviewboard usage and guidelines.
+
+2009-04-08 18:12 +0000 [r187108] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp_engine.c: Fix a bug where we would native bridge when we
+ did not want to.
+
+2009-04-08 17:51 +0000 [r187105] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Remove duplicate prototype for temp_peer().
+
+2009-04-08 17:08 +0000 [r187050] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_odbc.c: If the first column is empty, output a
+ delimiter anyway. (closes issue #14848) Reported by: john8675309
+ Patches: 20090408__bug14848.diff.txt uploaded by tilghman
+ (license 14) Tested by: john8675309
+
+2009-04-08 16:52 +0000 [r187046] Mark Michelson <mmichelson@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 187045 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed,
+ 08 Apr 2009) | 10 lines Fix a small logical error when loading
+ moh classes. We were unconditionally incrementing the number of
+ mohclasses registered. However, we should actually only increment
+ if the call to moh_register was successful. While this probably
+ has never caused problems, I noticed it and decided to fix it
+ anyway. ........
+
+2009-04-08 16:27 +0000 [r187036] Joshua Colp <jcolp@digium.com>
+
+ * res/res_rtp_asterisk.c, main/rtp_engine.c: Turn a warning message
+ into a debug message and do not treat two situations as errors
+ when they are not.
+
+2009-04-08 15:27 +0000 [r186985] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 186984 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr
+ 2009) | 24 lines Make a couple of changes with regards to a new
+ message printed in ast_read(). "ast_read() called with no
+ recorded file descriptor" is a new message added after a bug was
+ discovered. Unfortunately, it seems there are a bunch of places
+ that potentially make such calls to ast_read() and trigger this
+ error message to be displayed. This commit does two things to
+ help to make this message appear less. First, the message has
+ been downgraded to a debug level message if dev mode is not
+ enabled. The message means a lot more to developers than it does
+ to end users, and so developers should take an effort to be sure
+ to call ast_read only when a channel is ready to be read from.
+ However, since this doesn't actually cause an error in operation
+ and is not something a user can easily fix, we should not spam
+ their console with these messages. Second, the message has been
+ moved to after the check for any pending masquerades. ast_read()
+ being called with no recorded file descriptor should not
+ interfere with a masquerade taking place. This could be seen as a
+ simple way of resolving issue #14723. However, I still want to
+ try to clear out the existing ways of triggering this message,
+ since I feel that would be a better resolution for the issue.
+ ........
+
+2009-04-08 13:38 +0000 [r186928-186957] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/doxygen/releases.h: Add some additional notes on
+ release numbering.
+
+ * Makefile, include/asterisk/doxygen/releases.h (added),
+ include/asterisk/doxyref.h, contrib/asterisk-ng-doxygen,
+ include/asterisk/doxygen (added),
+ include/asterisk/doxygen/commits.h (added),
+ include/asterisk/doxygen/licensing.h (added), main/asterisk.c:
+ Start splitting up miscellaneous doxygen documentation into
+ separate files. doxyref.h was created to hold miscellaneous
+ documentation that was not specific to a part of the code. This
+ file has grown quite a bit so I decided to start splitting parts
+ of it out into new files. Now, you can drop a new file into
+ include/asterisk/doxygen/ and it will be processed by doxygen.
+
+ * channels/chan_sip.c: Update some comments and resolve potential
+ memory corruption in chan_sip. While browsing chan_sip the other
+ day, I noticed this dangerous code in dialog_needdestroy(). This
+ function is an ao2_callback. It is absolutely _not_ okay to
+ unlock the container from within this function. It's also not
+ clear why it was useful. Given that it could cause memory
+ corruption, I have removed it. There was also a TODO comment left
+ describing a potential implementation of an improvement to the
+ needdestroy handling. I'm not convinced that what was described
+ is the best choice here, so I have briefly described the way that
+ this function is used today that could be improved.
+
+2009-04-08 05:06 +0000 [r186899] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Add lastms to the require API call.
+
+2009-04-08 00:09 +0000 [r186833-186842] Mark Michelson <mmichelson@digium.com>
+
+ * /, formats/format_wav.c, formats/format_wav_gsm.c: Merged
+ revisions 186841 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr
+ 2009) | 8 lines Fix a few typos of the word "frequency." (closes
+ issue #14842) Reported by: jvandal Patches: frequency-typo.diff
+ uploaded by jvandal (license 413) ........
+
+ * channels/chan_sip.c: Fix bad merge from fix for issue 13867.
+ (closes issue #14686) Reported by: davidw
+
+ * main/channel.c, /: Merged revisions 186832 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr
+ 2009) | 8 lines Set the AST_FEATURE_WARNING_ACTIVE flag when a
+ p2p bridge returns AST_BRIDGE_RETRY. Without this flag set,
+ warning sounds will not be properly played to either party of the
+ bridge. (closes issue #14845) Reported by: adomjan ........
+
+2009-04-07 22:23 +0000 [r186799] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_macro.c: Merged revisions 186775 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186775 | tilghman | 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009)
+ | 3 lines Fix Macro documentation to match current (and intended)
+ behavior. (See -dev mailing list) ........
+
+2009-04-07 20:46 +0000 [r186720] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c, /: Merged revisions 186719 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186719 | mmichelson | 2009-04-07 15:43:49 -0500 (Tue, 07 Apr
+ 2009) | 6 lines Ensure that \r\n is printed after the ActionID in
+ an OriginateResponse. (closes issue #14847) Reported by: kobaz
+ ........
+
+2009-04-06 23:11 +0000 [r186624-186687] Joshua Colp <jcolp@digium.com>
+
+ * res/res_rtp_asterisk.c: Fix a log message getting output when it
+ should not have been.
+
+ * channels/chan_sip.c: Fix problem when authenticating a non-RTP
+ dialog.
+
+ * channels/chan_sip.c, doc/tex/channelvariables.tex, CHANGES: Add
+ support for changing the outbound codec on a SIP call using a
+ dialplan variable. This adds a dialplan variable
+ (SIP_CODEC_OUTBOUND) which controls the codec offered for an
+ outgoing SIP call. This is much like the SIP_CODEC dialplan
+ variable and has the same restrictions. The codec set must be one
+ that is configured for the call. (closes issue #13243) Reported
+ by: samdell3 Patches: 13243.diff uploaded by file (license 11)
+
+2009-04-06 16:06 +0000 [r186620] Mark Michelson <mmichelson@digium.com>
+
+ * funcs/func_connectedline.c (added), funcs/func_redirecting.c
+ (added): Silly svn. These files didn't get merged over in the
+ merge of the issue8824 branch.
+
+2009-04-06 13:23 +0000 [r186563] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp_engine.c: Pass the correct value to sizeof when copying
+ address information. (issue #14827) Reported by: pj Patches:
+ 14827.diff uploaded by file (license 11) Tested by: pj
+
+2009-04-04 00:13 +0000 [r186537] Richard Mudgett <rmudgett@digium.com>
+
+ * /: Remove merged branch properties accidentally merged to trunk.
+
+2009-04-03 22:41 +0000 [r186525] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_unistim.c, channels/misdn/isdn_lib_intern.h,
+ channels/chan_local.c, main/rtp_engine.c, /,
+ channels/misdn/isdn_msg_parser.c, channels/chan_iax2.c,
+ channels/misdn/isdn_lib.c, channels/misdn_config.c,
+ include/asterisk/callerid.h, main/channel.c, main/dial.c,
+ channels/misdn/isdn_lib.h, channels/chan_dahdi.c,
+ channels/chan_phone.c, channels/chan_skinny.c, main/features.c,
+ configs/sip.conf.sample, include/asterisk/frame.h,
+ include/asterisk/rtp_engine.h, channels/chan_mgcp.c,
+ apps/app_dial.c, res/res_rtp_asterisk.c, main/stun.c,
+ channels/chan_sip.c, channels/chan_agent.c,
+ configs/misdn.conf.sample, include/asterisk/channel.h, CHANGES,
+ apps/app_queue.c, channels/chan_misdn.c,
+ apps/app_directed_pickup.c, channels/misdn/chan_misdn_config.h,
+ channels/chan_h323.c, main/callerid.c, include/asterisk/stun.h:
+ This commit introduces COLP/CONP and Redirecting party
+ information into Asterisk. The channel drivers which have been
+ most heavily tested with these enhancements are chan_sip and
+ chan_misdn. Further work is being done to add Q.SIG support and
+ will be introduced in a later commit. chan_skinny has code added
+ to it here, but according to user pj, the support on chan_skinny
+ is not working as of now. This will be fixed in a later commit. A
+ special thanks goes out to bugtracker user gareth for getting the
+ ball rolling and providing the initial support for this work.
+ Without his initial work on this, this would not have been nearly
+ as painless as it was. This functionality has been tested by
+ Digium's product quality department, as well as a customer site
+ running thousands of calls every day. In addition, many many many
+ many bugtracker users have tested this, too. (closes issue #8824)
+ Reported by: gareth Review: http://reviewboard.digium.com/r/201
+
+2009-04-03 20:20 +0000 [r186461] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 186458 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03
+ Apr 2009) | 5 lines Fix a bug where DAHDI/Zaptel channels would
+ not properly switch formats when requested Don't offer
+ AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could
+ provide a slight performance benefit, the translation core in
+ Asterisk has some flaws when a channel driver offers multiple raw
+ formats. this fix is much simpler than fixing the translation
+ core to solve that issue (although that will be done later).
+ ........
+
+2009-04-03 19:59 +0000 [r186444-186447] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 186445 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r186445 | tilghman | 2009-04-03 14:56:48 -0500 (Fri, 03
+ Apr 2009) | 2 lines Found a conflict in the last commit, due to
+ multiple targets ........
+
+ * /, configs/voicemail.conf.sample, apps/app_voicemail.c: Merged
+ revisions 186415 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009)
+ | 7 lines Distinguish in a sent email between simple sends and
+ forwards. (closes issue #11678) Reported by: jamessan Patches:
+ 20090330__bug11678.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman, lmadsen ........
+
+2009-04-03 16:47 +0000 [r186382] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c, channels/chan_sip.c, channels/chan_iax2.c,
+ include/asterisk/frame.h: Add better support for relaying success
+ or failure of the ast_transfer() API call. This API call now
+ waits for a special frame from the underlying channel driver to
+ indicate success or failure. This allows the return value to
+ truly convey whether the transfer worked or not. In the case of
+ the Transfer() dialplan application this means the value of the
+ TRANSFERSTATUS dialplan variable is actually true. (closes issue
+ #12713) Reported by: davidw Tested by: file
+
+2009-04-03 16:29 +0000 [r186379] David Vossel <dvossel@digium.com>
+
+ * main/audiohook.c: audio_audiohook_write_list() did not correctly
+ update sample size after ast_translate.
+ audio_audiohook_write_list() did not take into account that the
+ sample size may change after translation depending on if the
+ original frame is is 8khz or 16khz. the sample size is now
+ updated after translating to reflect this possibility. This
+ caused the audio on the receiving end to sound terrible. Thanks
+ to jcolp and mmichelson for helping me work this out. (issue
+ AST-197)
+
+2009-04-03 15:52 +0000 [r186321] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/crypto.h, /: Merged revisions 186320 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5
+ lines Fix a problem with the crypto variable definitions not
+ actually being defined properly. (closes issue #14804) Reported
+ by: jvandal ........
+
+2009-04-03 15:18 +0000 [r186297] Tilghman Lesher <tlesher@digium.com>
+
+ * main/stdtime/localtime.c: Compatibility fix for glibc 2.4 (Closes
+ issue #14820) Reported by: phsultan
+
+2009-04-03 14:32 +0000 [r186286] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Fix the ability to retrieve voicemail
+ messages from IMAP. A recent change made interactive vm_states no
+ longer get added to the list of vm_states and instead get stored
+ in thread-local storage. In trunk and all the 1.6.X branches, the
+ problem is that when we search for messages in a voicemail box,
+ we would attempt to update the appropriate vm_state struct by
+ directly searching in the list of vm_states instead of using the
+ get_vm_state_by_imap_user function. This meant we could not find
+ the interactive vm_state that we wanted. (closes issue #14685)
+ Reported by: BlargMaN Patches: 14685.patch uploaded by mmichelson
+ (license 60) Tested by: BlargMaN, qualleyiv, mmichelson
+
+2009-04-03 02:03 +0000 [r186230] Russell Bryant <russell@digium.com>
+
+ * /, cdr/cdr_radius.c: Merged revisions 186229 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009)
+ | 21 lines Fix a memory leak in cdr_radius. I came across this
+ while doing some testing of my ast_channel_ao2 branch. After
+ running a test overnight that generated over 5 million calls,
+ Asterisk had taken up about 1 GB of my system memory. So, I
+ re-ran the test with MALLOC_DEBUG turned on. However, it showed
+ no leaks in Asterisk during the test, even though Asterisk was
+ still consuming it somehow. Instead, I turned to valgrind, which
+ when run with --leak-check=full, told me exactly where the leak
+ came from, which was from allocations inside the radiusclient-ng
+ library. This explains why MALLOC_DEBUG did not report it. After
+ a bit of analysis, I found that we were leaking a little bit of
+ memory every time a CDR record was passed to cdr_radius. I don't
+ actually have a radius server set up to receive CDR records.
+ However, I always have my development systems compile and install
+ all modules. In addition to making sure there are not build
+ errors across modules, always loading modules helps find bugs
+ like this, too, so it is strongly recommend for all developers.
+ ........
+
+2009-04-02 21:56 +0000 [r186175] Mark Michelson <mmichelson@digium.com>
+
+ * /, configs/features.conf.sample: Merged revisions 186174 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr
+ 2009) | 5 lines Fix instructions in one-step parking comment to
+ make more sense. Changed a capital K to a lowercase k. ........
+
+2009-04-02 17:26 +0000 [r186101] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 186081 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02
+ Apr 2009) | 3 lines ensure that the buffer passed to
+ DAHDI_SET_BUFINFO is fully initialized ........
+
+2009-04-02 17:20 +0000 [r186078] Joshua Colp <jcolp@digium.com>
+
+ * res/res_rtp_asterisk.c (added), channels/chan_unistim.c,
+ apps/app_dial.c, main/stun.c (added), main/rtp_engine.c (added),
+ channels/chan_local.c, channels/chan_sip.c,
+ channels/chan_bridge.c, main/Makefile, channels/chan_agent.c,
+ include/asterisk/rtp.h (removed), UPGRADE.txt,
+ channels/chan_gtalk.c, include/asterisk/_private.h, main/rtp.c
+ (removed), main/loader.c, channels/chan_jingle.c,
+ channels/chan_skinny.c, channels/chan_h323.c,
+ configs/sip.conf.sample, include/asterisk/stun.h (added),
+ include/asterisk/rtp_engine.h (added), main/asterisk.c,
+ channels/chan_mgcp.c: Merge in the RTP engine API. This API
+ provides a generic way for multiple RTP stacks to be integrated
+ into Asterisk. Right now there is only one present,
+ res_rtp_asterisk, which is the existing Asterisk RTP stack.
+ Functionality wise this commit performs the same as previously.
+ API documentation can be viewed in the rtp_engine.h header file.
+ Review: http://reviewboard.digium.com/r/209/
+
+2009-04-02 17:10 +0000 [r186021-186060] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+ 186059 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500
+ (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02
+ Apr 2009) | 2 lines Fix for AST-2009-003 ........
+ ................
+
+ * main/strings.c: Missed a common case for needing to extend the
+ buffer. (closes issue #14716) Reported by: sum Patches:
+ 20090402__bug14716.diff.txt uploaded by tilghman (license 14)
+ Tested by: sum
+
+2009-04-02 13:51 +0000 [r185953] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 185952 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02
+ Apr 2009) | 5 lines the DAHDI_GETCONF, DAHDI_SETCONF and
+ DAHDI_GET_PARAMS ioctls were recently corrected to show that they
+ do, in fact, read data from userspace as part of their work. due
+ to this fix, valgrind now reports a number of cases where
+ chan_dahdi passed an uninitialized (or partially) buffer to these
+ ioctls, which could lead to unexpected behavior. this patch
+ corrects chan_dahdi to ensure that buffers passed to these ioctls
+ are always fully initialized. ........
+
+2009-04-01 20:13 +0000 [r185912] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/res_odbc.h, include/asterisk.h, main/strings.c,
+ main/manager.c, main/tdd.c, include/asterisk/astobj2.h,
+ main/ast_expr2f.c, include/asterisk/pbx.h,
+ include/asterisk/strings.h, main/taskprocessor.c, res/res_odbc.c:
+ Merge changes from str_substitution that are unrelated to that
+ branch. Included is a small bugfix to an ast_str helper, but most
+ of these changes are simply doxygen fixes.
+
+2009-04-01 19:03 +0000 [r185846] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 185845 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009)
+ | 10 lines Fixes issue with dropped calles due to re-Invite glare
+ and re-Invites never executing after a 491 Acknowledgement for
+ 491 responses were never being processed because it didn't match
+ our pending invite's seqno. Since the ACK was never processed,
+ the 491 frame would continue to be retransmitted until eventually
+ the call was dropped due to max retries. Now during a pending
+ invite, if we receive another invite, we send an 491 and hold on
+ to that glare invite's seqno in the "glareinvite" variable for
+ that sip_pvt struct. When ACK's are received, we first check to
+ see if it is in response to our pending invite, if not we check
+ to see if it is in response to a glare invite. In this case, it
+ is in response to the glare invite and must be dealt with or the
+ call is dropped. I've changed the wait time for resending the
+ re-Invite after receving a 491 response to comply with RFC 3261.
+ Before this patch the scheduled re-Invite would only change a
+ flag indicating that the re-Invite should be sent out, now it
+ actually sends it out as well. (closes issue #12013) Reported by:
+ alx Review: http://reviewboard.digium.com/r/213/ ........
+
+2009-04-01 13:59 +0000 [r185777] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c: Address Russell's comments regarding rev 185704.
+ Use ast_debug and ast_softhangup_nolock.
+
+2009-04-01 13:48 +0000 [r185741-185772] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 185771 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009)
+ | 6 lines Fix a case where DTMF could bypass audiohooks. This
+ change fixes a situation where an audiohook that wants DTMF would
+ not actually get it. This is in the code path where we end DTMF
+ digit length emulation while handling a NULL frame. ........
+
+ * include/asterisk/stringfields.h: Fix dev-mode build on my box.
+
+2009-04-01 00:39 +0000 [r185704] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c, CHANGES: Allow the AMI Hangup command to accept a
+ Cause header. (closes issue #14695) Reported by: mneuhauser
+ Patches: cause-for-hangup-manager-action.patch uploaded by
+ mneuhauser (license 425)
+
+2009-03-31 22:35 +0000 [r185664] Kevin P. Fleming <kpfleming@digium.com>
+
+ * utils: ignore copied (generated) file
+
+2009-03-31 22:12 +0000 [r185600-185604] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix trunk's compilation.
+
+ * /, apps/app_queue.c: Merged revisions 185599 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185599 | mmichelson | 2009-03-31 17:00:01 -0500 (Tue, 31 Mar
+ 2009) | 6 lines Fix crash that would occur if an empty member was
+ specified in queues.conf. (closes issue #14796) Reported by: pida
+ ........
+
+2009-03-31 21:29 +0000 [r185581] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/utils.c, include/asterisk/stringfields.h: Optimizations to
+ the stringfields API This patch provides a number of
+ optimizations to the stringfields API, focused around saving (not
+ wasting) memory whenever possible. Thanks to Mark Michelson for
+ inspiring this work and coming up with the first two
+ optimizations that are represented here: Changes: - Cleanup of
+ some code, fix incorrect doxygen comments - When a field is
+ emptied or replaced with a new allocation, decrease the amount of
+ 'active' space in the pool it was held in; if that pool reaches
+ zero active space, and is not the current pool, then free it as
+ it is no longer in use - When allocating a pool, try to allocate
+ a size that will fit in a 'standard' malloc() allocation without
+ wasting space - When allocating space for a field, store the
+ amount of space in the two bytes immediately preceding the field;
+ this eliminates the need to call strlen() on the field when
+ overwriting it, and more importantly it 'remembers' the amount of
+ space the field has available, even if a shorter string has been
+ stored in it since it was allocated - Don't automatically double
+ the size of each successive pool allocated; it's wasteful
+ http://reviewboard.digium.com/r/165/
+
+2009-03-31 19:46 +0000 [r185469] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 185468 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r185468 | mmichelson | 2009-03-31 14:45:30 -0500 (Tue,
+ 31 Mar 2009) | 8 lines Fix Russian voicemail intro to say the
+ word "messages" properly. (closes issue #14736) Reported by:
+ chappell Patches: voicemail_no_messages.diff uploaded by chappell
+ (license 8) ........
+
+2009-03-31 19:07 +0000 [r185432] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Improve performance of the code handling
+ the frame queue in chan_iax2. In my tests that exercised full
+ frame handling in chan_iax2, the version with these changes took
+ 30% to 40% of the CPU time compared to the same test of Asterisk
+ trunk before these modifications. While doing some profiling for
+ <http://reviewboard.digium.com/r/205/>, one function that caught
+ my eye was network_thread() in chan_iax2.c. After the things that
+ I was working on there, it was the next target for analysis and
+ optimization. I used oprofile's source annotation functionality
+ and found that the loop traversing the frame queue in
+ network_thread() was to blame for the excessive CPU cycle
+ consumption. The frame_queue in chan_iax2 previously held all
+ frames that either were pending transmission or had been
+ transmitted and are still pending acknowledgment. In
+ network_thread(), the previous code would go back through the
+ main for loop after reading a single incoming frame or after
+ being signaled because a frame had been queued up for initial
+ transmission. In each iteration of the loop, it traverses the
+ entire frame queue looking for frames that need to be
+ transmitted. On a busy server, this could easily be quite a few
+ entries. This patch is actually quite simple. The frame_queue has
+ become only a list of frames pending acknowledgment. Frames that
+ need to be transmitted are queued up to a dedicated transmit
+ thread via the taskprocessor API. As a result, the code in
+ network_thread() becomes much simpler, as its only job is to read
+ incoming frames. In addition to the previously described changes,
+ this patch includes some additional changes to the frame_queue.
+ Instead of one big frame_queue, now there is a list per call
+ number to further reduce wasted list traversals. The biggest
+ impact of this change is in socket_process(). For additional
+ details on testing and test results, see the review request.
+ Review: http://reviewboard.digium.com/r/212/
+
+2009-03-31 16:46 +0000 [r185363] David Brooks <dbrooks@digium.com>
+
+ * /, channels/chan_gtalk.c: Merged revisions 185362 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31
+ Mar 2009) | 35 lines Fix incorrect parsing in chan_gtalk when
+ xmpp contains extra whitespaces To drill into the xmpp to find
+ the capabilities between channels, chan_gtalk calls iks_child()
+ and iks_next(). iks_child() and iks_next() are functions in the
+ iksemel xml parsing library that traverse xml nodes. The bug here
+ is that both iks_child() and iks_next() will return the next
+ iks_struct node *regardless* of type. chan_gtalk expects the next
+ node to be of type IKS_TAG, which in most cases, it is, but in
+ this case (a call being made from the Empathy IM client), there
+ exists iks_struct nodes which are not IKS_TAG data (they are
+ extraneous whitespaces), and chan_gtalk doesn't handle that case,
+ so capabilities don't match, and a call cannot be made.
+ iks_first_tag() and iks_next_tag(), on the other hand, will not
+ return the very next iks_struct, but will check to see if the
+ next iks_struct is of type IKS_TAG. If it isn't, it will be
+ skipped, and the next struct of type IKS_TAG it finds will be
+ returned. This assures that chan_gtalk will find the iks_struct
+ it is looking for. This fix simply changes all calls to
+ iks_child() and iks_next() to become calls to iks_first_tag() and
+ iks_next_tag(), which resolves the capability matching. The
+ following is a payload listing from Empathy, which, due to the
+ extraneous whitespace, will not be parsed correctly by iksemel:
+ <iq from='dbrooksjab@235-22-24-10/Telepathy'
+ to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'>
+ <session xmlns='http://www.google.com/session'
+ initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate'
+ id='1837267342'> <description
+ xmlns='http://www.google.com/session/phone'> <payload-type
+ clockrate='16000' name='speex' id='96'/> <payload-type
+ clockrate='8000' name='PCMA' id='8'/> <payload-type
+ clockrate='8000' name='PCMU' id='0'/> <payload-type
+ clockrate='90000' name='MPA' id='97'/> <payload-type
+ clockrate='16000' name='SIREN' id='98'/> <payload-type
+ clockrate='8000' name='telephone-event' id='99'/> </description>
+ </session> </iq> Review: http://reviewboard.digium.com/r/181/
+ ........
+
+2009-03-31 14:53 +0000 [r185261] Russell Bryant <russell@digium.com>
+
+ * apps/app_queue.c: Don't free() an astobj2 object. (closes issue
+ #14672) Reported by: makoto
+
+2009-03-31 14:07 +0000 [r185197] Joshua Colp <jcolp@digium.com>
+
+ * /, main/audiohook.c: Merged revisions 185196 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8
+ lines Fix crash when moving audiohooks between channels. Handle
+ the scenario where we are called to move audiohooks between
+ channels and the source channel does not actually have any on it.
+ (closes issue #14734) Reported by: corruptor ........
+
+2009-03-30 20:42 +0000 [r185122-185123] Richard Mudgett <rmudgett@digium.com>
+
+ * /, configs/misdn.conf.sample, channels/misdn_config.c: Merged
+ revisions 185121 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009)
+ | 1 line Update the channel allocation method documentation.
+ ........
+
+ * /, channels/misdn/isdn_lib.c: Merged revisions 185120 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009)
+ | 19 lines Make chan_misdn BRI TE side normally defer channel
+ selection to the NT side. Channel allocation collisions are not
+ handled by chan_misdn very well. This patch simply avoids the
+ problem for BRI only. For PRI, allocation collisions are still
+ possible but less likely since there are simply more channels
+ available and each end could use a different allocation strategy.
+ misdn.conf options available: te_choose_channel - Use to force
+ the TE side to allocate channels. method - Specify the channel
+ allocation strategy. (closes issue #13488) Reported by:
+ Christian_Pinedo Patches: isdn_lib.patch.txt uploaded by crich
+ Tested by: crich, siepkes, festr ........
+
+2009-03-30 16:26 +0000 [r185072] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 185031 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar
+ 2009) | 39 lines Fix queue weight behavior so that calls in
+ low-weight queues are not inappropriately blocked. (This is
+ copied and pasted from the review request I made for this patch)
+ Asterisk has some odd behavior when queue weights are used. The
+ current logic used when potentially calling a queue member is: If
+ the member we are going to call is part of another queue and
+ _that other queue has any callers in it_ and has a higher weight
+ than the queue we are calling from, then don't try to contact
+ that member. The issue here is what I have marked with
+ underscores. If the higher-weighted queue has any callers in it
+ at all, then the queue member will be unreachable from the
+ lower-weighted queue. This has the potential to be really really
+ bad if using a queue strategy, such as leastrecent or
+ fewestcalls, with the potential to call the same member
+ repeatedly. The fix proposed by garychen on issue 13220 is very
+ simple and, as far as I can see, works well for this situation.
+ With this set of changes, the logic used becomes: If the member
+ we are going to call is part of another queue, the other queue
+ has a higher weight than the queue we are calling from, and the
+ higher weight queue has at least as many callers as available
+ members, then do not try to contact the queue member. If the
+ higher weighted queue has fewer callers than available members,
+ then there is no reason to deny the call to this member since the
+ other queue can afford to spare a member. Since the fix involved
+ writing a generic function for determining the number of
+ available members in the queue, I also modified the is_our_turn
+ function to make use of the new num_available_members function to
+ determine if it is our turn to try calling a member. There is one
+ small behavior change. Before writing this patch, if you had
+ autofill disabled, then if you were the head caller in a queue,
+ you would automatically be told that it was your turn to try
+ calling a member. This did not take into account whether there
+ were actually any queue members available to take the call. Now
+ we actually make sure there is at least one member available to
+ take the call if autofill is disabled. (closes issue #13220)
+ Reported by: garychen Review:
+ http://reviewboard.digium.com/r/202/ ........
+
+2009-03-30 14:37 +0000 [r184948] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 184947 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) |
+ 14 lines Improve our handling of T38 in the initial INVITE from a
+ device. We now answer with matching media streams to what is
+ requested. If an INVITE is received with both a T38 and RTP media
+ stream this means we answer with both. For any outgoing calls
+ created as a result of this inbound one no T38 is requested in
+ the initial INVITE. Instead if we start receiving udptl packets
+ we trigger a reinvite on the outbound side. (closes issue #12437)
+ Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu
+ Review: http://reviewboard.digium.com/r/208/ ........
+
+2009-03-30 13:55 +0000 [r184910] Russell Bryant <russell@digium.com>
+
+ * channels/h323/Makefile.in: Fix build error when chan_h323 is not
+ being built. (reported by cai1982 in #asterisk-dev)
+
+2009-03-29 05:52 +0000 [r184838-184843] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_followme.c: Merged revisions 184842 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009)
+ | 5 lines Ensure targs variable is fully initialized. (closes
+ issue #14758) Reported by: tim_ringenbach ........
+
+ * channels/Makefile: Simplify chan_h323 build to not require a
+ second run of "make". (closes issue #14715) Reported by: jthurman
+ Patches: h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman
+ (license 614) Tested by: tzafrir, russell
+
+2009-03-27 20:08 +0000 [r184798-184801] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_ices.c: Fix a typo in app_ices. (closes issue #14765)
+ Reported by: timeshell Patches: app_ices.svn-1.6.0.diff uploaded
+ by timeshell (license 399)
+
+ * include/asterisk/doxyref.h: Update commit message guidelines in
+ re: to punctuation. The doxygen documentation has now been
+ updated to state explicitly that I want punctuation atthe end of
+ the first sentence in a commit message. :).
+
+2009-03-27 19:10 +0000 [r184762] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/channel.c, bridges/bridge_softmix.c,
+ include/asterisk/timing.h, include/asterisk/channel.h,
+ channels/chan_iax2.c, main/timing.c: Improve timing interface to
+ remember which provider provided a timer The ability to
+ load/unload timing interfaces is nice, but it means that when a
+ timer is allocated, it may come from provider A, but later
+ provider B becomes the 'preferred' provider. If this happens, all
+ timer API calls on the timer that was provided by provider A will
+ actually be handed to provider B, which will say WTF and return
+ an error. This patch changes the timer API to include a pointer
+ to the provider of the timer handle so that future operations on
+ the timer will be forwarded to the proper provider. (closes issue
+ #14697) Reported by: moy Review:
+ http://reviewboard.digium.com/r/211/
+
+2009-03-27 18:04 +0000 [r184693-184726] Russell Bryant <russell@digium.com>
+
+ * main/manager.c, apps/app_minivm.c: Use ast_random() instead of
+ rand() to ensure we use the best RNG available.
+
+ * include/asterisk/app.h, apps/app_dumpchan.c, main/app.c,
+ apps/app_queue.c, apps/app_voicemail.c, main/cli.c: Change
+ global_app_buf to ast_str_thread_global_buf.
+
+2009-03-27 15:57 +0000 [r184639-184677] Joshua Colp <jcolp@digium.com>
+
+ * bridges/bridge_softmix.c: Fix a potential timer leak in
+ bridge_softmix. It is possible for a bridge to be created without
+ actually being used. In that scenario a timing file descriptor
+ would be opened and not closed. To fix this the timing file
+ descriptor is now closed in the destroy callback, not the thread
+ function.
+
+ * res/res_agi.c: Fix speech structure leak in the AGI speech
+ recognition integration. The AGI dialplan applications did not
+ destroy the speech structure automatically if it was not
+ destroyed by the running AGI script. They will now do this.
+ (issue LUMENVOX-15)
+
+ * bridges/bridge_softmix.c: Remove a cast that is not needed.
+
+2009-03-27 14:00 +0000 [r184630] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/utils.h, main/pbx.c, res/ais/evt.c,
+ main/event.c, pbx/pbx_dundi.c, main/asterisk.c: Change g_eid to
+ ast_eid_default.
+
+2009-03-27 13:57 +0000 [r184566-184628] Joshua Colp <jcolp@digium.com>
+
+ * bridges/bridge_softmix.c: Fix a potential race condition when
+ creating a software based mixing bridge. It was possible for no
+ timer to become available between creating the bridge and
+ starting it. We now open a timer when creating it and keep it
+ open until the bridge is destroyed.
+
+ * /, channels/chan_sip.c: Merged revisions 184565 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9
+ lines Fix an issue where nat=yes would not always take effect for
+ the RTP session on outgoing calls. If calls were placed using an
+ IP address or hostname the global nat setting was copied over but
+ was not set on the RTP session itself. This caused the RTP stack
+ to not perform symmetric RTP actions. (closes issue #14546)
+ Reported by: acunningham ........
+
+2009-03-27 02:20 +0000 [r184512-184531] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/lock.h: Fix some issues with rwlock corruption
+ that caused deadlock like symptoms. When dvossel and I were doing
+ some load testing last week, we noticed that we could make
+ Asterisk trunk lock up instantly when we started generating a
+ bunch of calls. The backtraces of locked threads were bizarre,
+ and many were stuck on an _unlock_ of an rwlock. The changes are:
+ 1) Fix a number of places where a backtrace would be loaded into
+ an invalid index of the backtrace array. It's an off by one
+ error, which ends up writing over the rwlock itself. 2) Ensure
+ that in the array of held locks, we NULL out an index once it is
+ not being used so that it's not confusing when analyzing its
+ contents. 3) Remove a bunch of logging referring to an rwlock
+ operating being done with "deep reentrancy". It is normal for
+ _many_ threads to hold a read lock on an rwlock.
+
+ * main/file.c: Don't act surprised if we get a -1 indication.
+
+ * main/heap.c, include/asterisk/heap.h: Pass more useful
+ information through to lock tracking when DEBUG_THREADS is on.
+
+2009-03-26 22:18 +0000 [r184448] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, sounds/Makefile: Merged revisions 184447 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184447 | kpfleming | 2009-03-26 17:17:32 -0500 (Thu, 26 Mar
+ 2009) | 3 lines use new, improved 8kHz prompts ........
+
+2009-03-26 21:09 +0000 [r184389] David Vossel <dvossel@digium.com>
+
+ * /, apps/app_test.c: Merged revisions 184388 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184388 | dvossel | 2009-03-26 16:07:32 -0500 (Thu, 26 Mar 2009)
+ | 8 lines pri loop TestClient/TestServer fails: server SEND DTMF
+ 8 app_test was failing when sending the last DTMF digit, 8,
+ because of the 100ms pause issued after DTMF is sent. During this
+ pause the other side would hang up causing the test to look like
+ it failed. Now the other side waits a second before hanging up.
+ (closes issue #12442) Reported by: tzafrir ........
+
+2009-03-25 22:11 +0000 [r184339-184344] Russell Bryant <russell@digium.com>
+
+ * main/event.c: Remove unneeded AST_LIST_ENTRY() and comment on the
+ purpose of ast_event_ref.
+
+ * channels/chan_unistim.c, channels/chan_dahdi.c,
+ include/asterisk/devicestate.h, include/asterisk/event.h,
+ channels/chan_sip.c, apps/app_minivm.c, res/ais/evt.c,
+ main/devicestate.c, main/event.c, include/asterisk/_private.h,
+ include/asterisk/strings.h, channels/chan_iax2.c,
+ main/asterisk.c, channels/chan_mgcp.c, apps/app_voicemail.c:
+ Improve performance of the ast_event cache functionality. This
+ code comes from svn/asterisk/team/russell/event_performance/.
+ Here is a summary of the changes that have been made, in order of
+ both invasiveness and performance impact, from smallest to
+ largest. 1) Asterisk 1.6.1 introduces some additional logic to be
+ able to handle distributed device state. This functionality comes
+ at a cost. One relatively minor change in this patch is that the
+ extra processing required for distributed device state is now
+ completely bypassed if it's not needed. 2) One of the things that
+ I noticed when profiling this code was that a _lot_ of time was
+ spent doing string comparisons. I changed the way strings are
+ represented in an event to include a hash value at the front. So,
+ before doing a string comparison, we do an integer comparison on
+ the hash. 3) Finally, the code that handles the event cache has
+ been re-written. I tried to do this in a such a way that it had
+ minimal impact on the API. I did have to change one API call,
+ though - ast_event_queue_and_cache(). However, the way it works
+ now is nicer, IMO. Each type of event that can be cached (MWI,
+ device state) has its own hash table and rules for hashing and
+ comparing objects. This by far made the biggest impact on
+ performance. For additional details regarding this code and how
+ it was tested, please see the review request. (closes issue
+ #14738) Reported by: russell Review:
+ http://reviewboard.digium.com/r/205/
+
+2009-03-25 19:22 +0000 [r184280] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix issue with a T38 reinvite being sent
+ even if not configured to do so. If we receive a T38 request
+ negotiate control frame we should only attempt to do so if the
+ option is enabled on the dialog.
+
+2009-03-25 14:38 +0000 [r184220] Eliel C. Sardanons <eliels@gmail.com>
+
+ * /, main/asterisk.c: Merged revisions 184188 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) |
+ 13 lines Avoid destroying the CLI line when moving the cursor
+ backward and trying to autocomplete. When moving the cursor
+ backward and pressing TAB to autocomplete, a NULL is put in the
+ line and we are loosing what we have already wrote after the
+ actual cursor position. (closes issue #14373) Reported by: eliel
+ Patches: asterisk.c.patch uploaded by eliel (license 64) Tested
+ by: lmadsen ........
+
+2009-03-25 14:33 +0000 [r184147-184219] Russell Bryant <russell@digium.com>
+
+ * main/timing.c: Include poll-compat.h
+
+ * main/timing.c: Change poll() to ast_poll().
+
+ * utils/Makefile, include/asterisk/compat.h: Fix build issues on
+ Mac OSX. (closes issue #14714) Reported by: ygor
+
+2009-03-24 22:40 +0000 [r184079] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_senddtmf.c: Merged revisions 184078 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar
+ 2009) | 9 lines Change NULL pointer check to be ast_strlen_zero.
+ The 'digit' variable is guaranteed to be non-NULL, so the if
+ statement could never evaluate true. Changing to ast_strlen_zero
+ makes the logic correct. This was found while reviewing
+ ast_channel_ao2 code review. ........
+
+2009-03-24 22:00 +0000 [r184037-184043] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Put siren7 and siren14 in ast_best_codec() just
+ so they're in there somewhere.
+
+ * channels/chan_iax2.c: Exclude slin16, siren7, and siren14 from
+ bandwidth=low and =medium The default codec configuration for
+ chan_iax2 is bandwidth=low. I noticed slin16 being negotiated as
+ the codec in some test calls, but that no longer happens after
+ this change.
+
+2009-03-24 20:01 +0000 [r183995] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: SIP
+ preferred codec only feature Added an option to respond to a SIP
+ invite with only the single most preferred joint codec. This
+ limits the options of what codecs the other side can use. (closes
+ issue #12485) Reported by: bamby Review:
+ http://reviewboard.digium.com/r/206/
+
+2009-03-24 15:26 +0000 [r183865-183914] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configs/voicemail.conf.sample: Merged revisions 183913 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009)
+ | 3 lines Additionally note that the operator option needs an 'o'
+ extension. (Related to issue #14731) ........
+
+ * main/http.c: Allow browsers to cache images and other static
+ content.
+
+2009-03-23 22:35 +0000 [r183831] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, channels/misdn/Makefile,
+ channels/misdn/chan_misdn_config.h, channels/misdn/ie.c,
+ channels/misdn/isdn_msg_parser.c, channels/misdn/portinfo.c,
+ channels/misdn/isdn_lib.c, channels/misdn_config.c: Removed
+ trailing whitespace in chan_misdn files.
+
+2009-03-23 18:58 +0000 [r183766] Mark Michelson <mmichelson@digium.com>
+
+ * /, res/res_monitor.c: Merged revisions 183700 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar
+ 2009) | 7 lines Fix a memory leak in res_monitor.c The only way
+ that this leak would occur is if Monitor were started using the
+ Manager interface and no File: header were given. Discovered
+ while reviewing the ast_channel_ao2 review request. ........
+
+2009-03-23 18:06 +0000 [r183701] Leif Madsen <lmadsen@digium.com>
+
+ * channels/chan_dahdi.c: Fixes a documentation error introduced
+ during the CLI cleanup at AstriDevCon 2008. (closes issue #14655)
+ Reported by: ulogic Patches: chan_dahdi.patch uploaded by ulogic
+ (license 728) Tested by: lmadsen
+
+2009-03-22 21:00 +0000 [r183652] Joshua Colp <jcolp@digium.com>
+
+ * main/bridging.c: Fix a minor logic flaw with the bridge generic
+ thread. We only want to move the channel pointers that are
+ actually present.
+
+2009-03-20 17:00 +0000 [r183560] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 183559 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20
+ Mar 2009) | 2 lines Fix a crash in IAX2 registration handling
+ found during load testing with dvossel. ........
+
+2009-03-20 16:25 +0000 [r183553-183555] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix chan_sip so it builds.
+
+ * include/asterisk/rtp.h, main/rtp.c, main/asterisk.exports: Remove
+ symbols I just added to main/asterisk.exports and instead rename
+ the functions.
+
+ * main/asterisk.exports: Add some missing symbols to
+ main/asterisk.exports Hey! chan_sip.so loads now!
+
+2009-03-20 12:12 +0000 [r183511] Eliel C. Sardanons <eliels@gmail.com>
+
+ * channels/chan_dahdi.c: Remove duplicate <description> inside the
+ xml documentation.
+
+2009-03-19 20:30 +0000 [r183436] David Vossel <dvossel@digium.com>
+
+ * apps/app_dial.c, /, main/features.c, include/asterisk/features.h:
+ Merged revisions 183386 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009)
+ | 6 lines Cleaning up a few things in detect disconnect patch
+ Initialized ast_call_feature in detect_disconnect to avoid
+ accessing uninitialized memory. Cleaned up /param tags in
+ features.h. No longer send dynamic features in
+ ast_feature_detect. issue #11583 ........
+
+2009-03-19 19:22 +0000 [r183321-183345] Tilghman Lesher <tlesher@digium.com>
+
+ * /: Recorded merge of revisions 183342 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183342 | tilghman | 2009-03-19 14:21:30 -0500 (Thu, 19 Mar 2009)
+ | 2 lines Reordering, to change prior to unlocking ........
+
+ * channels/chan_dahdi.c, /: Merged revisions 183319 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r183319 | tilghman | 2009-03-19 14:15:33 -0500 (Thu, 19
+ Mar 2009) | 8 lines Delay signalling progress until a PRI channel
+ really signals progress. (closes issue #13034) Reported by:
+ klaus3000 Patches: 20090316__bug13034.diff.txt uploaded by
+ tilghman (license 14) patch_trunk_183progress_klaus3000.txt
+ uploaded by klaus3000 (license 65) Tested by: klaus3000 ........
+
+2009-03-19 18:34 +0000 [r183312] Jason Parker <jparker@digium.com>
+
+ * /, main/asterisk.exports: Merged revisions 183291 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r183291 | qwell | 2009-03-19 13:28:16 -0500 (Thu, 19 Mar
+ 2009) | 1 line Export some more required symbols. ........
+
+2009-03-19 18:10 +0000 [r183244] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix a memory leak associated with queues. For
+ every attempt that app_queue made to place an outbound call to a
+ queue member, we would allocate a queue_end_bridge structure.
+ When the bridge for the call had completed, we would free the
+ structure. Unfortunately not all call attempts actually end up
+ bridged to a member, so we need to be more selective of when to
+ allocate the structure. With this change, the allocation occurs
+ in an area where we can guarantee that the call will be bridged.
+ (closes issue #14680) Reported by: caspy Patches: 14680.patch
+ uploaded by mmichelson (license 60) Tested by: caspy
+
+2009-03-19 18:00 +0000 [r183239-183242] Russell Bryant <russell@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ main/loader.c: Merged revisions 183241 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009)
+ | 2 lines Remove the use of RTLD_NOLOAD, as it is not behaving
+ like expected. ........
+
+ * /, main/asterisk.exports: Merged revisions 183238 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r183238 | russell | 2009-03-19 12:41:39 -0500 (Thu, 19
+ Mar 2009) | 1 line Allow the AES API to work. ........
+
+2009-03-19 17:00 +0000 [r183196] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_odbc.exports: 2 symbols defined when DEBUG_THREADS
+
+2009-03-19 16:28 +0000 [r183172] David Vossel <dvossel@digium.com>
+
+ * apps/app_dial.c, /, main/features.c, include/asterisk/features.h:
+ Merged revisions 183126 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009)
+ | 17 lines Allow disconnect feature before a call is bridged
+ feature.conf has a disconnect option. By default this option is
+ set to '*', but it could be anything. If a user wishes to
+ disconnect a call before the other side answers, only '*' will
+ work, regardless if the disconnect option is set to something
+ else. This is because features are unavailable until bridging
+ takes place. The default disconnect option, '*', was hardcoded in
+ app_dial, which doesn't make any sense from a user perspective
+ since they may expect it to be something different. This patch
+ allows features to be detected from outside of the bridge, but
+ not operated on. In this case, the disconnect feature can be
+ detected before briding and handled outside of features.c.
+ (closes issue #11583) Reported by: sobomax Patches:
+ patch-apps__app_dial.c uploaded by sobomax (license 359)
+ 11583.latest-patch uploaded by murf (license 17)
+ detect_disconnect.diff uploaded by dvossel (license 671) Tested
+ by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/
+ ........
+
+2009-03-19 16:22 +0000 [r183124-183148] Russell Bryant <russell@digium.com>
+
+ * /, main/asterisk.exports: Merged revisions 183145 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r183145 | russell | 2009-03-19 11:21:56 -0500 (Thu, 19
+ Mar 2009) | 1 line Add missing semicolon in exports script.
+ ........
+
+ * /, main/asterisk.exports: Merged revisions 183123 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r183123 | russell | 2009-03-19 11:13:18 -0500 (Thu, 19
+ Mar 2009) | 2 lines Allow the CallerID API to work again.
+ ........
+
+2009-03-19 16:07 +0000 [r183117] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 183115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar
+ 2009) | 14 lines Fix an issue where cancelled outgoing SIP calls
+ would erroneously report the device as "in use." A user was
+ having an issue where if an outgoing SIP call was canceled, the
+ SIP device would remain in use if we had not received any
+ response to the initial INVITE we sent out. The SIP device would
+ remain in use until the autocongestion timer was exhausted. I
+ tracked down the cause of this to be the section of code I am
+ removing here. I asked several people what the purpose of this
+ code was meant to be, but no one could give me any sort of answer
+ as to why this was here. The person who was having this issue has
+ been using this patch for several months and it has stopped the
+ problems they have had. AST-196 ........
+
+2009-03-19 15:37 +0000 [r183057-183108] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Improve our triggering of a T38 switchover
+ internally when triggered by a received reinvite. Previously we
+ reached across the channel bridge to get the other party's SIP
+ dialog structure in order to trigger an outgoing reinvite. This
+ is extremely dangerous to do and only works if bridged to another
+ SIP channel. This patch changes this to use the T38 control frame
+ method of requesting a switchover. This change also causes the
+ SIP channel driver to propogate back whether the switchover
+ worked or not instead of blindly accepting the incoming T38
+ reinvite. Review: http://reviewboard.digium.com/r/200/
+
+ * main/channel.c: Fix an issue where a T38 control frame would get
+ dropped. If two channels were bridged together using a generic
+ bridge the T38 control frame would get passed up instead of being
+ indicated on the other channel.
+
+2009-03-18 21:28 +0000 [r183032] Kevin P. Fleming <kpfleming@digium.com>
+
+ * res/res_ael_share.exports (added): allow this module to export
+ everything for now
+
+2009-03-18 21:18 +0000 [r183028] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/h323/ast_h323.cxx: Add some code removed by mistake from
+ commit 182722 that works around a file descriptor leak in
+ versions of PWLib prior to 1.12.0.
+
+2009-03-18 19:41 +0000 [r182960] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.exports: Fixing a lost symbol in manager.c
+
+2009-03-18 11:40 +0000 [r182848-182883] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/callerid.h, channels/chan_dahdi.c, /,
+ main/callerid.c: Merged revisions 182882 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r182882 | kpfleming | 2009-03-18 06:31:41 -0500 (Wed, 18 Mar
+ 2009) | 3 lines fix another symbol namespace issue (reported by
+ Andrew on asterisk-dev) ........
+
+ * res/res_phoneprov.c, res/res_config_ldap.c, res/res_curl.c,
+ res/res_config_sqlite.c, res/res_jabber.exports, res/res_odbc.c,
+ res/res_odbc.exports: a few more namespace updates...
+ res_ael_share still needs some work before this can be merged to
+ other release branches
+
+2009-03-18 02:28 +0000 [r182847] Russell Bryant <russell@digium.com>
+
+ * apps/app_nbscat.c, /, main/Makefile,
+ include/asterisk/autoconfig.h.in, configure.ac, main/utils.c,
+ include/asterisk/io.h, include/asterisk/channel.h, main/poll.c,
+ main/io.c, main/channel.c, channels/chan_skinny.c, configure,
+ apps/app_mp3.c, res/res_agi.c, channels/chan_alsa.c,
+ include/asterisk/poll-compat.h, main/asterisk.c: Merged revisions
+ 182810 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009)
+ | 44 lines Fix cases where the internal poll() was not being used
+ when it needed to be. We have seen a number of problems caused by
+ poll() not working properly on Mac OSX. If you search around,
+ you'll find a number of references to using select() instead of
+ poll() to work around these issues. In Asterisk, we've had poll.c
+ which implements poll() using select() internally. However, we
+ were still getting reports of problems. vadim investigated a bit
+ and realized that at least on his system, even though we were
+ compiling in poll.o, the system poll() was still being used. So,
+ the primary purpose of this patch is to ensure that we're using
+ the internal poll() when we want it to be used. The changes are:
+ 1) Remove logic for when internal poll should be used from the
+ Makefile. Instead, put it in the configure script. The logic in
+ the configure script is the same as it was in the Makefile.
+ Ideally, we would have a functionality test for the problem, but
+ that's not actually possible, since we would have to be able to
+ run an application on the _target_ system to test poll()
+ behavior. 2) Always include poll.o in the build, but it will be
+ empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll()
+ throughout the source tree to ast_poll(). I feel that it is good
+ practice to give the API call a new name when we are changing its
+ behavior and not using the system version directly in all cases.
+ So, normally, ast_poll() is just redefined to poll(). On systems
+ where AST_POLL_COMPAT is defined, ast_poll() is redefined to
+ ast_internal_poll(). 4) Change poll() in main/poll.c to be
+ ast_internal_poll(). It's worth noting that any code that still
+ uses poll() directly will work fine (if they worked fine before).
+ So, for example, out of tree modules that are using poll() will
+ not stop working or anything. However, for modules to work
+ properly on Mac OSX, ast_poll() needs to be used. (closes issue
+ #13404) Reported by: agalbraith Tested by: russell, vadim
+ http://reviewboard.digium.com/r/198/ ........
+
+2009-03-18 02:21 +0000 [r182826] Kevin P. Fleming <kpfleming@digium.com>
+
+ * res/res_config_pgsql.c, /, res/res_snmp.c, res/res_smdi.exports
+ (added), main/Makefile, include/asterisk/astobj2.h,
+ res/res_agi.exports (added), Makefile.rules, main/astobj2.c,
+ main/asterisk.exports (added), res/res_odbc.exports (added),
+ res/res_speech.exports (added), res/res_config_odbc.c,
+ res/res_features.exports (added), build_tools/strip_nonapi
+ (removed), res/res_adsi.exports (added), default.exports (added),
+ makeopts.in, res/res_jabber.exports (added),
+ res/res_monitor.exports (added): Merged revisions 182808 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r182808 | kpfleming | 2009-03-17 20:55:22 -0500 (Tue, 17 Mar
+ 2009) | 5 lines Improve the build system to *properly* remove
+ unnecessary symbols from the runtime global namespace. Along the
+ way, change the prefixes on some internal-only API calls to use a
+ common prefix. With these changes, for a module to export symbols
+ into the global namespace, it must have *both* the
+ AST_MODFLAG_GLOBAL_SYMBOLS flag and a linker script that allows
+ the linker to leave the symbols exposed in the module's .so file
+ (see res_odbc.exports for an example). ........
+
+2009-03-17 21:28 +0000 [r182762] Russell Bryant <russell@digium.com>
+
+ * funcs/func_channel.c, CHANGES: Add support for the "name" option
+ in the CHANNEL() function. Review:
+ http://reviewboard.digium.com/r/199/
+
+2009-03-17 20:47 +0000 [r182722] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/h323/compat_h323.cxx, channels/h323/ast_h323.cxx,
+ configure, autoconf/ast_check_openh323.m4,
+ channels/h323/compat_h323.h, channels/chan_h323.c,
+ channels/h323/ast_h323.h, channels/h323/chan_h323.h: Allow H.323
+ Plus library to be used in addition to the OpenH323 library
+ Chan_h323 can now be compiled against both the previously
+ supported versions of OpenH323 as well as the current H.323 Plus
+ (version 1.20.2). The configure script has been modified to look
+ in the default install location of h323 to hopefully help avoid
+ using the environment variables OPENH323DIR and PWLIBDIR. Also,
+ the CLI command "h323 show version" has been added which
+ indicates which version of h323 is in use. (closes issue #11261)
+ Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch
+ uploaded by jthurman (license 614)
+
+2009-03-17 18:06 +0000 [r182596-182607] David Vossel <dvossel@digium.com>
+
+ * CHANGES: Fixing CHANGES in rev 182596. Progress DTMF was added
+ into app_dial's D() option. In CHANGES it should have been
+ updated under 1.6.3 rather than 1.6.2.
+
+ * apps/app_dial.c, CHANGES: Option to send DTMF when receiving
+ PROGRESS status The D() option in app_dial is only able to send
+ DTMF after the call has been answered. A progress option has been
+ added to D() to allow DTMF to be sent upon receiving PROGRESS.
+ This allows DTMF to be sent before the call is answered. (closes
+ issue #12123) Reported by: VoipForces Patches:
+ app_dial.c_patch_trunk_valid uploaded by VoipForces (license 419)
+ dtmf_progress.patch uploaded by dvossel (license 671) Tested by:
+ VoipForces, dvossel
+
+2009-03-17 15:22 +0000 [r182553] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Tweak the handling of the frame list inside of
+ ast_answer(). This does not change any behavior, but moves the
+ frames from the local frame list back to the channel read queue
+ using an O(n) algorithm instead of O(n^2).
+
+2009-03-17 14:59 +0000 [r182525-182530] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/channel.c: correct logic flaw in ast_answer() changes in
+ r182525
+
+ * main/channel.c, main/features.c, include/asterisk/channel.h:
+ Improve behavior of ast_answer() to not lose incoming frames
+ ast_answer(), when supplied a delay before returning to the
+ caller, use ast_safe_sleep() to implement the delay.
+ Unfortunately during this time any incoming frames are discarded,
+ which is problematic for T.38 re-INVITES and other sorts of
+ channel operations. When a delay is not passed to ast_answer(),
+ it still delays for up to 500 milliseconds, waiting for media to
+ arrive. Again, though, it discards any control frames, or
+ non-voice media frames. This patch rectifies this situation, by
+ storing all incoming frames during the delay period on a list,
+ and then requeuing them onto the channel before returning to the
+ caller. http://reviewboard.digium.com/r/196/
+
+2009-03-17 14:24 +0000 [r182521] Sean Bright <sean@malleable.com>
+
+ * autoconf/ast_ext_lib.m4: Don't include a space before the
+ optional extra text that may follow a help string.
+
+2009-03-17 05:51 +0000 [r182450] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/db.c: Merged revisions 182449 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009)
+ | 7 lines Fix race in astdb The underlying db1 implementation
+ does not fully isolate the pages retrieved from astdb, so the
+ lock protecting accesses needs to be extended until the copy from
+ the shared memory structure is done. (closes issue #14682)
+ Reported by: makoto ........
+
+2009-03-17 01:54 +0000 [r182408] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: OPENR2 uses an incorrect string value if
+ the extension delimiter is not present. * Fixed OPENR2 using an
+ incorrect string value if the extension delimiter is not present
+ in the Dial() function. This was fixed for SS7 and PRI in trunk
+ -r172400. * Made OPENR2 stripmsd behavior the same as the SS7,
+ PRI, and others. * Removed trailing whitespace that appeared with
+ OPENR2.
+
+2009-03-16 20:53 +0000 [r182362] Russell Bryant <russell@digium.com>
+
+ * UPGRADE.txt, CHANGES: Update UPGRADE.txt and CHANGES for 1.6.3
+
projects / thirdparty / asterisk.git / commitdiff
