From: Asterisk Autobuilder Date: Mon, 8 Oct 2012 20:41:50 +0000 (+0000) Subject: Importing files for 1.8.18.0-rc1 release. X-Git-Tag: 1.8.18.0-rc1~2 X-Git-Url: http://git.ipfire.org/?a=commitdiff_plain;h=d69b1c4861e74351b76e49a86c0f73356b70bc30;p=thirdparty%2Fasterisk.git Importing files for 1.8.18.0-rc1 release. git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.8.18.0-rc1@374679 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- diff --git a/.lastclean b/.lastclean new file mode 100644 index 0000000000..c7f5afc52a --- /dev/null +++ b/.lastclean @@ -0,0 +1,3 @@ +39 + + diff --git a/.version b/.version new file mode 100644 index 0000000000..209b684e35 --- /dev/null +++ b/.version @@ -0,0 +1 @@ +1.8.18.0-rc1 diff --git a/ChangeLog b/ChangeLog new file mode 100644 index 0000000000..89997df1cb --- /dev/null +++ b/ChangeLog @@ -0,0 +1,42458 @@ +2012-10-08 Asterisk Development Team + + * Asterisk 1.8.18.0-rc1 Released. + +2012-10-05 20:20 +0000 [r374570-374581] dlee : + + * 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 + + * 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 + + * 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 + + * 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 : + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * channels/chan_sip.c: Be consistent, send From: "Anonymous" + When setting + CALLERID(pres)=unavailable in the dialplan, the From header in + the SIP message contains "Anonymous" + . 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 : + + * 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 + + * Asterisk 1.8.17.0-rc1 Released. + +2012-09-12 15:42 +0000 [r372959] Matthew Jordan + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 : + + * 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 + + * 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 + + * 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 + + * 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 : + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * apps/app_meetme.c: app_meetme: Adding test events for following + activity in MeetMe. + +2012-08-29 19:38 +0000 [r371860-371888] Richard Mudgett + + * 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 + + * 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 + + * 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 : + + * 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 + + * 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 + + * main/dsp.c: mf_detect: incorrectly used DTMF_GSIZE instead of + MF_GSIZE + +2012-08-21 20:35 +0000 [r371590] Mark Michelson + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * channels/chan_sip.c: Fix problem where incorrect pointer was + checked for nullity. + +2012-08-10 21:21 +0000 [r371141] Mark Michelson + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * main/channel.c: Add missing AST_CAUSE_* -> text translations + +2012-08-06 15:00 +0000 [r370797] Mark Michelson + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * Asterisk 1.8.16.0 Released. + +2012-09-11 Asterisk Development Team + + * 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 + + * Asterisk 1.8.16.0-rc1 Released. + +2012-07-30 16:47 +0000 [r370563] Richard Mudgett + + * channels/chan_misdn.c: Release B channel allocation on error path + in chan_misdn. + +2012-07-25 21:00 +0000 [r370494] Jonathan Rose + + * 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 + + * 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 + + * channels/chan_oss.c: chan_oss: fix "sample rate" error message + +2012-07-23 21:09 +0000 [r370360-370383] Kevin P. Fleming + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * Asterisk 1.8.15.0 Released. + +2012-07-11 Asterisk Development Team + + * Asterisk 1.8.15.0-rc1 Released. + +2012-07-10 13:33 +0000 [r369869] Kinsey Moore + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * Asterisk 1.8.14.0 Released. + +2012-07-06 Asterisk Development Team + + * 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 + + * Asterisk 1.8.14.0-rc1 Released. + +2012-06-06 21:27 +0000 [r368644] Richard Mudgett + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * tests/test_config.c: Fix type punned compiler warning in + test_config.c + +2012-05-07 18:36 +0000 [r365474] Matthew Jordan + + * 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 , 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 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * Asterisk 1.8.13.0 Released. + +2012-05-30 Asterisk Development Team + + * 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 + + * Asterisk 1.8.13.0-rc1 Released. + +2012-05-02 17:02 +0000 [r365006-365068] Terry Wilson + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * funcs/func_version.c: Fix documentation for + ${VERSION(ASTERISK_VERSION_NUM)}. + +2012-04-19 21:05 +0000 [r362680] Michael L. Young + + * 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 + + * configs/queues.conf.sample: Update membermacro and membergosub + documentation in queues.conf.sample. + +2012-04-19 15:53 +0000 [r362586] Sean Bright + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * Asterisk 1.8.12.0 Released. + +2012-05-01 Asterisk Development Team + + * 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 + + * Asterisk 1.8.12.0-rc2 Released. + + * AST-2012-004 + + * AST-2012-005 + + * AST-2012-006 + +2012-04-04 Asterisk Development Team + + * Asterisk 1.8.12.0-rc1 Released. + +2012-04-04 16:29 +0000 [r361090-361142] Jonathan Rose + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * configure, configure.ac: Also detect gmime 2.6 Also detect gmime + version 2.6 (Michael Biebl) Signed-off-by: Tzafrir Cohen (License + #5035) + +2012-03-21 13:19 +0000 [r360086] Matthew Jordan + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + +2012-03-07 15:16 +0000 [r358435-358438] Terry Wilson + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * Asterisk 1.8.11.0 Released. + +2012-03-26 Asterisk Development Team + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + ,,)> 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * Asterisk 1.8.11.0-rc2 Released. + +2012-03-05 Asterisk Development Team + + * Asterisk 1.8.10.0 Released. + +2012-03-01 Asterisk Development Team + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * channels/chan_sip.c: Cleanup dialog-info+xml Notify dialog Make + similar to other Notify messages. sample output: + terminated 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * res/res_rtp_asterisk.c: Eliminate odd initialization of probation + variable. + +2012-01-17 16:55 +0000 [r351287] Jonathan Rose + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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" . 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * Asterisk 1.8.9.0 Released. + + * Test results: + http://bamboo.asterisk.org/browse/TESTING-ASTERISK1890RCS-6 + +2012-01-24 Asterisk Development Team + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * apps/app_dial.c: Fix missing doc tags found while fixing + ASTERISK-18689 Add missing tags in app_dial + documentation. + +2011-12-23 02:09 +0000 [r348940] Richard Mudgett + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * channels/chan_sip.c: respect case changes in peer names on sip + reload ASTERISK-18669 + +2011-11-07 21:13 +0000 [r343637] Richard Mudgett + + * 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 + + * 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 + + * 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 + + * 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 + + * funcs/func_dialgroup.c: Remove invalid flag given to iterator in + func_dialgroup.c + +2011-11-03 16:15 +0000 [r343281] Alexandr Anikin + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * Asterisk 1.8.8.0 Released. + +2011-12-09 Asterisk Development Team + + * 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 + + * 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 + + * 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 + + * 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 + + * Asterisk 1.8.8.0-rc1 Released. + +2011-10-05 21:30 +0000 [r339566] Leif Madsen + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * addons/ooh323c/src/memheap.c: fix forget declaration in previous + change + +2011-10-03 20:12 +0000 [r339144-339147] Leif Madsen + + * 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 + + * 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 + + * 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 + + * 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 + + * configs/queues.conf.sample: Adds documentation for + QueueMemberStatus event generation + +2011-09-30 16:27 +0000 [r338663] Richard Mudgett + + * 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 + + * apps/app_osplookup.c, configure.ac: Remove r338137 and r338138. + +2011-09-29 21:12 +0000 [r338555] Paul Belanger + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * apps/app_macro.c: Upgrade app_macro to core + +2011-09-26 19:30 +0000 [r337973] Richard Mudgett + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * apps/app_directed_pickup.c: Remove obsolete todo comment about + PICKUPRESULT. + +2011-09-13 21:33 +0000 [r335716] Tzafrir Cohen + + * 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) + + +2011-09-13 21:30 +0000 [r335714] Paul Belanger + + * apps/app_meetme.c: Meetme should have 'core' support level + (closes issue ASTERISK-18542) + +2011-09-13 18:52 +0000 [r335655] Tilghman Lesher + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * Asterisk 1.8.7.0 Released. + +2011-09-19 Asterisk Development Team + + * 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 + + * Asterisk 1.8.7.0-rc1 Released. + +2011-09-06 13:48 +0000 [r334453] Gregory Nietsky + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * apps/app_voicemail.c: Fixed improperly formatted TestEvent AMI + message in app_voicemail + +2011-08-29 15:55 +0000 [r333569] Jonathan Rose + + * 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 + + * 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) , + +2011-08-26 16:19 +0000 [r333378] Jonathan Rose + + * 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 + + * 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 + + * Makefile: Fix for DESTDIR spaces patch. + +2011-08-25 18:47 +0000 [r333265] Jonathan Rose + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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,) on the channel. Having + chan_sip set HASH(SIP_CAUSE,) 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * tests/test_netsock2.c: Fix netsock2 multiple zero-expansion test + Remove erroneous single bracket. + +2011-08-12 16:20 +0000 [r331649] Kinsey Moore + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * Asterisk 1.8.6.0 Released. + +2011-08-25 Asterisk Development Team + + * 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 + + * 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 + + * Asterisk 1.8.6.0-rc1 Released. + +2011-08-10 13:47 +0000 [r331315] Kinsey Moore + + * 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 + + * 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 + + * 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 + + * doc/asterisk.8: Regenerate asterisk man page from sgml. + +2011-08-08 20:52 +0000 [r331038] Kinsey Moore + + * 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 + +2011-08-04 20:29 +0000 [r330843] Terry Wilson + + * configure, configure.ac: Make libsrtp instructions more explicit + when linking fails (closes issue ASTERISK-18139) + +2011-08-04 19:37 +0000 [r330827] Alexandr Anikin + + * 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 + + * 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 + + * res/res_jabber.c: Convert an error message to actually be + helpful. + +2011-08-02 16:15 +0000 [r330575-330581] David Vossel + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * main/term.c: Make console colors work for TERM=xterm-256color + +2011-07-28 17:04 +0000 [r330050] Richard Mudgett + + * 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 + + * 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 + + * 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 + + * channels/chan_sip.c: Make the output of Externhost in 'sip show + settings' more consistent. + +2011-07-27 19:27 +0000 [r329782] Leif Madsen + + * apps/app_confbridge.c: Change support for ConfBridge() in 1.8 to + Extended. + +2011-07-27 19:17 +0000 [r329767] Sean Bright + + * 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 + + * 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 + + * 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 + + * 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 + + * main/enum.c: Decrease verbose messages to debug, to help clean up + CLI. + +2011-07-22 21:10 +0000 [r329144-329333] Richard Mudgett + + * 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 + + * UPGRADE.txt: Asterisk now requires libpri 1.4.11+ for PRI + support. + +2011-07-20 20:52 +0000 [r329012] Richard Mudgett + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * funcs/func_odbc.c: Typo + +2011-07-15 20:41 +0000 [r328446] Leif Madsen + + * 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 + + * 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 + + * 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 + + * 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 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 + + * 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 + + * configs/features.conf.sample: Add ATXFER_NULL_TECH note in + features.conf.sample. + +2011-07-12 22:53 +0000 [r327950] Kevin P. Fleming + + * 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 + + * 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 + + * Makefile: Fix uninstall target, so that modules dir gets cleared + again. + +2011-07-12 19:10 +0000 [r327852] Matthew Jordan + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * res/res_http_post.c: libgen.h is also needed on Darwin for + basename(3) + +2011-07-07 16:04 +0000 [r326689] Jonathan Rose + + * 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 + + * 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 + + * 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 + + * 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 "" 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * Asterisk 1.8.5-rc1 Released. + +2011-06-29 18:59 +0000 [r325673] David Vossel + + * 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 + + * 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 + + * 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 + + * channels/chan_sip.c: Fix random misspelling noticed on + asterisk-users. + +2011-06-28 20:31 +0000 [r325339] David Vossel + + * 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 + + * /, 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 + + * 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 + + * 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 + + * build_tools/prep_tarball: Remove line from prep_tarball that + kills mkrelease. + +2011-06-27 16:30 +0000 [r324955] Tilghman Lesher + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * channels/chan_sip.c: Use correct variable for text SRTP media. + +2011-06-22 18:52 +0000 [r324484] Terry Wilson + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * main/cdr.c: CDR comment tweaks. + +2011-06-01 20:10 +0000 [r321537] Brett Bryant + + * 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 + + * 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 + + * include/asterisk/dnsmgr.h, include/asterisk/acl.h: Update some + comments. + +2011-05-31 18:52 +0000 [r321515] David Vossel + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * include/asterisk/netsock2.h: Update ast_sockaddr comment with an + important note. + +2011-05-26 17:29 +0000 [r321042] Terry Wilson + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 "" output, that might be a bug as "" + 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * channels/chan_iax2.c: chan_iax2: change LOG_NOTICE to LOG_DEBUG + in iax2_read(). + +2011-05-09 23:15 +0000 [r318351] Richard Mudgett + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * apps/app_minivm.c: Resolve another warning. + +2011-05-03 21:37 +0000 [r316330] David Vossel + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * main/asterisk.c: Set the copyright year to 2011 in the startup + message. + +2011-04-27 12:36 +0000 [r315765] Leif Madsen + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * apps/app_dial.c: Add missing set of name valid flag when dialing. + +2011-04-26 17:40 +0000 [r315446] Russell Bryant + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * /: 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 + + * channels/chan_dahdi.c: Correct DAHDIShowChannels XML + documentation. + +2011-04-21 18:24 +0000 [r314628] Matthew Nicholson + + * 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 + + * /, 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 + + * include/asterisk/frame.h: AST_CONTROL_XXX comment changes. + +2011-04-20 05:25 +0000 [r314358] Terry Wilson + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * channels/chan_dahdi.c: Update some setup_dahdi_int() comments. + +2011-03-29 07:08 +0000 [r311799] Tilghman Lesher + + * 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 + + * apps/app_voicemail.c: Cross-reference VoiceMail() and + VoiceMailMain() in the xml docs. + +2011-03-27 21:47 +0000 [r311687] Alexandr Anikin + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * res/res_fax.c: Properly populate the LOCALSTATIONID channel + variable. + +2011-03-18 02:59 +0000 [r311295-311297] Richard Mudgett + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * addons/chan_ooh323.c: Fix branch compile. + +2011-03-15 01:00 +0000 [r310781] Alec L Davis + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * channels/sig_pri.c: Make pri parameter description consistent. + +2011-03-07 22:07 +0000 [r309858] Jonathan Rose + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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/[:]- 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * Asteris 1.8.4 Released. + +2011-04-25 Leif Madsen + + * 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 + + * 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 + + * Asterisk 1.8.4-rc1 Released. + +2011-02-23 23:38 +0000 [r308622] Richard Mudgett + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * utils: A few more (copies of) files to ignore in this directory. + +2011-02-18 00:07 +0000 [r308242] Alexandr Anikin + + * 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 + + * addons/ooh323c/src/ooSocket.c: Fix FreeBSD builds. + +2011-02-16 07:57 +0000 [r308098] Alexandr Anikin + + * addons/ooh323c/src/ooSocket.c: ifdef __linux__ keepalive + variables also + +2011-02-15 23:34 +0000 [r308010] Jason Parker + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 ", 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * main/manager.c: Only check container count if it exists. + +2011-01-19 23:49 +0000 [r302834] Sean Bright + + * 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 + + * contrib/scripts/install_prereq: Add binutils-dev for + BETTER_BACKTRACES + +2011-01-19 23:06 +0000 [r302785-302789] Russell Bryant + + * 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 + + * /, 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 + + * 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 + + * res/res_fax.c: Fix typo pointed out on asterisk-users list. + +2011-01-19 19:03 +0000 [r302505-302555] Sean Bright + + * 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 + + * /, 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 + + * 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 + + * main/features.c: Use the expanded format type instead of plain + int. + +2011-01-18 21:43 +0000 [r302314] Matthew Nicholson + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * Asterisk 1.8.3 Released. + + * Merged changes related to AST-2011-002 + +2011-02-16 Leif Madsen + + * 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 + + * 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 + + * Asterisk 1.8.3-rc1 Released. + +2011-01-18 18:11 +0000 [r302174] Richard Mudgett + + * /, 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 + + * configs/sip.conf.sample: Document "encryption" option in + sip.conf.sample + +2011-01-14 21:09 +0000 [r301946] Richard Mudgett + + * 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 + + * 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 + + * 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 + + * 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 + + * funcs/func_base64.c, /, funcs/func_aes.c: Add relationships to + function documentation. + +2011-01-14 17:32 +0000 [r301790] Jeff Peeler + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * main/file.c: Removal of unused variables so Asterisk will + compile. + +2011-01-12 15:57 +0000 [r301444] Stefan Schmidt + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 to conference /: + 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * main/aoc.c, channels/sig_pri.h, channels/chan_sip.c: Typos: + recieved => received + +2010-12-18 00:09 +0000 [r298818-298963] Tilghman Lesher + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * Asterisk 1.8.2-rc1 Released. + +2010-12-11 21:45 +0000 [r298099] Alexandr Anikin + + * 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 + + * res/res_fax.c: Prevent a memcpy overlap in + GENERIC_FAX_EXEC_SET_VARS + +2010-12-10 16:26 +0000 [r298051] Tilghman Lesher + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + s, these are for descriptions. ........ + +2010-12-03 15:21 +0000 [r297486-297495] Matthew Nicholson + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * res/res_fax.c: Changed some NOTICE and WARNING messages to DEBUG + messages. + +2010-12-01 17:53 +0000 [r297075] Jeff Peeler + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * Asterisk 1.8.1 Released. + +2010-11-16 Leif Madsen + + * Asterisk 1.8.1-rc1 Released. + +2010-11-15 18:30 +0000 [r294989-295078] Tilghman Lesher + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * main/channel.c: Allow ast_do_masquerade() failure to be reported + again. + +2010-11-09 20:33 +0000 [r294430] Tilghman Lesher + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * channels/chan_sip.c: Fixes ringback tone on sip semi-attended + transfer. ABE-2168 + +2010-11-04 13:27 +0000 [r293887] Paul Belanger + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 '' on channel + ((), line )". ........ ................ + +2010-11-02 20:45 +0000 [r293611] Paul Belanger + + * main/manager.c: If manager and tls are disabled, do not display + TCP/TLS Bindaddress. + +2010-11-01 17:29 +0000 [r293530] Richard Mudgett + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * contrib/scripts/install_prereq: Add resample and imap_tk + dependencies. + +2010-10-19 19:27 +0000 [r292309] Terry Wilson + + * 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 + + * Asterisk 1.8.0 Released. + +2010-10-18 Leif Madsen + + * Asterisk 1.8.0-rc5 Released. + +2010-10-18 22:02 +0000 [r292230] Leif Madsen + + * 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 + + * 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 + + * 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 + + * 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 + + * main/netsock2.c: Fixes build error for systems not supporting + IPV6_TCLASS. + +2010-10-18 17:15 +0000 [r292122] Matthew Nicholson + + * addons/chan_mobile.c: Fix the cmgr parser. (closes issue 0018152) + Reported by: menschentier + +2010-10-18 Leif Madsen + + * Asterisk 1.8.0-rc4 Released + +2010-10-18 16:02 +0000 [r292085] David Vossel + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * doc/tex/secure-calls.tex: Fix a typo - s/seucre/secure/ + +2010-10-13 23:45 +0000 [r291656] Richard Mudgett + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * Asterisk 1.8.0-rc3 Released. + +2010-10-07 11:00 +0000 [r290713] Russell Bryant + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * funcs/func_env.c: Solaris fixes. + +2010-09-30 19:53 +0000 [r289554] Matthew Nicholson + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * main/channel.c: Make development error message indicate which + channel. + +2010-09-29 15:04 +0000 [r289179] Matthew Nicholson + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * Asterisk 1.8.0-rc2 Released. + +2010-09-23 18:05 +0000 [r288507-288572] Terry Wilson + + * 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 + + * Asterisk 1.8.0-rc1 Released. + +2010-09-22 17:49 +0000 [r288345-288418] David Vossel + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * main/channel.c: Revert change in favor of a more targeted fix + +2010-09-21 18:32 +0000 [r287929] David Vossel + + * 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 + + * main/features.c: Cut-n-paste error in builtin_blindtransfer(). + +2010-09-21 15:43 +0000 [r287895] Russell Bryant + + * 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 + + * 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 + + * main/logger.c: Fix a regression in verbose logger processing. + +2010-09-21 04:37 +0000 [r287833] Terry Wilson + + * main/channel.c: Don't generate connected line buffer twice for + comparison + +2010-09-21 00:00 +0000 [r287760] Brett Bryant + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * main/features.c: fix uninintialized variable + +2010-09-15 20:53 +0000 [r287017] Richard Mudgett + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /: 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 + + * doc/externalivr.txt, CHANGES: Added missing documentation for + ExternalIVR feature added in January 2010 + +2010-09-10 05:32 +0000 [r285931-285962] Tilghman Lesher + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * funcs/func_channel.c: Documentation only + +2010-09-07 22:22 +0000 [r285455] Jason Parker + + * 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 + + * 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 + + * main/features.c: Fix cut-n-paste error. + +2010-09-07 20:58 +0000 [r285369] Jason Parker + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * include/asterisk/cli.h: Add a C++ compatible version of + AST_CLI_DEFINE(). + +2010-09-03 23:19 +0000 [r285017] Terry Wilson + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * Asterisk 1.8.0-beta5 released. + +2010-09-01 18:44 +0000 [r284477] Terry Wilson + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * pbx/pbx_realtime.c: Print exten@context:priority in verbose + messages from pbx_realtime. + +2010-08-27 20:31 +0000 [r283882] Jason Parker + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * channels/chan_sip.c: Convert ast_log(LOG_DEBUG, ...) to + ast_debug(...) + +2010-08-24 20:34 +0000 [r283493] David Vossel + + * 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 + + * 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 + + * /, 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 + + * funcs/func_odbc.c: Don't attempt to release a NULL ODBC handle. + +2010-08-23 21:33 +0000 [r283319] Tilghman Lesher + + * 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 + + * Asterisk 1.8.0-beta4 Released. + +2010-08-23 13:35 +0000 [r283177-283241] Russell Bryant + + * 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 + + * 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 + + * 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 + + * /: 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 + + * 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 + + * /, 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 + + * /, 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 + + * main/netsock2.c: Only output debugging if the debug level is on. + +2010-08-19 02:18 +0000 [r282740] Terry Wilson + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * channels/chan_sip.c: Avoid a deadlock in + add_header_max_forwards(). Related to r276951 + +2010-08-11 15:18 +0000 [r281723] Tilghman Lesher + + * /, 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] + + * 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 + + * 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 + + * Asterisk 1.8.0-beta3 Released. + +2010-08-10 17:48 +0000 [r281529-281568] Russell Bryant + + * 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 + + * apps/app_osplookup.c: Fixed the issue caused by EXTEN including + user parameters. + +2010-08-09 23:04 +0000 [r281466] Jeff Peeler + + * channels/chan_local.c: Add some more stuff to copy from 281429. + +2010-08-09 20:47 +0000 [r281432] David Vossel + + * /, 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 + + * 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 + + * res/res_fax.c: Validate minrate, maxrate, and modem settings + before attempting a fax session. FAX-224 + +2010-08-09 14:31 +0000 [r281356] + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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] + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * channels/chan_usbradio.c: Use PRId64 with format_t + +2010-07-28 20:49 +0000 [r280269] Jeff Peeler + + * channels/sip/reqresp_parser.c: Give test category missing leading + slash + +2010-07-28 20:12 +0000 [r280235] Richard Mudgett + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * res/res_crypto.c: s/init keys/keys init/ + +2010-07-28 01:37 +0000 [r280023] Paul Belanger + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * Asterisk 1.8.0-beta2 Released. + +2010-07-26 23:29 +0000 [r279689] Paul Belanger + + * UPGRADE.txt, CHANGES: Updated documentation for FAX logger level. + +2010-07-26 23:03 +0000 [r279658] Jason Parker + + * 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 + + * main/channel.c: Ignore a control subclass of -1 in + ast_waitfordigit_full(). + +2010-07-26 21:20 +0000 [r279599-279619] Tilghman Lesher + + * /, 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 + + * /: 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * tests/test_func_file.c: Add trailing backslash to silence warning + message. + +2010-07-25 18:21 +0000 [r279390-279410] Tilghman Lesher + + * 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 + + * doc/asterisk.8: Minor update to man page + +2010-07-24 20:47 +0000 [r279273-279314] Paul Belanger + + * 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 + + * 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 + + * Asterisk 1.8.0-beta1 Released. + +2010-07-23 18:56 +0000 [r279113] Tilghman Lesher + + * res/res_odbc.c: Silly 64-bit compilers (who uses 64-bit anyway?) + +2010-07-23 18:23 +0000 [r279056-279094] Russell Bryant + + * /: 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 + + * 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 + + * 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 + + * 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 + + * doc/asterisk.8: Some left-over hyphen-minus fixes in the man page + +2010-07-23 15:57 +0000 [r278944-278945] Russell Bryant + + * 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 + + * addons/chan_ooh323.c: Well, who knew chan_ooh323 used udptl? I + sure didn't! + +2010-07-23 15:41 +0000 [r278942] Richard Mudgett + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * include/asterisk/doxyref.h: Add the full current set of CDR + drivers + +2010-07-21 19:16 +0000 [r278539] David Vossel + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * channels/chan_dahdi.c: include stat.h for everybody, needed for + device2chan + +2010-07-20 23:23 +0000 [r278275-278307] Tilghman Lesher + + * 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 + + * channels/sig_pri.c: Reference correct struct member for unlikely + event PRI_EVENT_CONFIG_ERR. + +2010-07-20 22:26 +0000 [r278272] Tilghman Lesher + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * tests/test_devicestate.c: updated devicestate test for device + state changes + +2010-07-16 19:22 +0000 [r277366] Jeff Peeler + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * channels/sip/reqresp_parser.c: Fix up some weird indentation + problems in reqresp_parser.c + +2010-07-16 15:20 +0000 [r277143] Sean Bright + + * 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 + + * CREDITS: Add Despegar.com (my main sponsor) to the CREDITS file. + +2010-07-16 13:32 +0000 [r276950-277102] Olle Johansson + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * contrib/scripts/install_prereq: Add lua5.1 to the handy dandy + list of packages. + +2010-07-14 22:58 +0000 [r276571] Jeff Peeler + + * 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 + + * 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 + + * channels/chan_h323.c: Make compile again. + +2010-07-14 21:11 +0000 [r276490-276493] Tilghman Lesher + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * res/snmp/agent.c: Make compile again. + +2010-07-14 16:36 +0000 [r276389] Jeff Peeler + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /: 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * channels/chan_sip.c: Don't try to ref authpeer when it isn't set + +2010-07-12 17:54 +0000 [r275725] Richard Mudgett + + * main/channel.c: Add which ITU spec specifies the numbering plan. + +2010-07-12 17:21 +0000 [r275682] Jeff Peeler + + * 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 + + * 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 + + * 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 + + * configs/osp.conf.sample, apps/app_osplookup.c: Added support for + indirect work mode. + +2010-07-10 20:49 +0000 [r275509] Eliel C. Sardanons + + * 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 + + * 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 + + * 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 + + * 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 + + * main/features.c: Add missing ao2_iterator_destroy(). + + * apps/app_voicemail.c: Fix compile error. + +2010-07-09 19:46 +0000 [r275308] Mark Michelson + + * 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 + + * 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 + + * 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 + + * /, 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 + + * apps/app_meetme.c, channels/chan_iax2.c: Weird, no output and + Bamboo still fails... + +2010-07-09 18:24 +0000 [r275186] Matthew Nicholson + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * channels/chan_sip.c: Return logic of sip_debug_test_addr() to its + original functionality. + +2010-07-09 16:05 +0000 [r275028] Matthew Nicholson + + * 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 + + * 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 + + * 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] + + * 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 + + * include/asterisk/indications.h: Extend length limit on country + name in indications.conf. + +2010-07-09 11:06 +0000 [r274866] Olle Johansson + + * 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 + + * 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 + + * /: 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 + + * 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 + + * 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 + + * channels/chan_sip.c: Fixes some ref count issues introduced by + r274539 + +2010-07-07 18:32 +0000 [r274595-274639] Richard Mudgett + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * main/pbx.c: Uh, yeah. + +2010-07-05 13:53 +0000 [r273886] Paul Belanger + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * res/res_fax.c: Properly handle failures of fax->start_session() + FAX-177 + +2010-07-01 16:40 +0000 [r273427] David Vossel + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * apps/app_meetme.c: Don't start the sla thread unless we realy + need it + +2010-06-23 18:25 +0000 [r272145] Tilghman Lesher + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * main/file.c: file.c was truncating audio file formats to the + lower 32bits. + +2010-06-18 18:36 +0000 [r271336] Jeff Peeler + + * /: 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 + + * 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 + + * channels/sig_analog.c: Change expected operation from error to + debug message + +2010-06-17 00:30 +0000 [r271089] Paul Belanger + + * 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 + + * channels/sip/reqresp_parser.c: addition of more parse_uri test + cases + +2010-06-16 21:17 +0000 [r270987] Paul Belanger + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * channels/sip/dialplan_functions.c: Make SIP tests compile again. + +2010-06-07 22:56 +0000 [r268817-268818] Tilghman Lesher + + * 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 + + * 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 + + * 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 + + * 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 + + * main/manager.c: Event well was going dry. (issue #17234) + +2010-06-07 17:34 +0000 [r268690] Paul Belanger + + * 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 + + * res/res_smdi.c: Avoid unloading res_smdi twice. (closes issue + #17237) Reported by: pabelanger + +2010-06-07 15:51 +0000 [r268578] Richard Mudgett + + * 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 + + * 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 + + * CHANGES: Typo fix. + + * CHANGES: Grammatical error fix. + +2010-06-05 02:51 +0000 [r268321] Tilghman Lesher + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * codecs/codec_dahdi.c: Make compile again. + +2010-06-03 17:31 +0000 [r267537] Russell Bryant + + * channels/chan_usbradio.c: Don't stop Asterisk if chan_usbradio + isn't configured. + +2010-06-03 17:09 +0000 [r267492] Mark Michelson + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * apps/app_voicemail.c: Silence a compiler warning. + +2010-06-02 17:29 +0000 [r267065] Jeff Peeler + + * 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 + + * /, 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 + + * 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 + + * 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 + + * res/res_calendar_exchange.c: Use the correct ical.h file + +2010-06-01 21:28 +0000 [r266828] Tilghman Lesher + + * 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 + + * 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 + + * 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 + + * funcs/func_env.c: Needs to be wrapped in + + * 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 + + * 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 . + Since we now test for libical/ical.h, we can change the #includes + back to and remove the test which specifically + adds /usr/include/libical as an include directory. + +2010-05-28 22:50 +0000 [r266337-266385] Tilghman Lesher + + * 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 + + * 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 + + * configure, configure.ac, makeopts.in: More build fixes for + ical/neon and res_calendar_ews + +2010-05-27 20:08 +0000 [r266240] Jeff Peeler + + * pbx/pbx_realtime.c: fix compile error + +2010-05-27 19:25 +0000 [r266146-266238] Tilghman Lesher + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * addons/ooh323c/src/oochannels.c: Typos: 'succesful' (lintian) + +2010-05-24 22:21 +0000 [r265467] Terry Wilson + + * 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 + + * 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 + + * apps/app_rpt.c: Make app_rpt.c able to compile again. + +2010-05-24 19:42 +0000 [r265366] David Vossel + + * 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 + + * 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 + + * 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 + + * main/channel.c: fixes segfault when using generic plc + +2010-05-23 18:23 +0000 [r265227] Alexandr Anikin + + * addons/chan_ooh323.c: small changes to avoiding 'freeing unused + memory...' + +2010-05-21 22:46 +0000 [r265174] Richard Mudgett + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * main/cli.c: Fix compilation problem with previous commit. (issue + #16009) + +2010-05-19 15:29 +0000 [r264160] Kevin P. Fleming + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * res/res_timing_kqueue.c: Make happy green color come back + +2010-05-18 20:09 +0000 [r263810] Mark Michelson + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * CHANGES: Update CHANGES to reflect DAHDI buffer dialstring option + backport to 1.6.2 + +2010-05-16 16:31 +0000 [r263250] Tzafrir Cohen + + * 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 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 + + * 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 + + * 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 + + * channels/chan_dahdi.c: Fix inverted logic in cli command: ss7 set + debug on/off + +2010-05-13 20:25 +0000 [r263028] Tzafrir Cohen + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * channels/chan_dahdi.c: Avoid a crash on SS7 channels. + +2010-05-05 20:48 +0000 [r261364] Russell Bryant + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * configs/extensions.conf.sample: Pattern match fail. + +2010-04-29 15:33 +0000 [r260050] David Vossel + + * /, 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 + + * include/asterisk/extconf.h: Fix comment. + +2010-04-28 22:34 +0000 [r259957] Mark Michelson + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * main/ccss.c: Shuffle some casts to make builds on bamboo happier. + +2010-04-27 21:49 +0000 [r259527] Leif Madsen + + * /, 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 + + * /: 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * main/channel.c: Fix previous commit. + +2010-04-22 22:10 +0000 [r258673-258674] Jason Parker + + * 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 + + * 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 + + * 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 + + * apps/app_voicemail.c: Pass interactive = 0 and fix a compile + error. + +2010-04-22 19:08 +0000 [r258557] Jason Parker + + * 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 + + * 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 + Eliel C. Sardanons (LU1ALY) + 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 + + * doc/tex/channelvariables.tex: Add MEETMEBOOKID from r256019. + +2010-04-21 21:56 +0000 [r258433] Jeff Peeler + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * apps/app_voicemail.c: Bad merge fix + +2010-04-19 18:42 +0000 [r257851] Mark Michelson + + * 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 + + * 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 + + * configs/cdr_odbc.conf.sample: Removing unused configuration + parameters + +2010-04-16 21:22 +0000 [r257713] Dwayne M. Hubbard + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * main/event.c: Remove a debugging log entry. + +2010-03-27 23:51 +0000 [r255199] Alexandr Anikin + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * .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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * main/stdtime/localtime.c: Just in case of a race, send the signal + on interrupt. + +2010-03-17 19:06 +0000 [r253205] Leif Madsen + + * 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 + + * 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 + + * 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 + + * 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 + + * tests/test_time.c: Re-enable test_time on non-Linux. + +2010-03-16 19:36 +0000 [r252848] Sean Bright + + * 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 + + * 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 + + * 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 + + * 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 + + * codecs/gsm/Makefile: OSARCH is not inherited to this directory + +2010-03-16 18:36 +0000 [r252759] Russell Bryant + + * tests/test_time.c: Disable this test on non-Linux for now. + +2010-03-15 22:48 +0000 [r252709] Kevin P. Fleming + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * channels/chan_dahdi.c: add missing mfcr2_skip_category setting + +2010-03-12 19:43 +0000 [r251989] Tilghman Lesher + + * 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 + + * /: 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * addons/ooh323c/src/ooq931.c: small log issue from bug 0016664 + +2010-03-06 14:16 +0000 [r251137] Russell Bryant + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /: 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * main/test.c: adds Time field to "test show results" cli command + +2010-02-26 17:13 +0000 [r249101-249105] Mark Michelson + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * res/res_rtp_asterisk.c: Improve support for RTCP reports without + report blocks + +2010-02-19 18:38 +0000 [r248003] Moises Silva + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * apps/app_osplookup.c, UPGRADE.txt, CHANGES: Updated doc for OSP + lookup application. + +2010-02-11 21:57 +0000 [r246299-246338] David Vossel + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * channels/sip: Add ignore *.i files property to the new + channels/sip directory. + +2010-02-03 20:48 +0000 [r244598] Jeff Peeler + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * channels/iax2-parser.c: Informational message, not an error. + +2010-01-28 18:35 +0000 [r243780-243860] Russell Bryant + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * main/acl.c: Add missing argument to ast_calloc calls. + +2010-01-21 21:05 +0000 [r242043] Olle Johansson + + * main/acl.c: Make sure we initialize the ast_ha structure with + ast_calloc + +2010-01-21 15:27 +0000 [r241938] Sean Bright + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * channels/chan_vpb.cc: Fix up compile breakage from + ast_tvdiff_ms() API change. + +2010-01-20 08:18 +0000 [r241416] Alec L Davis + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * funcs/func_strings.c, CHANGES: Make HASHes inheritable across + channel creation. + +2010-01-18 18:00 +0000 [r240973-240974] David Ruggles + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * res/res_musiconhold.c: Get MoH building on OpenSolaris. + +2010-01-15 23:50 +0000 [r240629] Tilghman Lesher + + * Makefile, main/asterisk.c: Err, oops, it was already the way I + intended. + +2010-01-15 23:09 +0000 [r240548-240552] Russell Bryant + + * 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 + + * res/res_timing_timerfd.c: Clarify error message in + res_timing_timerfd. + +2010-01-15 21:42 +0000 [r240421-240500] Tilghman Lesher + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * configure: Update configure script for an OSP toolkit related + change. + +2010-01-15 16:28 +0000 [r240328] Kevin P. Fleming + + * configs/sip.conf.sample: Clarify RTP NAT handling a bit. + +2010-01-14 23:13 +0000 [r240226-240271] Sean Bright + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * channels/sig_pri.c: Fix using the wrong pointer type in + do_idle_thread(). + +2010-01-07 20:42 +0000 [r238361-238492] David Vossel + + * 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 + + * configs/sip.conf.sample: Document the usefulness of explicit + udp:// in the register string + +2010-01-06 21:45 +0000 [r238231] Tilghman Lesher + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * channels/chan_sip.c: Whoa, duplicate setting (dead code). + +2010-01-05 23:08 +0000 [r237920] David Vossel + + * 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 + + * apps/app_dial.c: Mismerged a bit. + +2010-01-05 19:29 +0000 [r237839] David Vossel + + * 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 + + * 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 + + * 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 + + * 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 + + * apps/app_mixmonitor.c: Make CLI command 'mixmonitor start|stop + 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * channels/chan_sip.c: global_contact_ha was renamed in trunk + +2010-01-02 09:54 +0000 [r237136] Olle Johansson + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * include/asterisk/strings.h: Is it Friday yet? + +2009-12-15 23:41 +0000 [r235226] Jeff Peeler + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * res/res_musiconhold.c: Missed a case that emits a WARNING where + none is warranted. + +2009-12-10 17:31 +0000 [r234173] Jeff Peeler + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * main/utils.c: fixes incorrect logic in ast_uri_encode issue + #16299 + +2009-12-07 23:10 +0000 [r233577] Atis Lezdins + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * apps/app_osplookup.c: Added full number portability parameter + support. + +2009-11-12 23:43 +0000 [r229750-229754] Jason Parker + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * channels/chan_sip.c: Reverted revision 201717. (closes issue + 0016175) Reported by: paul-tg + +2009-11-10 15:27 +0000 [r229093] David Vossel + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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() R Reverse charging indication (Collect calls) + The earlier Dial(DAHDI/g1[/K][/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 + + * 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 + + * 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 + + * main/frame.c: Fix compiler warning gcc 4.2.4 found + +2009-11-06 19:47 +0000 [r228620] Matthew Nicholson + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * funcs/func_cdr.c: Fix XML in func_cdr.c + +2009-11-05 22:12 +0000 [r228191-228196] Tilghman Lesher + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * include/asterisk/unaligned.h: Fixes for gcc 4.4 + +2009-11-04 20:13 +0000 [r227759] Matthew Nicholson + + * 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 + + * /, 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 + + * main/frame.c: Add a couple more casts so that code compiles + correctly. + +2009-11-04 16:35 +0000 [r227645] Tilghman Lesher + + * 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 + + * channels/chan_dahdi.c: fix trunk building + +2009-11-04 16:17 +0000 [r227579-227615] Tilghman Lesher + + * 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 + + * 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 + + * apps/app_queue.c: Don't crash when state_interface is NULL. + +2009-11-03 22:13 +0000 [r227462-227464] Russell Bryant + + * 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 + + * 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 + + * 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 + + * configs/queues.conf.sample, apps/app_queue.c: Add support for + using a hint when configuring a state interface using the format + hint:@. (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 + + * 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 + + * 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 + + * 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 + + * /: 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 + + * channels/chan_mgcp.c: Code guidelines fixes only + +2009-11-03 17:12 +0000 [r227238] David Vossel + + * 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 + + * 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 + + * /: 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * main/http.c: Adding external reference for doxygen + +2009-11-02 18:08 +0000 [r226890] Joshua Colp + + * 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 + + * 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/[:]- 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * channels/chan_local.c: Doxygen documentation update + +2009-10-28 20:50 +0000 [r226453] Tzafrir Cohen + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * Makefile: Adding compile time flags for Snow Leopard, Leopard and + some other animals + +2009-10-27 20:22 +0000 [r226159] Tilghman Lesher + + * 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 + + * res/res_http_post.c: Don't prepend the URI prefix to the post + directory + +2009-10-27 13:30 +0000 [r226060] Joshua Colp + + * 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 + + * /, 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 + + * 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 + + * 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 + + * channels/chan_dahdi.c: Make conditionals create previous code + when libpri/ss7 are present. + +2009-10-26 13:29 +0000 [r225767-225836] Tzafrir Cohen + + * 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 + + * 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 + + * 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][/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 + + * 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 + + * channels/chan_sip.c: Fixes an iterator memory leak and + uninitialized memory + +2009-10-23 14:02 +0000 [r225582] Kevin P. Fleming + + * 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 + + * 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 + + * channels/sig_pri.c: Search for the subaddress only within the + extension section of the dial string. + Dial(DAHDI/(g|G|r|R)[c|r|d][/extension]) + +2009-10-22 19:55 +0000 [r225445] David Vossel + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * channels/sig_pri.c: Make PRI_SUBCMD_xxx handling subaddress + friendly. + +2009-10-20 22:09 +0000 [r224856] Tilghman Lesher + + * 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 + + * /, 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 + + * CHANGES: Added information to CHANGES about the dynamic range + compression feature added to dahdi. + +2009-10-19 23:47 +0000 [r224671] Kevin P. Fleming + + * 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 + + * 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 + + * 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 + + * formats/format_siren14.c: Remove useless debugging message. + +2009-10-19 15:50 +0000 [r224527] Tilghman Lesher + + * doc/janitor-projects.txt: Remove a completed project and add + another + +2009-10-19 14:32 +0000 [r224491] Joshua Colp + + * 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 + + * 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 + + * include/asterisk/app.h, main/app.c: Remove unnecessary typedef + +2009-10-17 02:01 +0000 [r224331-224335] Jeff Peeler + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * res/res_calendar.c: use Calendar: instead of Calendar/ for + devstate + +2009-10-13 17:11 +0000 [r223911-223912] Richard Mudgett + + * 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 + + * 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 + + * 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 + + * configs/iax.conf.sample: Clarifies trunkmaxsize, trunkfreq, and + trunkmtu iax2 options SWP-151 + +2009-10-12 15:32 +0000 [r223652-223693] Kevin P. Fleming + + * /: 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 + + * 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 + + * /: 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /: 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * UPGRADE.txt, CHANGES: Move DAHDI/ISDN channel naming note from + CHANGES to UPGRADE.txt. + +2009-10-01 20:09 +0000 [r221705] Tilghman Lesher + + * 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 + + * 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-- + +2009-10-01 19:33 +0000 [r221697] David Vossel + + * 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 + + * 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 + + * /, 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 + + * 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 + + * channels/chan_sip.c: Cleaned up merge from r221432 + +2009-09-30 21:15 +0000 [r221436] Matthias Nick + + * apps/app_queue.c: Prevents from division by zero + +2009-09-30 20:40 +0000 [r221432] Matthew Nicholson + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * channels/chan_dahdi.c, channels/sig_pri.c: Miscellaneous minor + changes. + +2009-09-28 19:11 +0000 [r220721] Sean Bright + + * /, 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 + + * 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 + + * 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 + + * 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 + + * include/asterisk/aes.h: Allow AES to compile, when OpenSSL is not + present. + +2009-09-25 19:56 +0000 [r220543] Richard Mudgett + + * channels/sig_pri.c: Reduce indentation in sig_pri_available(). + +2009-09-25 14:50 +0000 [r220494-220496] Kevin P. Fleming + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * channels/chan_sip.c: Ensure no spaces exist before "refresher=" + when doing the comparison. + +2009-09-17 16:25 +0000 [r219230] Sean Bright + + * apps/app_chanspy.c: Get this compiling under dev-mode. + +2009-09-17 15:18 +0000 [r219139] Matthew Nicholson + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * channels/chan_sip.c: upward bound checking for port string to int + conversion + +2009-09-15 16:15 +0000 [r218586] Matthew Nicholson + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /: 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 + + * channels/chan_sip.c: Fix off-by-one error when reading SDP sent + over TCP. + +2009-09-15 10:24 +0000 [r218465] Tzafrir Cohen + + * channels/chan_dahdi.c: Fix false error message on + DAHDI_EVENT_REMOVED (RESULT_SUCCESS == 0) + +2009-09-14 22:38 +0000 [r218430] Jeff Peeler + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * channels/chan_dahdi.c: get rid of mfcr2 monitor thread condition, + is problematic + +2009-09-12 13:08 +0000 [r218107] Michiel van Baak + + * res/res_rtp_asterisk.c: use the actual given ip address for 'rtp + set debug ip ' 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * channels/chan_sip.c: Don't assign UINT_MAX to an INT. + +2009-09-10 18:17 +0000 [r217638] Tilghman Lesher + + * 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 + + * 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 + + * channels/chan_dahdi.c: Fix available() for SS7, MFC/R2, and + pseudo channels. + +2009-09-09 21:48 +0000 [r217524] Moises Silva + + * channels/chan_dahdi.c: ast_log replaced for ast_verbose in MFCR2 + event notifications + +2009-09-09 20:09 +0000 [r217482] Olle Johansson + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * apps/app_meetme.c: Fix compilation of app_meetme. Reported by + ebroad in #asterisk-bugs + +2009-09-08 21:17 +0000 [r217236] Richard Mudgett + + * channels/sig_pri.c: Remove duplicate entry in the sig_pri_pri + private pointer array. + +2009-09-08 20:28 +0000 [r217199] Tilghman Lesher + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * apps/app_voicemail.c: Use ast_free() instead of free(). + +2009-09-04 17:50 +0000 [r216547-216551] Tilghman Lesher + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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. + +2009-09-04 10:48 +0000 [r216264] Russell Bryant + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * channels/chan_sip.c, channels/chan_skinny.c: add Parkinglot info + to sip show peer and skinny show line 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * channels/chan_sip.c: Removing whitespace that causes red dots in + reviewboard + +2009-08-31 22:02 +0000 [r215069-215070] Tilghman Lesher + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * include/asterisk/sched.h: Fix some incorrect documentation of + sched_thread functions. + +2009-08-28 16:50 +0000 [r214360-214611] Tilghman Lesher + + * 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 + + * 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 + + * configs/chan_dahdi.conf.sample: Minor punctuation change. + +2009-08-26 16:53 +0000 [r214199] Tilghman Lesher + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * channels/chan_sip.c: Clarifying comments in sip_register, and + removing a dead section + +2009-08-21 22:22 +0000 [r213716] David Vossel + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * channels/chan_dahdi.c: increment the mfcr2 monitor count when + clearing the call request + +2009-08-21 03:48 +0000 [r213450] Terry Wilson + + * main/loader.c: Make LOAD_ORDER actually work + +2009-08-20 22:13 +0000 [r213414] Tilghman Lesher + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * main/features.c: Don't blow up on a NULL cdr. Reported in + #asterisk-dev. + +2009-08-18 23:53 +0000 [r213007] Richard Mudgett + + * 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 + + * /: 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 + + * 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 + + * apps/app_meetme.c: Small doxygen changes + +2009-08-18 16:38 +0000 [r212764] Sean Bright + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * funcs/func_curl.c: Add SSL_VERIFYPEER, as requested on the -users + list + +2009-08-13 17:33 +0000 [r212199] Richard Mudgett + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /: 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * contrib/scripts/install_prereq: add OpenBSD to the install_prereq + script + +2009-07-25 12:28 +0000 [r208813-208848] Michiel van Baak + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * main/cel.c: Resolve compiler warning on mac. + +2009-07-22 22:42 +0000 [r208155] Jeff Peeler + + * 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 + + * /, 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 + + * 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 + + * 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 + + * include/asterisk/channel.h: Remove trailing whitespace. + +2009-07-22 14:35 +0000 [r208017] Mark Michelson + + * 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 + + * 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 + + * /, 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 + + * channels/chan_dahdi.c: whitespace fix only + +2009-07-21 22:22 +0000 [r207925] Russell Bryant + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /: 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 + + * doc/voicemail_odbc_postgresql.txt: Add flag here, too (as + requested by jsmith) + +2009-07-17 22:07 +0000 [r207225] David Vossel + + * 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 + + * doc/tex/odbcstorage.tex, UPGRADE.txt: Document the "flag" field + in the voicemessages table. + +2009-07-17 19:37 +0000 [r207095-207156] Jeff Peeler + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * channels/chan_sip.c: callerid(num) is wrong when username is + missing A domain only sip uri 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 + + * /, 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 + + * configs/chan_dahdi.conf.sample: fix a typo in sample config file + for option change + +2009-07-14 20:14 +0000 [r206567] Tilghman Lesher + + * 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 + + * 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 + + * apps/app_dial.c: I AM A TERRIBLE PERSON + +2009-07-14 17:01 +0000 [r206489] Richard Mudgett + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * .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 + + * 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 + + * 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 + + * main/udptl.c: Update comments about the level of T.38 support in + Asterisk. + +2009-07-10 17:39 +0000 [r205878] Mark Michelson + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * main/say.c: Oops, fixing build + +2009-07-08 16:54 +0000 [r205216] David Vossel + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * bootstrap.sh: FreeBSD now has autoconf 2.62 in the ports, 2.61 + has disappeared. + +2009-07-07 21:10 +0000 [r205086] Tilghman Lesher + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * apps/app_voicemail.c: Remove an unnecessary #ifdef + +2009-06-30 19:59 +0000 [r204530-204532] Mark Michelson + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * CHANGES: Another CHANGES spelling fix. + +2009-06-27 10:04 +0000 [r203960-203962] Russell Bryant + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * main/ast_expr2.fl, main/ast_expr2.c: fixes a few redundant + conditions (issue #15269) + +2009-06-25 21:34 +0000 [r203443] Richard Mudgett + + * channels/chan_dahdi.c: Picking nits + +2009-06-25 21:22 +0000 [r203402] Jeff Peeler + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * main/logger.c: Pass a logmsg to ast_log_vsyslog instead of + separate arguments. + +2009-06-25 16:18 +0000 [r203126] Doug Bailey + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * doc/tex/cdrdriver.tex: Clean up section hierarchy for the CDR + chapter. + +2009-06-23 22:08 +0000 [r202761] Matthew Fredrickson + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * channels/chan_sip.c: Use sched_yield() instead of usleep(1) + +2009-06-19 20:24 +0000 [r201994] David Vossel + + * /, 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 + + * CHANGES: Add note about the addition of calendar support + +2009-06-19 15:47 +0000 [r201904] Tilghman Lesher + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * CREDITS: Update my e-mail address (thanks for the props, russell + :)) + +2009-06-16 21:10 +0000 [r200985-201139] Kevin P. Fleming + + * 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 + + * 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 + + * 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 + + * CREDITS: Add Sean Bright to CREDITS - Thanks, Sean! + +2009-06-16 14:12 +0000 [r200841-200878] Eliel C. Sardanons + + * /: 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 + + * 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 + + * 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 + + * 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 + + * doc/tex/channelvariables.tex: Add missing closure of verbatim + environment. + +2009-06-16 01:03 +0000 [r200519-200726] Kevin P. Fleming + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * sample.call: Remove some trailing whitespace and steal revision + 200000. + +2009-06-10 20:15 +0000 [r199958] Mark Michelson + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * Makefile: Fix all of the parallel build warnings issued when + running make -j#. + +2009-06-09 16:22 +0000 [r199743] David Vossel + + * 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 + + * doc/janitor-projects.txt: Add sigaction janitor + +2009-06-08 19:33 +0000 [r199630] Sean Bright + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * channels/iax2-parser.c: Tell the IAX2 parser about more control + frame types. + +2009-06-01 20:57 +0000 [r198727] Mark Michelson + + * 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 + + * 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 + + * include/asterisk/frame.h: Minor whitespace fix. + +2009-06-01 19:37 +0000 [r198661] Eliel C. Sardanons + + * 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 + + * contrib/scripts/meetme.sql: Add information for new meetme + realtime fields + +2009-06-01 17:53 +0000 [r198561-198597] Eliel C. Sardanons + + * 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 + + * 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 + + * 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 + + * 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 + + * funcs/func_strings.c: Fix documentation for FIELDQTY. + +2009-05-31 02:09 +0000 [r198442] Eliel C. Sardanons + + * main/Makefile: Filter the say.o object, it is being added later. + +2009-05-31 01:40 +0000 [r198438] Russell Bryant + + * main/manager.c: Constification and remove some unused code. + +2009-05-31 01:22 +0000 [r198437] Eliel C. Sardanons + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * res/res_calendar.c: Add a couple of TODO items so I don't forget + +2009-05-29 20:06 +0000 [r198146] Russell Bryant + + * 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 + + * 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 + + * 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 + + * CREDITS: Apply anti-spam obfuscation to an email address. + +2009-05-29 19:04 +0000 [r198072] Matthew Nicholson + + * 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 + + * main/file.c: Fix a memory leak of the write buffer when writing a + file. + +2009-05-29 18:15 +0000 [r198000] Sean Bright + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * configs/calendar.conf.sample: Make note of Exchange calendar + support limitations + +2009-05-28 20:36 +0000 [r197775] Kevin P. Fleming + + * 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 + + * 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 + + * 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 + + * channels/chan_local.c: Add missing lock to local_indicate + function for connected line frames. + +2009-05-28 18:45 +0000 [r197697] Joshua Colp + + * 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 + + * /, 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 + + * 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 + + * /: 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * main/xml.c: Revert commit 192032. This define is needed on Mac OS + X. + +2009-05-27 22:42 +0000 [r197338] Russell Bryant + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 ' 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * channels/chan_dahdi.c: set MFCR2_CATEGORY just when starting the + pbx + +2009-05-22 21:11 +0000 [r196417] Sean Bright + + * 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 + + * 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 + + * 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 + + * 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: AMI action + synopsis. <-- for ActionID Description + ... AMI action + description ... + + +2009-05-22 16:53 +0000 [r196272] Tilghman Lesher + + * main/astmm.c: Two more minor fixes due to constification + +2009-05-22 16:51 +0000 [r196270] Sean Bright + + * 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 + + * channels/chan_sip.c: s/it's/its/ + +2009-05-22 16:20 +0000 [r196246] Russell Bryant + + * channels/chan_dahdi.c: resolve compiler warning + +2009-05-22 16:10 +0000 [r196227] Sean Bright + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * main/features.c: Pass connected line updates along during a + bridge. + +2009-05-21 17:15 +0000 [r195949] Sean Bright + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 ' it was autocompleting + everything else after the 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 + + * 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 + + * 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 + + * cdr/cdr_custom.c: Remove some unused code. + +2009-05-18 16:29 +0000 [r195266] Richard Mudgett + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /: 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 + + * 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: + (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 + + * /, 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 + + * /, 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 + + * CREDITS: add eliel + +2009-05-15 13:23 +0000 [r194635] Eliel C. Sardanons + + * 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: ... + 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * funcs/func_devstate.c: Fix the spelling of UNAVAILABLE in + func_devstate CLI completion. + +2009-05-08 14:52 +0000 [r193263] David Vossel + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * apps/app_forkcdr.c: Commit documentation changes related to issue + #14801. (issue #14801) + +2009-05-04 16:24 +0000 [r192059] Kevin P. Fleming + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * main/audiohook.c: Drop my IRC nickname. + +2009-05-01 09:50 +0000 [r191418] TransNexus OSP Development + + * configs/osp.conf.sample, apps/app_osplookup.c: Made security + features optional. + +2009-04-30 21:42 +0000 [r191411] Kevin P. Fleming + + * 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 + + * 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 + + * 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 + + * 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 + + * res/res_phoneprov.c: fix typos + +2009-04-29 22:23 +0000 [r191211] Tilghman Lesher + + * 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 + + * 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 + + * 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 + + * 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 + + * pbx/pbx_config.c: Removing crufty code that is no longer + necessary. Code cleanup. + +2009-04-29 14:39 +0000 [r191028] David Vossel + + * 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 + + * 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 + + * 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 + + * doc/tex/cdrdriver.tex: UniqueID column has a maximum size of 150 + +2009-04-28 14:15 +0000 [r190861-190865] Kevin P. Fleming + + * 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 + + * apps/app_osplookup.c: Updated for OSP Toolkit 3.5. + +2009-04-27 21:22 +0000 [r190735-190797] Richard Mudgett + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * funcs/func_strings.c: Fix example that could fail in certain + circumstances + +2009-04-22 21:38 +0000 [r190093] Tilghman Lesher + + * 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 + + * 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 + + * 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 + + * 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 + + * main/features.c: Fix call parking callback. Pipes -> Commas. + +2009-04-22 16:01 +0000 [r189911] Tilghman Lesher + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * main/asterisk.c: Fix copy/paste error with 'transmit silence' + flag. + +2009-04-17 15:44 +0000 [r189010] Matthew Nicholson + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * apps/app_queue.c: Fix a couple of queue member reference leaks. + +2009-04-14 17:40 +0000 [r188413] Joshua Colp + + * 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 + + * 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 + :, where: ::= { 'O' } // 'O' – Out Of + Service ::= { '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 pri service + enable channel 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * tests/test_heap.c, tests/test_sched.c: Disable test modules by + default. + +2009-04-10 15:59 +0000 [r187674] Tilghman Lesher + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * apps/app_dial.c: More changes concerning r187426. Revised where + locks are placed. + +2009-04-09 19:10 +0000 [r187491] Jeff Peeler + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /: 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 + + * 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 + + * 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 + + * channels/chan_sip.c: Remove duplicate prototype for temp_peer(). + +2009-04-08 17:08 +0000 [r187050] Tilghman Lesher + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * channels/chan_sip.c: Add lastms to the require API call. + +2009-04-08 00:09 +0000 [r186833-186842] Mark Michelson + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /: Remove merged branch properties accidentally merged to trunk. + +2009-04-03 22:41 +0000 [r186525] Mark Michelson + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * utils: ignore copied (generated) file + +2009-03-31 22:12 +0000 [r185600-185604] Mark Michelson + + * 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 + + * 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 + + * /, 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 + + * 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 + , 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 + + * /, 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: + + + Review: http://reviewboard.digium.com/r/181/ + ........ + +2009-03-31 14:53 +0000 [r185261] Russell Bryant + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * channels/chan_dahdi.c: Remove duplicate inside the + xml documentation. + +2009-03-19 20:30 +0000 [r183436] David Vossel + + * 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 + + * /: 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 + + * /, 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 + + * 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 + + * /, 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 + + * res/res_odbc.exports: 2 symbols defined when DEBUG_THREADS + +2009-03-19 16:28 +0000 [r183172] David Vossel + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * res/res_ael_share.exports (added): allow this module to export + everything for now + +2009-03-18 21:18 +0000 [r183028] Jeff Peeler + + * 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 + + * main/asterisk.exports: Fixing a lost symbol in manager.c + +2009-03-18 11:40 +0000 [r182848-182883] Kevin P. Fleming + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * UPGRADE.txt, CHANGES: Update UPGRADE.txt and CHANGES for 1.6.3 +