--- /dev/null
+2013-09-19 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 11.6.0-rc1 Released.
+
+2013-09-18 23:36 +0000 [r399442] Richard Mudgett <rmudgett@digium.com>
+
+ * main/udptl.c: UDPTL: Backport some fixes from v12 that should be
+ in v11. Backported the following as applied to udptl.c: *
+ -r398020 Fixup udpdl defaults if config file not present. *
+ -r398533 Fixup improper use of ao2_global_obj_replace().
+
+2013-09-18 19:55 +0000 [r399403] Kinsey Moore <kmoore@digium.com>
+
+ * main/abstract_jb.c, /: Fix jitter buffer log file creation This
+ adjusts '/'-to-'#' replacement to replace all instances of '/'
+ instead of just the first to ensure that the jitter buffer log
+ file gets the correct name as per Richard Kenner's suggestion.
+ (closes issue ASTERISK-21036) Reported by: Richard Kenner
+ ........ Merged revisions 399402 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-18 17:22 +0000 [r399353-399373] Matthew Jordan <mjordan@digium.com>
+
+ * /, build_tools/prep_tarball: Update prep_tarball with new
+ documentation files on the Asterisk wiki This will now pull both
+ a command reference for the version being prepared, as well as an
+ Admin Guide that applies to all versions of Asterisk. (issue
+ ASTERISK-22439) Reported by: Olle Johansson ........ Merged
+ revisions 399351 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * bridges/bridge_softmix.c: Add a WARNING in bridge_softmix when a
+ timing module isn't loaded If bridge_softmix fails to be created
+ because no timing source is present in Asterisk, this will
+ currently fail gracefully but with (most likely) a generic error
+ message by whatever module tried to create the softmix bridge.
+ This patch adds a more explicit warning so you can actually
+ diagnose and fix the problem. Review:
+ https://reviewboard.asterisk.org/r/2857/
+
+2013-09-18 01:34 +0000 [r399305] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, main/features.c: Fix Segfault When Syntax Of A Line Under
+ [applicationmap] Is Invalid When processing the lines under the
+ [applicationmap] context in features.conf, a segfault occurs from
+ attempting to process a line with an invalid syntax (basically
+ missing most of the arguments). Example: [applicationmap]
+ automon=*6 * This patch moves the checking for empty arguments to
+ before they are accessed. * Also, checked the "todo" comment and
+ removed it. Some applications do not require arguments. (closes
+ issue ASTERISK-22416) Reported by: CGI.NET Tested by: CGI.NET
+ Patches: asterisk-22416-check-syntax-first_v2.diff by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2803 ........ Merged revisions
+ 399304 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-17 18:32 +0000 [r399222-399267] Kevin Harwell <kharwell@digium.com>
+
+ * main/asterisk.c, main/logger.c: Remote console: more output
+ discrepancies The remote console continued to have issues with
+ its output. In this case CLI command output would either not show
+ up (if verbose level = 0) or would contain verbose prefixes (if
+ verbose level > 0) once log messages were sent to the remote
+ console. The fix now now adds verbose prefix data to all new
+ lines contained in a verbose log string. (closes issue
+ ASTERISK-22450) Reported by: David Brillert (closes issue
+ AST-1193) Reported by: Guenther Kelleter Review:
+ https://reviewboard.asterisk.org/r/2825/
+
+ * apps/confbridge/conf_state_multi_marked.c: Confbridge: empty
+ conference not being torn down Confbridge would not properly tear
+ down an empty conference bridge when all users were kicked via
+ end_marked=yes and at least one user was also set to wait_marked.
+ This occurred because while end_marked users were being kicked
+ and at least one was also set to wait_marked then the leave
+ wait_marked handler would be called on that user, but there would
+ be no waiting user (still considered active). The waiting users
+ would decrement and now be negative. The conference would remain,
+ but be put into an inactive state. The solution was to move from
+ the active list to the wait list, those users with wait_marked
+ set right before kicking. This allows both the active and wait
+ users to decrement correctly and the confbridge to tear down
+ properly. A crashed also occurred when trying to list the
+ specific conference from the CLI. This happened because the
+ conference specified was invalid. Since the conference properly
+ tears down now there is no way to reference it thus alleviating
+ the crash as well. (closes issue ASTERISK-21859) Reported by:
+ Chris Gentle Review: https://reviewboard.asterisk.org/r/2848/
+
+2013-09-16 16:42 +0000 [r399159] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_iax2.c, /: chan_iax2: Fix saving the wrong expiry
+ time in astdb. When a new IAX2 client registers, the astdb
+ database is updated with the value of minregexpire defined in
+ iax.conf instead of using the expiry time that is provided by the
+ client. The provided expiry time of the client is updated after
+ inserting the astdb entry. As a consequence, restarting or
+ reloading asterisk creates clients whose registration may expire
+ before they reregister. The clients are therefore unavailable
+ after minregexpire seconds until they reregister. * Move updating
+ of the expiry time to before inserting into the astdb. (closes
+ issue ASTERISK-22504) Reported by: Stefan Wachtler Patches:
+ chan_iax2.c.patch (license #6533) patch uploaded by Stefan
+ Wachtler ........ Merged revisions 399158 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-13 20:49 +0000 [r399099] David M. Lee <dlee@digium.com>
+
+ * main/astobj2.c, /: Don't write to /tmp/refs when REF_DEBUG is not
+ defined. If MALLOC_DEBUG is enabled, then the debug destructor
+ for the container is used, which would erroneously write to
+ /tmp/refs. This patch only uses the debug destructor if ref_debug
+ is used. (closes issue ASTERISK-22536) ........ Merged revisions
+ 399098 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-13 13:48 +0000 [r399034] Kinsey Moore <kmoore@digium.com>
+
+ * /, apps/app_meetme.c: Fix several crashes in MeetMeAdmin This
+ change ensures that MeetMeAdmin commands requiring a user
+ actually get a user and fixes another issue where an extra
+ dereference could occur for a last-entered user being ejected if
+ a user identifier was also provided. (closes issue
+ ASTERISK-21907) Reported by: Alex Epshteyn Review:
+ https://reviewboard.asterisk.org/r/2844/ ........ Merged
+ revisions 399033 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-12 20:19 +0000 [r398986] Jonathan Rose <jrose@digium.com>
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c: chan_sip:
+ Revert r398835 due to failing tests involving originate (issue
+ ASTERISK-22424) Reported by: Jonathan Rose ........ Merged
+ revisions 398977 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-12 00:02 +0000 [r398881-398885] Rusty Newton <rnewton@digium.com>
+
+ * /, apps/app_queue.c: 'queue add member' help text correction You
+ are adding dial strings to the queue, not channels. An aribitrary
+ string could be used, but you are typically referencing a
+ channel. Correcting the command help text. (issue ASTERISK-22263)
+ (closes issue ASTERISK-22263) Reported By: Rusty Newton ........
+ Merged revisions 398884 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * configs/chan_dahdi.conf.sample, /: Documentation fix -
+ waitfordialtone is not boolean, it's time in milliseconds
+ Changing text in chan_dahdi.conf sample to be accurate. (issue
+ ASTERISK-22308) (closes issue ASTERISK-22308) Reported By:
+ Malcolm Davenport ........ Merged revisions 398880 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-11 19:46 +0000 [r398836] Jonathan Rose <jrose@digium.com>
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c: chan_sip:
+ Reject calls without prior SDP on 200 OK If we receive a 200 OK
+ without SDP, we will now check to see if the remote address has
+ been established for that channel's RTP session and if the to tag
+ for that channel has changed from the most recent to tag in a
+ response less than 200. If either a change has been made since
+ the last to-tag was received or the remote address is unset, then
+ we will drop the call. (closes issue ASTERISK-22424) Reported by:
+ Jonathan Rose Review:
+ https://reviewboard.asterisk.org/r/2827/diff/#index_header
+ ........ Merged revisions 398835 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-11 18:01 +0000 [r398820] Russell Bryant <russell@russellbryant.com>
+
+ * configs/confbridge.conf.sample: Fix typo in
+ confbridge.conf.sample The denoise filter requires func_speex,
+ not codec_speex. Fix this in the description of the denoise=yes
+ option in confbridge.conf.
+
+2013-09-10 17:56 +0000 [r398758] Richard Mudgett <rmudgett@digium.com>
+
+ * main/event.c, res/res_musiconhold.c, main/indications.c,
+ main/asterisk.c, main/xmldoc.c, main/cli.c, /,
+ funcs/func_dialgroup.c, main/heap.c: Fix incorrect usages of
+ ast_realloc(). There are several locations in the code base where
+ this is done: buf = ast_realloc(buf, new_size); This is going to
+ leak the original buf contents if the realloc fails. Review:
+ https://reviewboard.asterisk.org/r/2832/ ........ Merged
+ revisions 398757 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-10 17:48 +0000 [r398749-398753] David M. Lee <dlee@digium.com>
+
+ * utils/check_expr.c, /: Fixed utils directory breakage from
+ r398748, this time with extra hate. ........ Merged revisions
+ 398752 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * utils/ael_main.c, utils/conf2ael.c, utils/check_expr.c, /: Fixed
+ utils directory breakage from r398648 ........ Merged revisions
+ 398748 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-09 23:21 +0000 [r398721] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/astmm.c: MALLOC_DEBUG: Change fence magic number to be
+ completely different from the freed magic number. Race conditions
+ between freeing a nul terminated string and ast_strdup()'ing it
+ are more likely to be detected if the fence and freed magic
+ numbers are completely different. ........ Merged revisions
+ 398703 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-09 20:02 +0000 [r398649] David M. Lee <dlee@digium.com>
+
+ * main/lock.c, /, main/utils.c, include/asterisk/lock.h: Fix
+ DEBUG_THREADS when lock is acquired in __constructor__ This patch
+ fixes some long-standing bugs in debug threads that were
+ exacerbated with recent Optional API work in Asterisk 12. With
+ debug threads enabled, on some systems, there's a lock ordering
+ problem between our mutex and glibc's mutex protecting its module
+ list (Ubuntu Lucid, glibc 2.11.1 in this instance). In one
+ thread, the module list will be locked before acquiring our
+ mutex. In another thread, our mutex will be locked before locking
+ the module list (which happens in the depths of calling
+ backtrace()). This patch fixes this issue by moving backtrace()
+ calls outside of critical sections that have the mutex acquired.
+ The bigger change was to reentrancy tracking for
+ ast_cond_{timed,}wait, which wrongly assumed that waiting on the
+ mutex was equivalent to a single unlock (it actually suspends all
+ recursive locks on the mutex). (closes issue ASTERISK-22455)
+ Review: https://reviewboard.asterisk.org/r/2824/ ........ Merged
+ revisions 398648 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-07 00:59 +0000 [r398510-398618] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_xmpp.c: Prevent XMPP timeout on blank responses Sometimes
+ the Google Voice servers have a bad habit of sending out 1 byte
+ replies to the xmpp resource. When a blank 1 byte reply is
+ received from the socket the buffer attempts to wait (endlessly)
+ for the rest of the reply from google which effectively blocks
+ the socket and google voice calls will no longer come into the
+ server. This patch allows the xmpp module to correctly detect
+ empty packets and send out ping replies to google. It also sets a
+ socket timeout on the default socket which prevents the xmpp
+ socket from closing and preventing future google voice calls from
+ coming into the server. Furthermore instead of sending an empty
+ reply back to google we send a proper xmpp ping reply back. This
+ also adds several more socket messages. (closes issue
+ ASTERISK-22347) Reported by: Andrew Nagy Review:
+ https://reviewboard.asterisk.org/r/2771 Patches: xmpp_fix_1.diff
+ uploaded by Andrew Nagy (License #6524)
+
+ * /, res/res_xmpp.c, res/res_jabber.c: Commit the remainder of
+ r398523 This is a missing part of the commit in revision 398523
+ that corrects the name of a variable. (issue ASTERISK-22435)
+ ........ Merged revisions 398576 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, res/res_xmpp.c, res/res_jabber.c: Fix Jabber/XMPP distributed
+ MWI The mailbox and context are swapped on the receiving end for
+ all users of Jabber and XMPP distributed MWI in Asterisk 1.8 and
+ all more recent versions. This swaps those values to be correct
+ when publishing to the internal event system from Jabber/XMPP
+ distributed MWI state. (closes issue ASTERISK-22435) Reported by:
+ abelbeck Tested by: Michael Keuter Patches:
+ asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by
+ abelbeck asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch
+ uploaded by abelbeck ........ Merged revisions 398523 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_h323.c: Fix chan_h323 compilation This fixes the
+ things in chan_h323 that were missed or ignored in the great
+ channel opaquification and gets chan_h323 back into a compiling
+ state. (closes issue ASTERISK-22365) Reported by: Dmitry Melekhov
+ Patches: chan_h323.patch uploaded by Dmitry Melekhov
+
+2013-09-05 19:13 +0000 [r398302-398457] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_iax2.c, /: chan_iax2: Reduce indentation in
+ __attempt_transmit(). * Reduce indentation in
+ __attempt_transmit(). * Don't update the static last error time
+ variable every time in __schedule_action() and socket_read().
+ ........ Merged revisions 398456 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_iax2.c, /: chan_iax2: Fix stray reference to worker
+ thread idle_list. * Fix stray reference to idle_list in
+ cleanup_thread_list(). This may be the reason for the note in
+ iax2_process_thread() about threads not being removed from the
+ task lists. * Move cleanup_thread_list(&idle_list) to after the
+ other lists are cleaned up. ........ Merged revisions 398416 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_iax2.c, /: chan_iax2: Fix bridgecallno deadlock
+ avoidance. * Fix bridgecallno deadlock avoidance. When doing
+ deadlock avoidance, you need to retest the status of values for
+ each loop to see if you still need the lock for bridgecallno. *
+ As a safety check, after acquiring the bridgecallno lock you
+ should check if iaxs[bridgecallno] is NULL just like the current
+ callno checks. * Move setting thread->iostate to IAX_IOSTATE_IDLE
+ to after processing any deferred frames to ensure that the
+ iostate is IDLE when it is placed back into the idle list.
+ defer_full_frame() tries to ensure iax2_process_thread() wakes up
+ to process the frame. ........ Merged revisions 398379 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/iax2-parser.c: chan_iax2: Add missing control frame
+ names to debug frame decode output. (Part 2)
+
+ * channels/iax2-parser.c, /: chan_iax2: Add missing control frame
+ names to debug frame decode output. ........ Merged revisions
+ 398301 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-04 21:33 +0000 [r398281-398285] Jonathan Rose <jrose@digium.com>
+
+ * tests/test_voicemail_api.c: unit tests: test_voicemail_api leaks
+ stringfields from snapshots (closes issue ASTERISK-22414)
+ Reported by: Corey Farrell Patches:
+ test_voicemail_api-leaks-11.patch uploaded by coreyfarrell
+ (license 5909)
+
+ * apps/app_voicemail.c: app_voicemail: Fix leaking config objects
+ when msg_id doesn't match (issues ASTERISK-22414) Reported by:
+ Corey Farrell Patch: test_voicemail_api-leaks-11.patch uploaded
+ by coreyfarrell (license 5909)
+
+2013-09-04 15:57 +0000 [r398236] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: chan_misdn: Fix misdn debug output
+ printed with arbitrary verbose levels. Fix the misdn debug output
+ to remote consoles. chan_misdn uses ast_console_puts() which
+ doesn't know about verbose levels. Better to use ast_verbose()
+ instead. Without this patch the misdn debug messages are appended
+ to the verbose level which ever was set by the message sent to
+ the console before, i.e. any undefined level. (closes issue
+ AST-1218) Reported by: Guenther Kelleter Patches: misdnlog.patch
+ (license #6372) patch uploaded by Guenther Kelleter ........
+ Merged revisions 398235 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-03 19:45 +0000 [r398214] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh245.c: Fix remote tcs sequence handling on
+ empty tcs received
+
+2013-09-02 07:28 +0000 [r398168] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, cel/cel_custom.c: Be a little more verbose when loading
+ cel_custom.conf. Review: https://reviewboard.asterisk.org/r/2805/
+ ........ Merged revisions 398167 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-30 19:16 +0000 [r398022-398103] Kevin Harwell <kharwell@digium.com>
+
+ * main/indications.c, main/config.c, res/res_security_log.c, /,
+ channels/chan_sip.c, main/translate.c, main/named_acl.c: Fix
+ various memory leaks main/config.c - cleanup cache fie includes
+ res/res_security_log.c - unregister logger level
+ channesl/chan_sip.c - cleanup io context and notify_types
+ main/translator.c - cleanup at shutdown main/named_acl.c -
+ cleanup cli commands main/indications.c -
+ ast_get_indication_tone() unref default_tone_zone if used (closes
+ issues ASTERISK-22378) Reported by: Corey Farrell Patches:
+ config_shutdown.patch uploaded by coreyfarrell (license 5909)
+ res_security_log.patch uploaded by coreyfarrell (license 5909)
+ chan_sip-11.patch uploaded by coreyfarrell (license 5909)
+ indications_refleak.patch uploaded by coreyfarrell (license 5909)
+ named_acl-cli_unreg-11.patch uploaded by coreyfarrell (license
+ 5909) translate_shutdown.patch uploaded by coreyfarrell (license
+ 5909) ........ Merged revisions 398102 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * res/res_agi.c, main/manager.c, /: Memory leak fix
+ ast_xmldoc_printable returns an allocated block that must be
+ freed by the caller. Fixed manager.c and res_agi.c to stop
+ leaking these results. (closes issue ASTERISK-22395) Reported by:
+ Corey Farrell Patches: manager-leaks-11.patch uploaded by
+ coreyfarrell (license 5909) res_agi-xmldoc-leaks.patch uploaded
+ by coreyfarrell (license 5909) ........ Merged revisions 398060
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/features.c: Fix memory leak Fixed a features.c test that
+ leaked a reference to a parked call. This caused chancount to
+ never reach 0, so graceful shutdown stops. Also added an
+ unregister test. (closes issue ASTERISK-22413) Reported by: Corey
+ Farrell Patches: features-TEST_FRAMEWORK.patch uploaded by
+ coreyfarrell (license 5909) ........ Merged revisions 398021 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-30 16:57 +0000 [r398019] Richard Mudgett <rmudgett@digium.com>
+
+ * tests/test_substitution.c, /: test_substituition: Fix failed test
+ reporting to actually report failure. You cannot put the "Testing
+ <blah> pass/fail" on a single line before actually performing the
+ test. Now any additional failure information is logged before the
+ test pass/fail announcement. * Added an additional CDR(answer,u)
+ test. ........ Merged revisions 398018 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-30 16:20 +0000 [r397948-398011] Kevin Harwell <kharwell@digium.com>
+
+ * /, apps/app_mixmonitor.c: Fix memory leaks (closes issue
+ ASTERISK-22368) Reported by: Corey Farrell Patches:
+ issueA22368_mixmonitor_free_filename.patch uploaded by wdoekes
+ (license 5674) ........ Merged revisions 398004 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/asterisk.c: Check return value on fwrite
+
+ * channels/chan_misdn.c, apps/app_dumpchan.c, main/features.c,
+ main/logger.c, apps/app_verbose.c, main/asterisk.c: Verbose
+ logging discrepancies Refactored cases where a combination of
+ ast_verbose/options_verbose were present. Also in general tried
+ to eliminate, in as many places as possible, where the
+ options_verbose global variable was being used. Refactored the
+ way local and remote consoles handle verbose message logging in
+ an attempt to solve the various discrepancies that sometimes
+ would show between the two. (closes issue AST-1193) Reported by:
+ Guenther Kelleter Review:
+ https://reviewboard.asterisk.org/r/2798/
+
+2013-08-27 18:03 +0000 [r397758] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: AST-2013-005: Fix crash caused by invalid
+ SDP If the SIP channel driver processes an invalid SDP that
+ defines media descriptions before connection information, it may
+ attempt to reference the socket address information even though
+ that information has not yet been set. This will cause a crash.
+ This patch adds checks when handling the various media
+ descriptions that ensures the media descriptions are handled only
+ if we have connection information suitable for that media. Thanks
+ to Walter Doekes, OSSO B.V., for reporting, testing, and
+ providing the solution to this problem. (closes issue
+ ASTERISK-22007) Reported by: wdoekes Tested by: wdoekes patches:
+ issueA22007_sdp_without_c_death.patch uploaded by wdoekes
+ (License 5674) ........ Merged revisions 397756 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 397757 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2013-08-27 16:40 +0000 [r397744] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c, channels/chan_motif.c, channels/chan_iax2.c,
+ channels/sig_pri.c, channels/sig_ss7.c, channels/chan_dahdi.c,
+ channels/sig_analog.c: Fix uninitialized value in struct
+ ast_control_pvt_cause_code usage.
+
+2013-08-27 15:55 +0000 [r397712] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: AST-2013-004: Fix crash when handling ACK
+ on dialog that has no channel A remote exploitable crash
+ vulnerability exists in the SIP channel driver if an ACK with SDP
+ is received after the channel has been terminated. The handling
+ code incorrectly assumed that the channel would always be
+ present. This patch adds a check such that the SDP will only be
+ parsed and applied if Asterisk has a channel present that is
+ associated with the dialog. Note that the patch being applied was
+ modified only slightly from the patch provided by Walter Doekes
+ of OSSO B.V. (closes issue ASTERISK-21064) Reported by: Colin
+ Cuthbertson Tested by: wdoekes, Colin Cutherbertson patches:
+ issueA21064_fix.patch uploaded by wdoekes (License 5674) ........
+ Merged revisions 397710 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 397711 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2013-08-23 21:57 +0000 [r397604] Joshua Colp <jcolp@digium.com>
+
+ * res/res_rtp_asterisk.c, UPGRADE.txt, res/Makefile: Make libuuid
+ an optional dependency for res_rtp_asterisk instead of a
+ requirement. Review: https://reviewboard.asterisk.org/r/2777/
+
+2013-08-23 16:07 +0000 [r397528] Richard Mudgett <rmudgett@digium.com>
+
+ * main/utils.c, include/asterisk/lock.h, main/astmm.c,
+ channels/sig_pri.c, main/astobj2.c, include/asterisk/logger.h,
+ main/lock.c, include/asterisk/utils.h, include/asterisk/astmm.h,
+ /, main/logger.c: Fix memory corruption when trying to get "core
+ show locks". Review https://reviewboard.asterisk.org/r/2580/
+ tried to fix the mismatch in memory pools but had a math error
+ determining the buffer size and didn't address other similar
+ memory pool mismatches. * Effectively reverted the previous patch
+ to go in the same direction as trunk for the returned memory pool
+ of ast_bt_get_symbols(). * Fixed memory leak in
+ ast_bt_get_symbols() when BETTER_BACKTRACES is defined. * Fixed
+ some formatting in ast_bt_get_symbols(). * Fixed sig_pri.c
+ freeing memory allocated by libpri when MALLOC_DEBUG is enabled.
+ * Fixed __dump_backtrace() freeing memory from
+ ast_bt_get_symbols() when MALLOC_DEBUG is enabled. * Moved
+ __dump_backtrace() because of compile issues with the utils
+ directory. (closes issue ASTERISK-22221) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/2778/ ........ Merged
+ revisions 397525 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-22 08:22 +0000 [r397378] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * default.exports, /, main/asterisk.exports.in: Add _IO_stdin_used
+ in version-script to fix SIGBUSes on Sparc. The
+ --version-script,asterisk.exports linker flag (and the module
+ exports) didn't provide _IO_stdin_used in the list of exported
+ symbols. That causes some kind of libc compatibility mode to kick
+ in, where stdio file structures (stdout/stderr) land somewhere
+ else. In the case of the Sparc, they landed on misaligned memory.
+ This became apparent first after r376428 (Reorder startup
+ sequence) when a lot of ast_log's were replaced with fprintf's.
+ Writing to stderr triggered a SIGBUS. (Compared to x86 and amd64
+ architectures, the Sparc is very picky about memory alignment.)
+ (issue ASTERISK-21763) (issue ASTERISK-21665) Reported by: Jeremy
+ Kister Review: https://reviewboard.asterisk.org/r/2760/ ........
+ Merged revisions 397377 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-21 23:02 +0000 [r397365] Jonathan Rose <jrose@digium.com>
+
+ * main/udptl.c: UDPTL: Fix a regression where UDPTL won't load
+ default settings If the file udptl.conf is unavailable at
+ startup, UDPTL will fail to initialize and while it makes some
+ noise, it isn't immediately obvious why consumers start to fail
+ when using it. This patch makes UDPTL load as though an empty
+ config was provided when udptl is unavailable at startup. (closes
+ issue ASTERISK-22349) Reported by: Jonathan Rose Review:
+ https://reviewboard.asterisk.org/r/2773/
+
+2013-08-21 17:07 +0000 [r397309] David M. Lee <dlee@digium.com>
+
+ * /, main/http.c: Complete http_shutdown. This patch frees up some
+ resources allocated in http.c. * tcp listeners stopped * tls
+ settings freed * uri redirects freed * unregister internal http.c
+ uri's (closes issue ASTERISK-22237) Reported by: Corey Farrell
+ Patches: http.patch uploaded by Corey Farrell (license 5909)
+ ........ Merged revisions 397308 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-21 15:12 +0000 [r397257] Matthew Jordan <mjordan@digium.com>
+
+ * /, include/asterisk/frame.h: Set 14400 as the default max bit
+ rate if T38MaxBitRate is not specified If an endpoint fails to
+ include the T38MaxBitRate attribute during negotiation, Asterisk
+ will negotiate a bit rate of 2400 instead of the ITU recommended
+ bit rate of 14400. This patch fixes this by making
+ AST_T38_RATE_14400 the 'default' value of the enum by assigning
+ it a value of 0, such that if an endpoint fails to include the
+ attribute, the default will be 14400. Note that Walter Doekes
+ included the nice comment in frame.h about why we are
+ purposefully assigning AST_T38_RATE_14400 a value of 0. (closes
+ issue ASTERISK-22275) Reported by: Andreas Steinmetz patches:
+ fax-fix.patch uploaded by anstein (License 6523) ........ Merged
+ revisions 397256 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-21 14:36 +0000 [r397254] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Prevent a crash on outbound SIP MESSAGE
+ requests. If a From header on an outbound out-of-call SIP MESSAGE
+ were malformed, the result could crash Asterisk. In addition, if
+ a From header on an incoming out-of-call SIP MESSAGE request were
+ malformed, the message was happily accepted rather than being
+ rejected up front. The incoming message path would not result in
+ a crash, but the behavior was bad nonetheless. (closes issue
+ ASTERISK-22185) reported by Zhang Lei
+
+2013-08-21 02:11 +0000 [r397205] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, channels/chan_sip.c: Fix Not Storing Current Incoming Recv
+ Address In 1.8, r384779 introduced a regression by retrieving an
+ old dialog and keeping the old recv address since recv was
+ already set. This has caused a problem when a proxy is involved
+ since responses to incoming requests from the proxy server, after
+ an outbound call is established, are never sent to the correct
+ recv address. In 11, r382322 introduced this regression. The fix
+ is to revert that change and always store the recv address on
+ incoming requests. Thank you Walter Doekes for helping to point
+ out this error and Mark Michelson for your input/review of the
+ fix. (closes issue ASTERISK-22071) Reported by: Alex Zarubin
+ Tested by: Alex Zarubin, Karsten Wemheuer Patches:
+ asterisk-22071-store-recvd-address.diff by Michael L. Young
+ (license 5026) ........ Merged revisions 397204 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-20 17:41 +0000 [r397133-397157] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Remove REF_DEBUG definition. ........
+ Merged revisions 397156 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c, channels/sip/dialplan_functions.c: Fix
+ refcounting of sip_pvt in test_sip_rtpqos test and unlink it from
+ the list of pvts. (closes issue ASTERISK-22248) reported by Corey
+ Farrell patches: test_sip_rtpqos.patch uploaded by Corey Farrell
+ (license #5909) ........ Merged revisions 397112 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-20 15:27 +0000 [r397034-397107] Kinsey Moore <kmoore@digium.com>
+
+ * /, main/threadstorage.c, main/astfd.c: Unregister CLI commands on
+ exit This patch ensures that CLI commands enabled by
+ DEBUG_FD_LEAKS and DEBUG_THREADLOCALS are cleaned up properly on
+ exit. (closes issue ASTERISK-22238) Reported by: Corey Farrell
+ Tested by: Corey Farrell Patches: debug_cli_unregister.patch
+ uploaded by Corey Farrell ........ Merged revisions 397106 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/xmldoc.c, /: Fix xmldoc memory leak This fixes a
+ single-attribute memory leak that was occurring when the
+ "required" attribute was not true. (closes issue ASTERISK-22249)
+ Reported by: Corey Farrell Tested by: Corey Farrell Patches:
+ xmldoc-free_attr_required.patch uploaded by Corey Farrell
+ ........ Merged revisions 397064 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/cel.c, /: Protect CEL from an invalid config on reload This
+ patch fixes CEL to properly handle an invalid config on reload.
+ (closes issue ASTERISK-22259) Reported by: Corey Farrell Tested
+ by: Corey Farrell Patches: cel-config.patch uploaded by Corey
+ Farrell ........ Merged revisions 397033 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-20 11:47 +0000 [r396995] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * configs/h323.conf.sample, /, configs/sip.conf.sample: Add
+ "autoframing" option to sip.conf.sample and h323.conf.sample. The
+ autoframing option was added to chan_sip.c in r43243 (mogorman,
+ 2006-09-19 01:32:57), but never made its way into the sample
+ configs. Review: https://reviewboard.asterisk.org/r/2768/
+ ........ Merged revisions 396994 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-20 01:18 +0000 [r396944-396961] Matthew Jordan <mjordan@digium.com>
+
+ * main/data.c, /: Fix invalid access to disposed memory in
+ main/data unit test It is not safe to iterate over a macro'd list
+ of ao2 objects, deref them such that the item's destructor is
+ called, and leave them in the list. The list macro to iterate
+ over items requires the item to be a valid allocated object in
+ order to proceed to the next item; with MALLOC_DEBUG on the
+ corruption of the linked list is caught in the crash. This patch
+ fixes the invalid access to free'd memory by removing the ao2
+ item from the list before de-refing it. Note that this is a
+ backport of r396915 from Asterisk trunk. ........ Merged
+ revisions 396958 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_queue.c: Let Queue wrap up time influence member
+ availability Queue members who happen to be in multiple queues at
+ the same time may not have any wrap up time. This problem
+ occurred due to a code change in Asterisk 11.3.0 that unified
+ device state tracking of Queue members in multiple Queues (which
+ fixed some other problems, but unfortunately caused this one).
+ This patch fixes the behavior by having the is_member_available
+ function check the queue's wrap up time and the time of the
+ member's last call, such that for a particular queue, the member
+ won't be considered available if their last call is within the
+ wrap up time. (closes issue ASTERISK-22189) Reported by: Tony
+ Lewis Tested by: Tony Lewis
+
+ * apps/app_meetme.c: Resolve conflicts between
+ CONFFLAG_DONT_DENOISE and CONFFLAG_INTROUSER_VMREC When r382230
+ added an option to not denoise the MeetMe conference (if a user
+ had a channel whose format's sample rate changed frequently, for
+ example), the value added was the maximum allowed value for the
+ constants that define the options for MeetMe in 1.8. Not so in 11
+ - unfortunately, the option CONFFLAG_DONT_DENOISE conflicts with
+ CONFFLAG_INTROUESR_VMREC. This patch fixes that, and also tweaks
+ one of the way in which the constants was declared for
+ consistency. Thanks to Tony Mountifield for pointing out the
+ problem and solution. (closes issue ASTERISK-22269) Reported by:
+ Tony Mountifield
+
+2013-08-16 22:45 +0000 [r396884] John Bigelow <jbigelow@digium.com>
+
+ * main/features.c: Add test suite events to indicate when a feature
+ is detected or not These are needed by the bridge test suite
+ tests for them to be able to run against Asterisk 11. Review:
+ https://reviewboard.asterisk.org/r/2751/
+
+2013-08-15 16:29 +0000 [r396746] Kinsey Moore <kmoore@digium.com>
+
+ * main/asterisk.c, main/cli.c, /: Remove leading spaces from the
+ CLI command before parsing If you've mistakenly put a space
+ before typing in a command, the leading space will be included as
+ part of the command, and the command parser will not find the
+ corresponding command. This patch rectifies that situation by
+ stripping the leading spaces on commands. Review:
+ https://reviewboard.asterisk.org/r/2709/ Patch-by: Tilghman
+ Lesher ........ Merged revisions 396745 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-14 19:06 +0000 [r396620-396657] Joshua Colp <jcolp@digium.com>
+
+ * tests/test_hashtab_thrash.c, /: Tweak comment for why usleep is
+ used. ........ Merged revisions 396656 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, tests/test_hashtab_thrash.c: Tweak test_hashtab_thrash test to
+ allow the critical threads to execute. Depending on certain
+ conditions it was possible for the hashtab counting thread to
+ starve other threads, preventing them from executing in the
+ expected fashion. This change adds a sleep to allow the others to
+ do what they need to do. While this doesn't thrash the hashtab as
+ much as previously, it at least works. (closes issue
+ ASTERISK-22276) Reported by: Matt Jordan ........ Merged
+ revisions 396619 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-13 18:45 +0000 [r396580-396583] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, channels/chan_sip.c: chan_sip: Convert 'just did sched_add
+ waitid...' from warning to debug message. Patches:
+ reviewboard-2377.patch uploaded by Paul Belanger Review:
+ https://reviewboard.asterisk.org/r/2377/ ........ Merged
+ revisions 396582 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: chan_sip: Fix IP-addr in warning when
+ rejecting a contact ACL. Patches: reviewboard-2155.patch uploaded
+ by Paul Belanger Review: https://reviewboard.asterisk.org/r/2155/
+ ........ Merged revisions 396579 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-08 20:21 +0000 [r396441] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * include/asterisk/logger.h, /, main/logger.c, main/utils.c,
+ main/astobj2.c: Consistent memory allocation by
+ ast_bt_get_symbols. Always use ast_alloc/ast_free. This is
+ handled differently in trunk (r391012). Review:
+ https://reviewboard.asterisk.org/r/2580/ ........ Merged
+ revisions 396427 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-08 07:03 +0000 [r396377] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * channels/chan_unistim.c: - Fix different issues with call
+ transfer cancel. In case 3rd party busy or congestion call was
+ not returned. - Fix displaying soft button 'Redial' in case of no
+ redial number exists
+
+2013-08-06 08:37 +0000 [r396287-396310] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * funcs/func_strings.c: Check result of ast_var_assign() calls for
+ memory allocation failure (2). Missed a spot in the previous
+ commit.
+
+ * apps/app_stack.c, apps/app_playback.c, funcs/func_global.c,
+ main/cdr.c, pbx/pbx_loopback.c, main/pbx.c, /,
+ funcs/func_strings.c, pbx/pbx_dundi.c, utils/extconf.c: Check
+ result of ast_var_assign() calls for memory allocation failure.
+ We try to keep the system running even when all available memory
+ is spent. Review: https://reviewboard.asterisk.org/r/2734/
+ ........ Merged revisions 396279 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-05 20:19 +0000 [r396197-396248] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, channels/chan_sip.c: Fix Registration Failure When A Peer And
+ TLS Are Used If a peer is used in a register line and TLS is
+ defined as the transport, the registration fails since the
+ transport on the dialog is never set properly resulting in UDP
+ being used instead of TLS. This patch sets the dialog's transport
+ based on the transport that was defined in the register line. If
+ the register line does not specify a transport, the parsing
+ function for the register line always defaults back to UDP.
+ (closes issue ASTERISK-21964) Reported by: Doug Bailey Tested by:
+ Doug Bailey Patches: asterisk-21964-set-reg-dialog-transport.diff
+ by Michael L. Young (license 5026) ........ Merged revisions
+ 396240 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * UPGRADE.txt: Change "from" to "From". (related to issue
+ ASTERISK-21903)
+
+ * /, UPGRADE.txt: Adding a note to UPGRADE.txt about a change made
+ to res_agi in order to indicate when streaming an audio file
+ fails like it is done in other parts of the code to indicate an
+ error. Note was requested by Paul Belanger:
+ http://lists.digium.com/pipermail/asterisk-dev/2013-July/061420.html
+ (related to issue ASTERISK-21903) ........ Merged revisions
+ 396196 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-07-22 13:50 +0000 [r394890-395033] Matthew Jordan <mjordan@digium.com>
+
+ * main/asterisk.c, /: Update copyright year to 2013 in asterisk.c;
+ some whitespace fixes (closes issue ASTERISK-22179) Reported by:
+ Malcolm Davenport ........ Merged revisions 395032 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * CHANGES, UPGRADE.txt: Add an upgrade note for libuuid dependency;
+ remove note in CHANGES This patch notes that libuuid is now a
+ dependency for res_rtp_asterisk; this was introduced in between
+ 11.4.0 and 11.5.0 to resolve a dependency for pjproject, which
+ res_rtp_asterisk uses for ICE/STUN/TURN support. It also removes
+ a conflicting note from CHANGES. While support for playing
+ prompts to the first participant was added for app_queue, it was
+ disabled by default and an option added to enable it. That was
+ properly noted in the UPGRADE.txt file.
+
+ * /, funcs/func_channel.c: Clean up documentation This patch cleans
+ up documentation in func_channel for the following items: *
+ rtpsource * secure_signaling * secure_media * various OOH323
+ parameters (closes issue ASTERISK-20969) Reported by: snuffy
+ patches: func_chan-update.diff uploaded by snuffy (License 5024)
+ ........ Merged revisions 394980 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, configs/indications.conf.sample: Provide proper ring tone in
+ indications.conf for Malaysia The ring tone provided in the
+ sample indications.conf was incorrect. This patch modifies the
+ sample ring tone to be what it should: ring =
+ 425/400,0/200,425/400,0/2000 This brings it in line with the tone
+ definition in DAHDI 2.7.0. (zonedata.c) (closes issue
+ ASTERISK-21997) Reported by: Filip Jenicek patches:
+ malaysia_ring.patch uploaded by phill (License 6277) ........
+ Merged revisions 394940 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/http.c, /: Tolerate presence of RFC2965 Cookie2 header by
+ ignoring it This patch modifies parsing of cookies in Asterisk's
+ http server by doing an explicit comparison of the "Cookie"
+ header instead of looking at the first 6 characters to determine
+ if the header is a cookie header. This avoids parsing "Cookie2"
+ headers and overwriting the previously parsed "Cookie" header.
+ Note that we probably should be appending the cookies in each
+ "Cookie" header to the parsed results; however, while clients can
+ send multiple cookie headers they never really do. While this
+ patch doesn't improve Asterisk's behavior in that regard, it
+ shouldn't make it any worse either. Note that the solution in
+ this patch was pointed out on the issue by the issue reporter,
+ Stuart Henderson. (closes issue ASTERISK-21789) Reported by:
+ Stuart Henderson Tested by: mjordan, Stuart Henderson ........
+ Merged revisions 394899 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * contrib/realtime/postgresql/realtime.sql, /: Update PostgreSQL
+ realtime scripts with schema for queue_log table This patch
+ updates the realtime SQL scripts with an entry that will create
+ the queue_log table. This brings the PostgreSQL scripts inline
+ with the MySQL scripts, with respect to what tables they will
+ create. (closes issue ASTERISK-21021) Reported by: Eugene
+ patches: queue_log.sql uploaded by varnav (license 6360) ........
+ Merged revisions 394896 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * configs/iax.conf.sample, /: Document connectedline parameter for
+ chan_iax2 The connectedline parameter for a chan_iax2 peer was
+ undocumented. This patch documents the options in the sample
+ configuration file. (closes issue ASTERISK-21953) Reported by:
+ Birger "WIMPy" Harzenetter ........ Merged revisions 394886 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-07-18 12:52 +0000 [r394641] Michael L. Young <elgueromexicano@gmail.com>
+
+ * res/res_agi.c, /: Properly indicate failure to open an audio
+ stream in res_agi If there is an error streaming an audio file,
+ the current return status makes it difficult for an AGI script to
+ determine that there was an error with the audio file. This
+ patches changes the result to return -1 and the function returns
+ RESULT_FAILURE instead of RESULT_SUCCESS. From looking at other
+ parts of res_agi, this would appear to be the proper way to
+ handle an error. (closes issue ASTERISK-21903) Reported by: Ariel
+ Wainer Tested by: Ariel Wainer Patches:
+ asterisk-21903-return-stream-res_1.8.diff by Michael L. Young
+ (license 5026) Review: https://reviewboard.asterisk.org/r/2625/
+ ........ Merged revisions 394640 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-07-14 02:34 +0000 [r394303-394345] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_queue.c: Provide error message for QUEUE_MEMBER when
+ member is not in queue When QUEUE_MEMBER is used and the member
+ specified is not in the queue, Asterisk provides an ERROR message
+ that indicates that the option specified is not valid. This patch
+ now properly displays an ERROR message that the member is not in
+ the queue if an interface is specified. (closes issue
+ ASTERISK-21980) Reported by: Avraam David
+
+ * /, funcs/func_strings.c: Clarify documentation for function
+ PASSTHRU It is not apparent to the average user that the PASSTHRU
+ function should not be passed as ${PASSTHRU(string)} but just as
+ PASSTHRU(string) to functions which take a variable name and not
+ its contents. This patch clarifies the behavior in the
+ documentation and provides an example. (closes issue
+ ASTERISK-21717) Reported by: Richard Miller patches:
+ func_strings.diff uploaded by Richard Miller (license 5685)
+ ........ Merged revisions 394302 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-07-11 21:28 +0000 [r394173] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, /: Fix a longstanding issue with MFC-R2
+ configuration that prevented users from mixing different variants
+ or general MFC-R2 settings within the same E1 line. Most users do
+ not have a problem with this since MFC-R2 lines are usually
+ fractional E1s, or the whole E1 has the same country variant and
+ R2 settings. In Venezuela however is common to have inbound
+ MFC-R2 and outbound DTMF-R2 within the same E1. This fix now
+ properly parses the chan_dahdi.conf file to generate a new openr2
+ context every time a new channel => section is found and the
+ configuration was changed. (closes issue ASTERISK-21117) Reported
+ by: Rafael Angulo Related Elastix issue:
+ http://bugs.elastix.org/view.php?id=1612 ........ Merged
+ revisions 394106 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-07-10 01:49 +0000 [r393929] Russell Bryant <russell@russellbryant.com>
+
+ * configs/sla.conf.sample, /, apps/app_meetme.c: astobj2-ify the
+ SLA code The SLA code within app_meetme was written before
+ asotbj2 had been merged into Asterisk. Worse, support for reloads
+ did not exist at first and was added later as a bolt-on feature.
+ I knew at the time that reloading was not safe at all while SLA
+ was in use, so the reload would be queued up to execute when the
+ system was idle. Unfortunately, this approach was still prone to
+ errors beyond the fact that this was the only place in Asterisk
+ where configuration was not reloaded instantly when requested.
+ This patch converts various SLA objects to be reference counted
+ objects using astobj2. This allows reloads to be processed while
+ the system is in use. The code ensures that the objects will not
+ disappear while one of the other threads is using them. However,
+ they will be immediately removed from the global trunk and
+ station containers so no new calls will use them if removed from
+ configuration. Review: https://reviewboard.asterisk.org/r/2581/
+ ........ Merged revisions 393928 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-07-03 23:52 +0000 [r393628-393630] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_mixmonitor.c: MixMonitor: Fix refleak in
+ manager_stop_mixmonitor() if could not stop monitoring. ........
+ Merged revisions 393490 from
+ http://svn.asterisk.org/svn/asterisk/trunk
+
+ * channels/chan_dahdi.c, /: chan_dahdi: Fix segfault reloading
+ chan_dahdi when round robin is used. * Clear round_robin[] in
+ dahdi_restart(). (closes issue ASTERISK-21847) Reported by: Ivo
+ Andonov Patches: jira_asterisk_21847_v1.8.patch (license #5621)
+ patch uploaded by rmudgett ........ Merged revisions 393627 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-07-02 10:14 +0000 [r393395] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * channels/chan_unistim.c: Fix issue with inability to cancell call
+ transfer made by on-sceen menus. Reported by: Igor Olhovskiy
+
+2013-06-25 01:07 +0000 [r392810] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_motif.c, main/http.c, main/config_options.c,
+ main/named_acl.c, res/res_calendar.c: Fix memory/ref counting
+ leaks in a variety of locations This patch fixes the following
+ memory leaks: * http.c: The structure containing the addresses to
+ bind to was not being deallocated when no longer used *
+ named_acl.c: The global configuration information was not
+ disposed of * config_options.c: An invalid read was occurring for
+ certain option types. * res_calendar.c: The loaded calendars on
+ module unload were not being properly disposed of. *
+ chan_motif.c: The format capabilities needed to be disposed of on
+ module unload. In addition, this now specifies the default
+ options for the maxpayloads and maxicecandidates in such a way
+ that it doesn't cause the invalid read in config_options.c to
+ occur. (issue ASTERISK-21906) Reported by: John Hardin patches:
+ http.patch uploaded by jhardin (license 6512) named_acl.patch
+ uploaded by jhardin (license 6512) config_options.patch uploaded
+ by jhardin (license 6512) res_calendar.patch uploaded by jhardin
+ (license 6512) chan_motif.patch uploaded by jhardin (license
+ 6512)
+
+2013-06-14 16:21 +0000 [r391794] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_mixmonitor.c, /: app_mixmonitor: Fix crashes caused by
+ unloading app_mixmonitor Unloading app_mixmonitor while active
+ mixmonitors were running would cause a segfault. This patch fixes
+ that by making it impossible to unload app_mixmonitor while
+ mixmonitors are active. Review:
+ https://reviewboard.asterisk.org/r/2624/ ........ Merged
+ revisions 391778 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-06-13 18:47 +0000 [r391700] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/confbridge/conf_config_parser.c,
+ apps/confbridge/include/confbridge.h, apps/app_confbridge.c:
+ app_confbridge: Fix memory leak on reload. The config framework
+ options should not be registered multiple times. Instead the
+ configuration just needs to be reprocessed by the config
+ framework.
+
+2013-06-12 21:00 +0000 [r391560] David M. Lee <dlee@digium.com>
+
+ * res/res_http_websocket.c: Fix segfault for certain invalid
+ WebSocket input. The WebSocket code would allocate, on the stack,
+ a string large enough to hold a key provided by the client, and
+ the WEBSOCKET_GUID. If the key is NULL, this causes a segfault.
+ If the key is too large, it could overflow the stack. This patch
+ checks the key for NULL and checks the length of the key to avoid
+ stack smashing nastiness. (closes issue ASTERISK-21825) Reported
+ by: Alfred Farrugia Tested by: Alfred Farrugia, David M. Lee
+ Patches: issueA21825_check_if_key_is_sent.patch uploaded by
+ Walter Doekes (license 5674)
+
+2013-06-12 02:25 +0000 [r391507] Matthew Jordan <mjordan@digium.com>
+
+ * main/loader.c, main/format.c, /: Fix memory leak while loading
+ priority modules and adding formats This patch fixes two memory
+ leaks: * When we load a module with the LOAD_PRIORITY flag, we
+ remove its entry from the load order list. Unfortunately, we
+ don't free the memory associated with entry in the list. This
+ patch corrects that and properly frees the memory for the module
+ in the list. * When adding a custom format (such as SILK or
+ CELT), the routine for adding the format was leaking a reference.
+ RAII_VAR cleans this up properly. ........ Merged revisions
+ 391489 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-06-11 10:22 +0000 [r391379] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * channels/chan_unistim.c: Fix issue with no sound in both way in
+ case of previous call to chan_unistim phone was canceled.
+ (related to ASTERISK-20183)
+
+2013-06-11 08:10 +0000 [r391334] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/chan_iax2.c, /: IAX2: Transfer Reject: Lock bridgecallno
+ before touching it, refactor 1). When touching the bridgecallno,
+ we need to lock it. 2). Remove magic number '0' and replace with
+ TRANSFER_NONE. 3). Exit early if no bridgecallno. 4). Reduce
+ indentation. Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2613/ ........ Merged
+ revisions 391333 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-07-15 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 11.5.0 Released.
+
+2013-07-12 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 11.5.0-rc2 Released.
+
+ * Properly lock and safely handle a transfer failure in IAX2
+
+ When touching the bridgecallno, we need to lock it - otherwise a
+ race condition can occur. This patch does the proper locking
+ of the bridgecallno before modifying its state.
+
+2013-06-10 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 11.5.0-rc1 Released.
+
+2013-06-10 14:25 +0000 [r391241] Matthew Jordan <mjordan@digium.com>
+
+ * /, configs/queues.conf.sample, UPGRADE.txt, apps/app_queue.c: Add
+ announce-to-first-user option for app_queue In r386792, the
+ ability to play prompts to the first caller in a call queue was
+ added. While this is arguably a bug fix for those who expect the
+ first caller to continue receiving prompts while the agent is
+ dialed, it has the side effect of preventing the first caller
+ from hearing the agent immediately upon bridging. This may not be
+ a problem for those who really want this option, but for those
+ who didn't care whether or not the first caller in queue heard
+ their position, it was an issue. This patch disables the ability
+ for the first caller in the queue to hear prompts and adds a new
+ option, announce-to-first-user, to queues.conf. Those who the
+ behavior can enable it by setting this value to True. Note that
+ if we ever implement the ability to have the prompts be stopped
+ upon bridging, this option can be removed. (closes issue
+ ASTERISK-21782) Reported by: Remi Quezada ........ Merged
+ revisions 391215 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-06-10 09:32 +0000 [r391063-391148] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, channels/chan_iax2.c: chan_iax2: nativebridge refactor, missed
+ unlock bridgecallno ........ Merged revisions 391143 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_iax2.c: fix bad edit after conflict resolution
+ ........ Merged revisions 391107 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_iax2.c: IAX2: refactor nativebridge transfer
+ remove triple checking of iaxs[fr->callno]->transferring reduce
+ indentation. Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2602/ ........ Merged
+ revisions 391065 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_iax2.c: IAX2: fix race condition with
+ nativebridge transfers. 1). When touching the bridgecallno, we
+ need to lock it. 2). stop_stuff() which calls
+ iax2_destroy_helper() Assumes the lock on the pvt is already
+ held, when iax2_destroy_helper() is called. Thus we need to lock
+ the bridgecallno pvt before we call
+ stop_stuff(iaxs[fr->callno]->bridgecallno); 3). When evaluating
+ the state of 'callno->transferring' of the current leg, we can't
+ change it to READY unless the bridgecallno is locked. Why, if we
+ are interrupted by the other call leg before 'transferring =
+ TRANSFER_RELEASED', the interrupt will find that it is READY and
+ that the bridgecallno is also READY so Releases the legs. (closes
+ issue ASTERISK-21409) Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2594/ ........ Merged
+ revisions 391062 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-31 10:34 +0000 [r390228-390229] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c: remove unnecessary declarations (issue
+ ASTERISK-21800)
+
+ * addons/chan_ooh323.c, /: reject call attempts when gatekeeper is
+ configured but not registered (closes issue ASTERISK-21800)
+ Reported by: Dmitry Melekhov Patches: ASTERISK-21800-1.patch
+ Tested by: Dmitry Melekhov ........ Merged revisions 390181 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 390223 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2013-05-29 20:18 +0000 [r390047] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /: Fix segfault when dealing with chan_agent
+ channels. Check the returned bridged pointer for NULL to avoid a
+ crash. It looks like chan_agent is returning a NULL pointer when
+ it probably should be returning a pointer to the channel the
+ Agent channel is pretending to be. (closes issue ASTERISK-21793)
+ Reported by: Rodrigo P. Telles Patches:
+ jira_asterisk_21793_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: Rodrigo P. Telles ........ Merged revisions
+ 390044 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-28 17:43 +0000 [r389896] Jonathan Rose <jrose@digium.com>
+
+ * /, main/slinfactory.c: Fix a memory copying bug in slinfactory
+ which was causing mixmonitor issues. Reported by: Michael Walton
+ Tested by: Jonathan Rose Patches:
+ slinfactory.c.ASTERISK-21799.patch uploaded by Michael Walton
+ (license 6502) (closes issue ASTERISK-21799) ........ Merged
+ revisions 389895 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-24 11:49 +0000 [r389677] Matthew Jordan <mjordan@digium.com>
+
+ * /, main/logger.c: Print all logger messages on shutdown When
+ Asterisk shuts down and shuts down the loggin gsubsystem, any
+ messages currently in flight will not get logged. This patch
+ prevents the loop writing messages from breaking out prematurely,
+ such that all of the messages are logged. (closes issue
+ ASTERISK-21716) Reported by: Corey Farrell patches:
+ logger-process-all-messages.patch uploaded by Corey Farrell
+ (license 5909) ........ Merged revisions 389676 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-24 10:12 +0000 [r389661] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * channels/chan_unistim.c: Fix several problems caused by multiple
+ line usage with i2004 phones. Reported by: Daniel Bohling,
+ MihaiMircea (closes issue ASTERISK-21061) (closes issue
+ ASTERISK-21120)
+
+2013-05-20 17:43 +0000 [r389245] Jason Parker <jparker@digium.com>
+
+ * /: Add doxygen.log to svn:ignore property. ........ Merged
+ revisions 389244 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-15 15:57 +0000 [r388839] kharwell <kharwell@localhost>:
+
+ * main/lock.c, /: Fix for segfault in __ast_rwlock_destroy with
+ DEBUG_THREADS If DEBUG_THREADS is enabled __ast_rwlock_destroy
+ causes a segfault while trying to access a possible NULL t->track
+ object. A NULL check has been added before trying to access the
+ memory. (closes issue ASTERISK-21724) Reported by: Corey Farrell
+ Fixed by: Corey Farrell Patches: ast_rwlock_destroy-segv.patch
+ uploaded by Corey Farrell (license 5909) ........ Merged
+ revisions 388838 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-15 14:25 +0000 [r388816] Jason Parker <jparker@digium.com>
+
+ * apps/app_voicemail.c: Fix VM snapshot handling for combined
+ INBOX. The snapshot API contains an option that allow for
+ combining of new and old messages within a single snapshot. New
+ messages, however, include options beyond just 'INBOX' - it also
+ includes the Urgent folder. A previous patch that combined INBOX
+ and Urgent accidentally impacted snapshots that attempted to gain
+ messages from just the Old folder. This patch fixes the snapshot
+ gathering such that the API returns the appropriate messages for
+ the folder selected, with and without the combine option. This
+ should make it more clear about what's happening. Review:
+ https://reviewboard.asterisk.org/r/2539/
+
+2013-05-15 12:39 +0000 [r388769] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_srtp.c, /, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Use srtp_shutdown when available This allows the
+ SRTP library to be shut down properly when the functionality is
+ offered by libsrtp. Review:
+ https://reviewboard.asterisk.org/r/2538/ (closes issue
+ ASTERISK-21719) ........ Merged revisions 388768 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-14 18:55 +0000 [r388700] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/astobj2.h, main/astobj2.c: Make ao2 global
+ objects not always use the debug version of the ao2_ref() calls.
+ The debug versions of ao2_ref() should only be used if REF_DEBUG
+ is enabled so nothing is written to /tmp/refs unexpectedly.
+ (closes issue ASTERISK-21785) Reported by: abelbeck Patches:
+ jira_asterisk_21785_v11.patch (license #5621) patch uploaded by
+ rmudgett Tested by: abelbeck
+
+2013-05-13 21:17 +0000 [r388601-388605] Michael L. Young <elgueromexicano@gmail.com>
+
+ * main/logger.c: Fix Missing CALL-ID When Logging Through Syslog
+ The CALL-ID (ie [C-00000074]) is missing when logging to syslog.
+ This was just an oversight when this feature was added. * Add
+ CALL-IDs when using syslog (closes issue ASTERISK-21430) Reported
+ by: Nikola Ciprich Tested by: Nikola Ciprich, Michael L. Young
+ Patches: asterisk-21430-syslog-callid_trunk.diff by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2526/
+
+ * channels/chan_sip.c: Fix Crash Caused By One-way Audio With
+ auto_* NAT Settings Fix The prior code committed, r385473, failed
+ to take into consideration that not all outgoing calls will be to
+ a peer. My fault. This patch does the following: * Check if there
+ is a related peer involved. If there is, check and set NAT
+ settings according to the peer's settings. * Fix a problem with
+ realtime peers. If the global setting has auto_force_rport set
+ and we issued a "sip reload" while a peer is still registered,
+ the peer's flags for NAT are reset to off. When this happens, we
+ were always setting the contact address of the peer to that of
+ the full contact info that we had. (closes issue ASTERISK-21374)
+ Reported by: jmls Tested by: Michael L. Young Patches:
+ asterisk-21374-fix-crash-and-rt-peers.diff by Michael L. Young
+ (license 5026) Review: https://reviewboard.asterisk.org/r/2524/
+
+2013-05-13 20:35 +0000 [r388597] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_srtp.c, /: Revert r388529 for now Adding the cleanup
+ function needs some deeper thought since it apparently doesn't
+ exist for all variants of libsrtp. ........ Merged revisions
+ 388596 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-13 19:24 +0000 [r388578] Jonathan Rose <jrose@digium.com>
+
+ * main/pbx.c, /: pbx: Fix lack of cleanup on macrolock and
+ context_table (closes issue ASTERISK-21723) Reported by: Corey
+ Farrell Patches: core-pbx-cleanup.patch uploaded by Correy
+ Farrell (license 5909) ........ Merged revisions 388532 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-13 18:09 +0000 [r388530] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_srtp.c, /: Close libsrtp properly Ensure that libsrtp is
+ shutdown properly when res_srtp is unloaded. (closes issue
+ ASTERISK-21719) Reported by: Corey Farrell Patches:
+ res_srtp-library-shutdown.patch uploaded by Corey Farrell
+ ........ Merged revisions 388529 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-13 14:26 +0000 [r388478] Richard Mudgett <rmudgett@digium.com>
+
+ * main/manager.c, /: Fix SendText AMI action to never return
+ non-zero. AMI actions must never return non-zero unless they
+ intend to close the AMI connection. (Which is almost never.)
+ (closes issue ASTERISK-21779) Reported by: Paul Goldbaum ........
+ Merged revisions 388477 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-10 22:11 +0000 [r388424-388426] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/misdn/isdn_msg_parser.c: Allow mISDN to send PROGRESS
+ messsage. * Made isdn_msg_parser.c build a progress message with
+ the mandatory progress indicator IE. (The mISDNuser NT state
+ machine rejected sending the incomplete message.) Note: The
+ associated mISDN and mISDNuser patches respectively are viewable
+ here: http://svnview.digium.com/svn/thirdparty?view=rev&rev=200
+ http://svnview.digium.com/svn/thirdparty?view=rev&rev=201 (closes
+ issue AST-1153) Reported by: Guenther Kelleter Patches:
+ progress-chan_misdn.diff (license #6372) patch uploaded by
+ Guenther Kelleter progress-misdn.diff (license #6372) mISDN patch
+ uploaded by Guenther Kelleter progress-misdnuser.diff (license
+ #6372) mISDNuser patch uploaded by Guenther Kelleter ........
+ Merged revisions 388425 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * utils, /: Add version.c to list of ignored files in the utils
+ directory. ........ Merged revisions 388423 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-10 20:41 +0000 [r388378] Mark Michelson <mmichelson@digium.com>
+
+ * /, pbx/pbx_dundi.c: Fix memory leak in pbx_dundi pbx_dundi added
+ an io context without removing it. This caused a memory leak when
+ the module was unloaded. (closes ASTERISK-21718) Reported by
+ Corey Farrell Patches: pbx_dundi-ast_io_remove.patch uploaded by
+ Corey Farrell (License #5909) ........ Merged revisions 388376
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-10 11:46 +0000 [r388253] Sean Bright <sean@malleable.com>
+
+ * channels/chan_sip.c: Fix copy/paste error in one-touch-recording
+ implementation.
+
+2013-05-09 04:10 +0000 [r388108-388112] Michael L. Young <elgueromexicano@gmail.com>
+
+ * res/res_rtp_asterisk.c, /: Fix The Payload Being Set On CN
+ Packets And Do Not Set Marker Bit When we send out a CN packet
+ (for instance, in the case of using rtpkeepalives), we are not
+ setting the payload code properly. Also, we are setting the
+ marker bit when we shouldn't be according to RFC 3389, section 4.
+ AST_RTP_CN is not defined by AST_FORMAT codes. Therefore, we
+ should be using ast_rtp_codecs_payload_code() rather than
+ ast_rtp_codecs_payload_lookup(). 11 and trunk already use the
+ appropriate function. * In 1.8, use ast_rtp_codecs_payload_code()
+ * Remove the setting of the marker bit * Fix the debug message by
+ incrementing the seqno after the debug message is set in order to
+ display the correct seqno that was sent out (closes issue
+ ASTERISK-21246) Reported by: Peter Katzmann Tested by: Peter
+ Katzmann, Michael L. Young Patches:
+ asterisk-21246-rtp-cng-payload-error_1.8_v2.diff uploaded by
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2500/ ........ Merged
+ revisions 388111 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_queue.c: Fix Segfault In app_queue When
+ "persistentmembers" Is Enabled And Using Realtime When the
+ "ignorebusy" setting was deprecated, we added some code to allow
+ us to be compatible with older setups that are still using the
+ "ignorebusy" setting instead of "ringinuse". We set a char
+ *variable with the column name to use, which helps the realtime
+ functions to use the correct column in their SQL queries. When
+ "persistentmembers" is enabled, we are not setting this variable
+ before the realtime functions were called to load members. This
+ results in the variable being NULL and therefore causing a
+ segfault when loading members during the module's process of
+ loading. The solution was to move the code that sets that
+ variable to be before these realtime functions are called during
+ the loading of the module. (closes issue ASTERISK-21738) Reported
+ by: JoshE Tested by: JoshE Patches:
+ asterisk-21738-rt-ringinuse-field-not-set.diff uploaded by
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2499/
+
+2013-05-08 07:19 +0000 [r387880] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, channels/chan_sip.c: chan_sip: NOTIFYs for BLF start queuing
+ up and fail to be sent out after retries fail RFC6665 4.2.2: ...
+ after a failed State NOTIFY transaction remove the subscription
+ The problem is that the State Notify requests rely on the 200OK
+ reponse for pacing control and to not confuse the notify
+ susbsystem. The issue is, the pendinginvite isn't cleared if a
+ response isn't received, thus further notify's are never sent.
+ The solution, follow RFC 6665 4.2.2's 'SHOULD' and remove the
+ subscription after failure. (closes issue ASTERISK-21677)
+ Reported by: Dan Martens Tested by: Dan Martens, David Brillert,
+ alecdavis alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2475/ ........ Merged
+ revisions 387875 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-07 18:29 +0000 [r387823] David M. Lee <dlee@digium.com>
+
+ * res/res_config_pgsql.c, main/manager.c: Minor fixups to Doxygen
+ comments. The \example tags marks an entire file as an example,
+ not a code snippet.
+
+2013-05-06 15:55 +0000 [r387689] Russell Bryant <russell@russellbryant.com>
+
+ * /, apps/app_meetme.c: Make SLA reload more paranoid. Reload
+ support was originally not included for SLA. It was added later,
+ but in a fairly non-traditional way. It basically sets a flag
+ indicating that a reload is pending, and then waits for a time
+ where it thinks everything SLA related is idle and unused, and
+ *then* executes the reload. It does this because the reload
+ process is destructive. It starts by throwing everything away and
+ starting over. There are a number of problems with this approach.
+ One of them is that the check to see if anything in use was
+ incomplete. This patch makes it more complete and thus less
+ likely for a crash to occur during reload processing. However,
+ this approach still has problems so some much more significant
+ reworking of this code will need to come in as a next step. Patch
+ credit and testing by CoreDial, LLC. ........ Merged revisions
+ 387688 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-02 17:15 +0000 [r387422] Matthew Jordan <mjordan@digium.com>
+
+ * utils/Makefile, /: Update utils Makefile to handle r387294 Alec's
+ patch that added the Asterisk version to 'core show locks'
+ angered the items in utils, as they exist somewhat outside of the
+ Asterisk build system. Some day, this Makefile should get nuked
+ from high orbit, but for now, include version.c in its list of
+ stuff to pile in. ........ Merged revisions 387421 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-02 08:09 +0000 [r387295-387345] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c: chan_sip:
+ Session-Expires: Set timer to correctly expire at (~2/3) of the
+ interval when not the refresher RFC 4028 Section 10 if the side
+ not performing refreshes does not receive a session refresh
+ request before the session expiration, it SHOULD send a BYE to
+ terminate the session, slightly before the session expiration.
+ The minimum of 32 seconds and one third of the session interval
+ is RECOMMENDED. Prior to this asterisk would refresh at 1/2 the
+ Session-Expires interval, or if the remote device was the
+ refresher, asterisk would timeout at interval end. Now, when not
+ refresher, timeout as per RFC noted above. (closes issue
+ ASTERISK-21742) Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2488/ ........ Merged
+ revisions 387344 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: chan_sip: Honor Session-Expires in 200OK
+ response when it's a RE-INVITE when asterisk is the refresher.
+ RFC 4028 Section 7.2 "UACs MUST be prepared to receive a
+ Session-Expires header field in a response, even if none were
+ present in the request." What changed After ASTERISK-20787,
+ inbound calls to asterisk with no Session-Expires in the INVITE
+ are now are offered a Session-Expires (1800 asterisk default) in
+ the response, with asterisk as the refresher. Symptom: After 900
+ seconds (asterisk default refresher period 1800), asterisk
+ RE-INVITEs the device, the device may respond with a much lower
+ Session-Expires (180 in our case) value that it is now using.
+ Asterisk ignores this response, as it's deemed both an INBOUND
+ CALL, and a RE-INVITE. After 180 seconds the device times out and
+ sends BYE (hangs up), asterisk is still working with the
+ refresher period of 1800 as it ignored the 'Session Expires: 180'
+ in the previous 200OK response. Fix: handle_response_invite()
+ when 200OK, remove check for outbound and reinvite. (closes issue
+ ASTERISK-21664) Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2463/ ........ Merged
+ revisions 387312 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_dahdi.c, /: chan_dahdi: fix lower bound check with
+ -ve integer conversion from a float Lower bound of a 16bit signed
+ int is -32768 not -32767 (closes issue ASTERISK-21744) Reported
+ by: alecdavis Tested by: alecdavis alecdavis (license 585)
+ ........ Merged revisions 387297 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/utils.c: Add Asterisk Version to core show locks Assist
+ with reporting 'core show locks' when submitting bug reports.
+ Example below: =========================== == SVN-branch-1.8-...
+ == Currently Held Locks =========================== (closes issue
+ ASTERISK-21743) Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) ........ Merged revisions 387294 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-01 21:17 +0000 [r387038-387216] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_rtp_asterisk.c, /: Clear the DTMF sending digit tracking
+ on off nominal paths In certain situations, when the RTP engine
+ goes to send a DTMF end digit it may be in a situation where the
+ remote address is no longer available, or the digit that was
+ supposed to be sent is invalid. In such cases, we need to clear
+ the RTP counters appropriately. Otherwise, when the RTP source is
+ set again, we'll continue to think that we're in the middle of
+ sending a DTMF digit, which can confuse the remote party
+ (signficantly). (closes issue ASTERISK-21522) Reported by: Corey
+ Farrell patches: rtp_dtmf_process_end.patch uploaded by Corey
+ Farrell (License 5909) ........ Merged revisions 387213 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_sip.c: Prevent crash in 'sip show peers' when the
+ number of peers on a system is large When you have lots of SIP
+ peers (according to the issue reporter, around 3500), the 'sip
+ show peers' CLI command or AMI action can crash due to a poorly
+ placed string duplication that occurs on the stack. This patch
+ refactors the command to not allocate the string on the stack,
+ and handles the formatting of a single peer in a separate
+ function call. (closes issue ASTERISK-21466) Reported by:
+ Guillaume Knispel patches:
+ fix_sip_show_peers_stack_overflow_asterisk_11.3.0-v2.patch
+ uploaded by gknispel (License 6492)
+
+ * /, main/features.c: Fix CDR not being created during an
+ externally initiated blind transfer Way back when in the dark
+ days of Asterisk 1.8.9, blind transferring a call in a context
+ that included the 'h' extension would inadvertently execute the
+ hangup code logic on the transferred channel. This was a "bad
+ thing". The fix was to properly check for the softhangup flags on
+ the channel and only execute the 'h' extension logic (and, in
+ later versions, hangup handler logic) if the channel was well and
+ truly dead (Jim). Unfortunately, CDRs are fickle. Setting the
+ softhangup flag when we detected that the channel was leaving the
+ bridge (but not to die) caused some crucial snippet of CDR code,
+ lying in ambush in the middle of the bridging code, to not get
+ executed. This had the effect of blowing away one of the CDRs
+ that is typically created during a blind transfer. While we live
+ and die by the adage "don't touch CDRs in release branches", this
+ was our bad. The attached patch restores the CDR behavior, and
+ still manages to not run the 'h' extension during a blind
+ transfer (at least not when it's supposed to). Thanks to Steve
+ Davies for diagnosing this and providing a fix. Review:
+ https://reviewboard.asterisk.org/r/2476 (closes issue
+ ASTERISK-21394) Reported by: Ishfaq Malik Tested by: Ishfaq
+ Malik, mjordan patches: fix_missing_blindXfer_cdr2 uploaded by
+ one47 (License 5012) ........ Merged revisions 387036 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-30 22:15 +0000 [r387030] Jonathan Rose <jrose@digium.com>
+
+ * main/event.c: Add forgotten event types to event_names array
+
+2013-04-30 13:46 +0000 [r386930] Sean Bright <sean@malleable.com>
+
+ * include/asterisk/utils.h, /: Use the proper lower bound when
+ doing saturation arithmetic. 16 bit signed integers have a range
+ of [-32768, 32768). The existing code was using the interval
+ (-32768, 32768) instead. This patch fixes that. Review:
+ https://reviewboard.asterisk.org/r/2479/ ........ Merged
+ revisions 386929 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-29 23:35 +0000 [r386878] Rusty Newton <rnewton@digium.com>
+
+ * /, sounds/Makefile: Modifying sounds/Makefile to pull down 1.4.24
+ core sounds 1.4.24 core sounds includes a full set of Italian
+ prompts for core sounds and a fix for the missing voicemail
+ prompts in the Russian language. (closes issue ASTERISK-19431)
+ (closes issue ASTERISK-19721) ........ Merged revisions 386877
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-29 08:54 +0000 [r386794] Olle Johansson <oej@edvina.net>
+
+ * /, CHANGES, apps/app_queue.c: Play periodic prompts for first
+ call in a call queue Review:
+ https://reviewboard.asterisk.org/r/2263/ ........ Merged
+ revisions 386792 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-26 21:27 +0000 [r386642-386677] Matthew Jordan <mjordan@digium.com>
+
+ * main/config.c, /: Clean up memory leak in config file on off
+ nominal paths when glob is allowed If a system allows for its
+ usage, Asterisk will use glob to help parse Asterisk .conf files.
+ The config file loading routine was leaking the memory allocated
+ by the glob() routine when the config file was in an unmodified
+ or invalid state. This patch properly calls globfree in those off
+ nominal paths. (closes issue ASTERISK-21412) Reported by: Corey
+ Farrell patches: config_glob_leak.patch uploaded by Corey Farrell
+ (license 5909) ........ Merged revisions 386672 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/features.c: Clean up resources in features on exit This
+ patch cleans up two things features: * It properly unregisters
+ the CLI commands that features registered * It cancels and
+ performs a pthread_join on the created parking thread. This not
+ only properly joins a non-detached thread, but also prevents
+ disposing of the parking lots prior to the parking thread
+ completely exiting. (closes issue ASTERISK-21407) Reported by:
+ Corey Farrell patches: features_shutdown-r2.patch uploaded by
+ Corey Farrell (License 5909) ........ Merged revisions 386641
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-25 03:02 +0000 [r386484-386486] Michael L. Young <elgueromexicano@gmail.com>
+
+ * channels/chan_sip.c: Fix Displaying Symmetric RTP Global Setting
+ * Use comedia_string() to display correctly the symmetric rtp
+ setting when running "sip show settings"
+
+ * /, channels/chan_sip.c: Change Case On Forcerport For Consistency
+ * Change "ForcerPort" to "Forcerport" to match everywhere else it
+ is displayed ........ Merged revisions 386483 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-22 16:30 +0000 [r386286] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /: Fix crash when AMI redirect action redirects
+ two channels out of a bridge. The two party bridging loops were
+ changing the bridge peer pointers without the channel locks held.
+ Thus when ast_channel_massquerade() tested and used the pointer
+ there is a small window of opportunity for the pointers to become
+ NULL even though the masquerade code has the channels locked.
+ (closes issue ASTERISK-21356) Reported by: William luke Patches:
+ jira_asterisk_21356_v11.patch (license #5621) patch uploaded by
+ rmudgett Tested by: William luke ........ Merged revisions 386256
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-19 22:25 +0000 [r386159] Matthew Jordan <mjordan@digium.com>
+
+ * /, res/res_timing_pthread.c: Prevent res_timing_pthread from
+ blocking callers There were several reports of deadlock when
+ using res_timing_pthread. Backtraces indicated that one thread
+ was blocked waiting for the write to the pipe to complete and
+ this thread held the container lock for the timers. Therefore any
+ thread that wanted to create a new timer or read an existing
+ timer would block waiting for either the timer lock or the
+ container lock and deadlock ensued. This patch changes the way
+ the pipe is used to eliminate this source of deadlocks: 1) The
+ pipe is placed in non-blocking mode so that it would never block
+ even if the following changes someone fail... 2) Instead of
+ writing bytes into the pipe for each "tick" that's fired the pipe
+ now has two states--signaled and unsignaled. If signaled, the
+ pipe is hot and any pollers of the read side filedescriptor will
+ be woken up. If unsigned the pipe is idle. This eliminates even
+ the chance of filling up the pipe and reduces the potential
+ overhead of calling unnecessary writes. 3) Since we're tracking
+ the signaled / unsignaled state, we can eliminate the exta poll
+ system call for every firing because we know that there is data
+ to be read. (closes issue ASTERISK-21389) Reported by: Matt
+ Jordan Tested by: Shaun Ruffell, Matt Jordan, Tony Lewis patches:
+ 0001-res_timing_pthread-Reduce-probability-of-deadlocking.patch
+ uploaded by sruffell (License 5417) (closes issue ASTERISK-19754)
+ Reported by: Nikola Ciprich (closes issue ASTERISK-20577)
+ Reported by: Kien Kennedy (closes issue ASTERISK-17436) Reported
+ by: Henry Fernandes (closes issue ASTERISK-17467) Reported by:
+ isrl (closes issue ASTERISK-17458) Reported by: isrl Review:
+ https://reviewboard.asterisk.org/r/2441/ ........ Merged
+ revisions 386109 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-19 05:18 +0000 [r386006-386051] David M. Lee <dlee@digium.com>
+
+ * main/cli.c, /: cli.c: Properly initialize debug_modules and
+ verbose_modules. This avoids some lock errors on the core set
+ {debug,verbose} commands. ........ Merged revisions 386049 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/message.c: Fix lock errors on startup. In messages.c, there
+ are several places in the code where we create a tmp_tech_holder
+ and pass that into an ao2_find call. Unfortunately, we weren't
+ initializing the rwlock on the tmp_tech_holder, which the hash
+ function was locking. It's apparently harmless, but still not the
+ best code. This patch extracts all that copy/pasted code into two
+ functions, msg_find_by_tech and msg_find_by_tech_name, which
+ properly initialize and destroy the rwlock on the
+ tmp_tech_holder. Review: https://reviewboard.asterisk.org/r/2454/
+
+2013-04-16 23:27 +0000 [r385917-385938] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * res/res_xmpp.c: Distributed Device State broken at sites using
+ res_xmpp or res_jabber where Secuity Advisory AST-2012-015 is
+ inplace res_xmpp was not adding AST_EVENT_IE_CACHABLE to the
+ event as each message came in, then
+ devstate_change_collector_cb() was unable to find
+ AST_EVENT_IE_CACHABLE in the event, so defaulted incorrectly to
+ AST_DEVSTATE_NOT_CACHABLE. (issue ASTERISK-20175) (closes issue
+ ASTERISK-21429) (closes issue ASTERISK-21069) (closes issue
+ ASTERISK-21164) Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2452/
+
+ * /, main/devicestate.c, res/res_jabber.c: Distributed Device State
+ broken at sites using res_xmpp or res_jabber where Secuity
+ Advisory AST-2012-015 is inplace res_jabber/res_xmpp were not
+ adding AST_EVENT_IE_CACHABLE to the event as each message came
+ in, then devstate_change_collector_cb() was unable to find
+ AST_EVENT_IE_CACHABLE in the event, so defaulted incorrectly to
+ AST_DEVSTATE_NOT_CACHABLE. (issue ASTERISK-20175) (closes issue
+ ASTERISK-21429) (closes issue ASTERISK-21069) (closes issue
+ ASTERISK-21164) Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2452/ ........ Merged
+ revisions 385916 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-15 17:23 +0000 [r385768] Jason Parker <jparker@digium.com>
+
+ * Makefile, /: Don't unnecessarily rebuild things on every run of
+ 'make'. Review: https://reviewboard.asterisk.org/r/2449/ ........
+ Merged revisions 385745 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-15 15:18 +0000 [r385689] David M. Lee <dlee@digium.com>
+
+ * channels/sig_ss7.c, channels/sip/include/security_events.h,
+ contrib/realtime/mysql/queue_log.sql,
+ channels/chan_multicast_rtp.c, channels/sig_ss7.h, /,
+ tests/test_expr.c, apps/app_saycounted.c,
+ channels/sip/security_events.c,
+ contrib/realtime/mysql/voicemail_messages.sql, BSDmakefile,
+ contrib/realtime/mysql/voicemail_data.sql,
+ build_tools/sha1sum-sh, res/res_mutestream.c,
+ configs/res_curl.conf.sample, tests/test_func_file.c,
+ include/asterisk/select.h, res/res_rtp_multicast.c,
+ include/asterisk/bridging_technology.h,
+ include/asterisk/bridging_features.h, tests/test_locale.c,
+ doc/Makefile, tests/test_poll.c,
+ contrib/realtime/mysql/musiconhold.sql, res/res_timing_kqueue.c:
+ Fix the svn:keywords property on several files. Normally I think
+ keyword expansion is silly, but the one time it would have been
+ good, it didn't work because the property had quotes in it. This
+ patch fixes obviously busted svn:keywords properties. ........
+ Merged revisions 385683 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-14 03:00 +0000 [r385634-385637] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_rtp_multicast.c, /: Calculate the timestamp for outbound
+ RTP if we don't have timing information This patch calculates the
+ timestamp for outbound RTP when we don't have timing information.
+ This uses the same approach in res_rtp_asterisk. Thanks to both
+ Pietro and Tzafrir for providing patches. (closes issue
+ ASTERISK-19883) Reported by: Giacomo Trovato Tested by: Pietro
+ Bertera, Tzafrir Cohen patches: rtp-timestamp-1.8.patch uploaded
+ by tzafrir (License 5035) rtp-timestamp.patch uploaded by
+ pbertera (License 5943) ........ Merged revisions 385636 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_alsa.c: Don't attempt to create a voice frame on
+ a read error Prior to this patch, a read error in snd_pcm_readi
+ would still be treated as a nominal result when constructing a
+ voice frame from the expected data. Since the value returned is
+ negative, as opposed to the number of samples read, this could
+ result in a crash. With this patch, we now return a null frame
+ when a read error is detected. Note that the patch on
+ ASTERISK-21329 was modified slightly for this commit, in that we
+ bail immediately on detecting the read error, rather than
+ bypassing the construction of the voice frame. (closes issue
+ ASTERISK-21329) Reported by: Keiichiro Kawasaki patches:
+ chan_alsa.diff uploaded by kawasaki (License 6489) ........
+ Merged revisions 385633 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-12 22:37 +0000 [r385594] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, apps/app_queue.c: Fix Manager Segfault When app_queue Is
+ Unloaded When app_queue is unloaded, some manager commands are
+ not being unregistered which result in a segfault. This patch
+ corrects this. (closes issue ASTERISK-21397) Reported by: Peter
+ Katzmann, Corey Farrell Tested by: Corey Farrell Patches:
+ asterisk-21397-missing-unreg-manager-cmd_1.8.diff Michael L.
+ Young (license 5026)
+ asterisk-21397-missing-unreg-manager-cmd_11.diff Michael L. Young
+ (license 5026) Review: https://reviewboard.asterisk.org/r/2444/
+ ........ Merged revisions 385593 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-12 22:25 +0000 [r385582] Kinsey Moore <kmoore@digium.com>
+
+ * codecs/codec_resample.c: Allow codec_resample to be unloaded
+ Ensure that trans_size is correct to prevent uninitialized
+ entries from preventing reload. (closes issue ASTERISK-21401)
+ Reported by: Corey Farrell Tested by: Corey Farrell Patches:
+ codec_resample-unload.patch uploaded by Corey Farrell
+
+2013-04-12 22:18 +0000 [r385473-385557] Michael L. Young <elgueromexicano@gmail.com>
+
+ * apps/app_voicemail.c, /: Fix app_voicemail Segfault And A Few
+ Memory Leaks The original report was that app_voicemail would
+ crash. This was caused by ast_config_load() returning
+ CONFIG_STATUS_FILEINVALID but no checks being performed for that
+ return status. After adding the initial patch to fix this issue,
+ Jaco Kroon (jkroon) added some fixes to memory leaks he had
+ discovered. During review, Walter Doekes (wdoekes) suggested
+ adding a helper function in order to determine if we had a valid
+ configuration or not. This patch does the following: * Creates a
+ helper function to check if the configuration is valid * Adds
+ calls to the new helper function where appropiate * Fixes memory
+ leaks where the code returned without running
+ ast_config_destroy() on the configuration that was loaded (closes
+ issue ASTERISK-21302) Reported by: Jaco Kroon Tested by: Jaco
+ Kroon, Michael L. Young Patches:
+ asterisk-11.3.0-app_voicemail-ast_config-fixes.patch Jaco Kroon
+ (license 5671) asterisk-21302-valid_cfg_and_mem_leaks_v3-1.8.diff
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2443/ ........ Merged
+ revisions 385551 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_sip.c: Fix One-Way Audio With auto_* NAT Settings
+ When SIP Calls Initiated By PBX When we reload Asterisk or
+ chan_sip, the flags force_rport and comedia that are turned on
+ and off when using the auto_force_rport and auto_comedia nat
+ settings go back to the default setting off. These flags are
+ turned on when needed or off when not needed at the time that a
+ peer registers, re-registers or initiates a call. This would
+ apply even when only the default global setting
+ "nat=auto_force_rport" is being used, which in this case would
+ only affect the force_rport flag. Everything is good except for
+ the following: The nat setting is set to auto_force_rport and
+ auto_comedia. We reload Asterisk and the peer's registration has
+ not expired. We load in the settings for the peer which turns
+ force_rport and comedia back to off. Since the peer has not
+ re-registered or placed a call yet, those flags remain off. We
+ then initiate a call to the peer from the PBX. The force_rport
+ and comedia flags stay off. If NAT is involved, we end up with
+ one-way audio since we never checked to see if the peer is behind
+ NAT or not. This patch does the following: * Moves the checking
+ of whether a peer is behind NAT into its own function * Create a
+ function to set the peer's NAT flags if they are using the auto_*
+ NAT settings * Adds calls in sip_request_call() to these new
+ functions in order to setup the dialog according to the peer's
+ settings (closes issue ASTERISK-21374) Reported by: Michael L.
+ Young Tested by: Michael L. Young Patches:
+ asterisk-21374-auto-nat-outgoing-fix_v2.diff Michael L. Young
+ (license 5026) Review: https://reviewboard.asterisk.org/r/2421/
+
+2013-04-12 08:50 +0000 [r385403-385430] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, channels/chan_iax2.c: IAX2 defer_full_frames fail to get sent
+ Ensure iax2_process_thread is signalled when a deferred frame is
+ queued to it. (issue ASTERISK-18827) Reported by: alecdavis
+ Tested by: alecdavis alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2426/ ........ Merged
+ revisions 385429 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_iax2.c: IAX2, prevent network thread starting
+ before all helper threads are ready On startup, it's possible for
+ a frame to arrive before the processing threads were ready. In
+ iax2_process_thread() the first pass through falls into
+ ast_cond_wait, should a frame arrive before we are at
+ ast_cond_wait, the signal will be ignored. The result
+ iax2_process_thread stays at ast_cond_wait forever, with deferred
+ frames being queued. Fix: When creating initial idle
+ iax2_process_threads, wait for init_cond to be signalled after
+ each thread is started. (issue ASTERISK-18827) Reported by:
+ alecdavis Tested by: alecdavis alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2427/ ........ Merged
+ revisions 385402 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-11 19:59 +0000 [r385356] Jason Parker <jparker@digium.com>
+
+ * res/res_rtp_asterisk.c, build_tools/menuselect-deps.in,
+ configure, include/asterisk/autoconfig.h.in, configure.ac,
+ makeopts.in: Add dependency on libuuid, for res_rtp_asterisk
+ pjproject is what actually requires libuuid. (closes issue
+ ASTERISK-21125) reported by Private Name (Ed. note: Really?
+ Private Name? I am rolling my eyes so hard right now.)
+
+2013-04-11 16:52 +0000 [r385313] Richard Mudgett <rmudgett@digium.com>
+
+ * configs/cli_aliases.conf.sample: Fix 'pri intense debug span'
+ alias.
+
+2013-04-10 14:25 +0000 [r385173-385199] Matthew Jordan <mjordan@digium.com>
+
+ * /, res/res_config_ldap.c: Use LDAP memory management functions
+ instead of Asterisk's When MALLOC_DEBUG is enabled with
+ res_config_ldap, issues (munmap_chunk: invalid pointer errors)
+ can occur as the memory is being allocated with Asterisk's
+ wrappers around malloc/calloc/free/strdup, as opposed to the LDAP
+ library's wrappers. This patch uses the LDAP library's wrappers
+ where appropriate, so that compiling with MALLOC_DEBUG doesn't
+ cause more problems than it solves. Note that the patch listed
+ below was modified slightly for this commit to account for some
+ additional memory allocation/deallocations. (closes issue
+ ASTERISK-17386) Reported by: John Covert Tested by: Andrew Latham
+ patches: issue18789-1.8-r316873.patch uploaded by seanbright
+ (License 5060) ........ Merged revisions 385190 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: Fix crash in chan_sip when a core
+ initiated op occurs at the same time as a BYE When a BYE request
+ is processed in chan_sip, the current SIP dialog is detached from
+ its associated Asterisk channel structure. The tech_pvt pointer
+ in the channel object is set to NULL, and the dialog persists for
+ an RFC mandated period of time to handle re-transmits. While this
+ process occurs, the channel is locked (which is good).
+ Unfortunately, operations that are initiated externally have no
+ way of knowing that the channel they've just obtained (which is
+ still valid) and that they are attempting to lock is about to
+ have its tech_pvt pointer removed. By the time they obtain the
+ channel lock and call the channel technology callback, the
+ tech_pvt is NULL. This patch adds a few checks to some channel
+ callbacks that make sure the tech_pvt isn't NULL before using it.
+ Prime offenders were the DTMF digit callbacks, which would crash
+ if AMI initiated a DTMF on the channel at the same time as a BYE
+ was received from the UA. This patch also adds checks on
+ sip_transfer (as AMI can also cause a callback into this
+ function), as well as sip_indicate (as lots of things can queue
+ an indication onto a channel). Review:
+ https://reviewboard.asterisk.org/r/2434/ (closes issue
+ ASTERISK-20225) Reported by: Jeff Hoppe ........ Merged revisions
+ 385170 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-08 23:36 +0000 [r385048] Rusty Newton <rnewton@digium.com>
+
+ * /, configs/extconfig.conf.sample: Modified the list of keys for
+ the driver backends for sake of sample clarity Added a line
+ showing the mapping of "mysql" to res_config_mysql available in
+ add-ons. We used "mysql" as an example driver key in the sample,
+ but didn't show what module it mapped too. Also added a subtitle
+ above the list of keys for driver backends. ........ Merged
+ revisions 385047 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-05 20:34 +0000 [r384827] Michael L. Young <elgueromexicano@gmail.com>
+
+ * channels/chan_sip.c, UPGRADE.txt: Fix For Not Overriding The
+ Default Settings In chan_sip The initial report was that the
+ "nat" setting in the [general] section was not having any effect
+ in overriding the default setting. Upon confirming that this was
+ happening and looking into what was causing this, it was
+ discovered that other default settings would not be overriden as
+ well. This patch works similar to what occurs in build_peer(). We
+ create a temporary ast_flags structure and using a mask, we
+ override the default settings with whatever is set in the
+ [general] section. In the bug report, the reporter who helped to
+ test this patch noted that the directmedia settings were being
+ overriden properly as well as the nat settings. This issue is
+ also present in Asterisk 1.8 and a separate patch will be applied
+ to it. (issue ASTERISK-21225) Reported by: Alexandre Vezina
+ Tested by: Alexandre Vezina, Michael L. Young Patches:
+ asterisk-21225-handle-options-default-prob_v4.diff Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2385/
+
+2013-04-03 20:18 +0000 [r384689] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, /, channels/sig_pri.c:
+ chan_dahdi: Add inband_on_proceeding compatibility option. The
+ new inband_on_proceeding option causes Asterisk to assume inband
+ audio may be present when a PROCEEDING message is received. Q.931
+ Section 5.1.2 says the network cannot assume that the CPE side
+ has attached to the B channel at this time without explicitly
+ sending the progress indicator ie informing the CPE side to
+ attach to the B channel for audio. However, some non-compliant
+ ISDN switches send a PROCEEDING without the progress indicator ie
+ indicating inband audio is available and assume that the CPE
+ device has connected the media path for listening to ringback and
+ other messages. ASTERISK-17834 which causes this issue was
+ dealing with a non-compliant network switch. (closes issue
+ ASTERISK-21151) Reported by: Gianluca Merlo Tested by: rmudgett
+ ........ Merged revisions 384685 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-03 17:10 +0000 [r384641] Matthew Jordan <mjordan@digium.com>
+
+ * /, funcs/func_channel.c: Update documentation for CHANNEL
+ function Document that you can read/write the 'accountcode' and
+ 'amaflags' on a channel. ........ Merged revisions 384640 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-02 17:34 +0000 [r384545] David M. Lee <dlee@digium.com>
+
+ * Makefile, /: Fixed spurious rebuilds of func_version.
+ func_version.so was being rebuilt every time, because build.h was
+ changing every build, because of the cleantest dependency that
+ was added in r384410 to fix parallel make bugs. Now build.h will
+ only be created if it does not exist, which was the original
+ behavior of the Makefile. ........ Merged revisions 384544 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-04-01 14:07 +0000 [r384414] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_voicemail.c: Remove silly use of strncmp.
+
+2013-04-01 13:28 +0000 [r384411] David M. Lee <dlee@digium.com>
+
+ * Makefile, /: Fix parallel make problems. Occasionally, make -j
+ would fail due to missing includes, or other unusual errors. This
+ was due to the 'cleantest' target, which was designed to force a
+ make clean when some change in the code would cause the typical
+ depedency checking to fail. Several targets in the main Makefile
+ did not depend upon cleantest, hence would run in parallel to it.
+ By adding the dependency, make -j runs happily now. Review:
+ https://reviewboard.asterisk.org/r/2418/ ........ Merged
+ revisions 384410 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-29 16:31 +0000 [r384326] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_voicemail.c, /: app_voicemail: Add blank argument to
+ externnotify if no context argument At least one call to
+ run_externnotify provides a NULL context parameter and because
+ the snprintf statement doesn't account for a NULL context
+ parameter, it simply writes '(null)' to the arguments string
+ instead. This patch makes it write two quotes back to back for
+ that argument instead in the event of a NULL context. (closes
+ issue ASTERISK-18207) Reported by: Barry L. Kline Patches:
+ modified from patch-20130306 uploaded by Karsten Wemheuer
+ (License 5930) ........ Merged revisions 384325 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-05-17 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 11.4.0 Released.
+
+2013-05-15 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 11.4.0-rc3 Released.
+
+ * Fix VM snapshot handling for combined INBOX.
+
+ The snapshot API contains an option that allow for combining of new
+ and old messages within a single snapshot. New messages, however,
+ include options beyond just 'INBOX' - it also includes the Urgent
+ folder. A previous patch that combined INBOX and Urgent accidentally
+ impacted snapshots that attempted to gain messages from just the Old
+ folder. This patch fixes the snapshot gathering such that the API
+ returns the appropriate messages for the folder selected, with and
+ without the combine option.
+
+2013-05-09 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 11.4.0-rc2 Released.
+
+ * Fix Segfault In app_queue When "persistentmembers" Is Enabled And
+ Using Realtime
+
+ When the "ignorebusy" setting was deprecated, we added some code to
+ allow us to be compatible with older setups that are still using the
+ "ignorebusy" setting instead of "ringinuse". We set a char *variable
+ with the column name to use, which helps the realtime functions to
+ use the correct column in their SQL queries. When "persistentmembers"
+ is enabled, we are not setting this variable before the realtime
+ functions were called to load members. This results in the variable
+ being NULL and therefore causing a segfault when loading members
+ during the module's process of loading.
+
+ The solution was to move the code that sets that variable to be
+ before these realtime functions are called during the loading of the
+ module.
+
+ * Distributed Device State broken at sites using res_xmpp or res_jabber
+ where Secuity Advisory AST-2012-015 is inplace
+
+ res_jabber/res_xmpp were not adding AST_EVENT_IE_CACHABLE to the
+ event as each message came in, then devstate_change_collector_cb()
+ was unable to find AST_EVENT_IE_CACHABLE in the event, so defaulted
+ incorrectly to AST_DEVSTATE_NOT_CACHABLE.
+
+ * Fix CDR not being created during an externally initiated blind
+ transfer
+
+ Way back when in the dark days of Asterisk 1.8.9, blind transferring
+ a call in a context that included the 'h' extension would
+ inadvertently execute the hangup code logic on the transferred
+ channel. This was a "bad thing". The fix was to properly check for
+ the softhangup flags on the channel and only execute the 'h'
+ extension logic (and, in later versions, hangup handler logic) if
+ the channel was well and truly dead (Jim).
+
+ Unfortunately, CDRs are fickle. Setting the softhangup flag when we
+ detected that the channel was leaving the bridge (but not to die)
+ caused some crucial snippet of CDR code, lying in ambush in the
+ middle of the bridging code, to not get executed. This had the
+ effect of blowing away one of the CDRs that is typically created
+ during a blind transfer.
+
+ While we live and die by the adage "don't touch CDRs in release
+ branches", this was our bad. The attached patch restores the CDR
+ behavior, and still manages to not run the 'h' extension during a
+ blind transfer (at least not when it's supposed to).
+
+ Thanks to Steve Davies for diagnosing this and providing a fix.
+
+ * Prevent res_timing_pthread from blocking callers
+
+ There were several reports of deadlock when using res_timing_pthread.
+ Backtraces indicated that one thread was blocked waiting for the
+ write to the pipe to complete and this thread held the container lock
+ for the timers. Therefore any thread that wanted to create a new
+ timer or read an existing timer would block waiting for either the
+ timer lock or the container lock and deadlock ensued.
+
+ This patch changes the way the pipe is used to eliminate this source
+ of deadlocks:
+
+ 1) The pipe is placed in non-blocking mode so that it would never
+ block even if the following changes someone fail...
+
+ 2) Instead of writing bytes into the pipe for each "tick" that's
+ fired the pipe now has two states--signaled and unsignaled. If
+ signaled, the pipe is hot and any pollers of the read side
+ filedescriptor will be woken up. If unsigned the pipe is idle.
+ This eliminates even the chance of filling up the pipe and reduces
+ the potential overhead of calling unnecessary writes.
+
+ 3) Since we're tracking the signaled / unsignaled state, we can
+ eliminate the exta poll system call for every firing because we know
+ that there is data to be read.
+
+ * Fix crash when AMI redirect action redirects two channels out of a
+ bridge.
+
+ The two party bridging loops were changing the bridge peer pointers
+ without the channel locks held. Thus when ast_channel_massquerade()
+ tested and used the pointer there is a small window of opportunity
+ for the pointers to become NULL even though the masquerade code has
+ the channels locked.
+
+2013-03-28 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 11.4.0-rc1 Released.
+
+2013-03-27 19:51 +0000 [r384163] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c, main/format_pref.c: Address uninitialized
+ conditional that valgrind found ........ Merged revisions 384162
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-27 18:51 +0000 [r384119] Matthew Jordan <mjordan@digium.com>
+
+ * /, main/http.c: Fix a file descriptor leak in off nominal path
+ While looking at the security vulnerability in ASTERISK-20967,
+ Walter noticed a file descriptor leak and some other issues in
+ off nominal code paths. This patch corrects them. Note that this
+ patch is not related to the vulnerability in ASTERISK-20967, but
+ the patch was placed on that issue. (closes issue ASTERISK-20967)
+ Reported by: wdoekes patches:
+ issueA20967_file_leak_and_unused_wkspace.patch uploaded by
+ wdoekes (License 5674) ........ Merged revisions 384118 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-27 17:06 +0000 [r384049] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_rtp_asterisk.c, /: Fix white noise on SRTP decryption
+ When res_rtp_asterisk.c was altered to avoid attempting to apply
+ unprotect algorithms to non-audio RTP packets, the test used was
+ incorrect. This caused the audio packets to not be decrypted and
+ resulted in loud white noise on the other endpoint (or both
+ endpoints depending on the call legs involved). The test now
+ properly checks the version field in the RTP header to ensure
+ that RTP and RTCP are decrypted while other types of packets are
+ not. (closes issue ASTERISK-21323) Reported by: andrea Tested by:
+ Kinsey Moore, andrea, John Bigelow Patches: whitenoise_fix.diff
+ uploaded by Kinsey Moore ........ Merged revisions 384048 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-27 15:23 +0000 [r383973-384003] Matthew Jordan <mjordan@digium.com>
+
+ * channels/sip/include/sip.h, channels/chan_sip.c,
+ channels/sip/security_events.c: AST-2013-003: Prevent username
+ disclosure in SIP channel driver When authenticating a SIP
+ request with alwaysauthreject enabled, allowguest disabled, and
+ autocreatepeer disabled, Asterisk discloses whether a user exists
+ for INVITE, SUBSCRIBE, and REGISTER transactions in multiple
+ ways. The information is disclosed when: * A "407 Proxy
+ Authentication Required" response is sent instead of a "401
+ Unauthorized" response * The presence or absence of additional
+ tags occurs at the end of "403 Forbidden" (such as "(Bad Auth)")
+ * A "401 Unauthorized" response is sent instead of "403
+ Forbidden" response after a retransmission * Retransmission are
+ sent when a matching peer did not exist, but not when a matching
+ peer did exist. This patch resolves these various vectors by
+ ensuring that the responses sent in all scenarios is the same,
+ regardless of the presence of a matching peer. This issue was
+ reported by Walter Doekes, OSSO B.V. A substantial portion of the
+ testing and the solution to this problem was done by Walter as
+ well - a huge thanks to his tireless efforts in finding all the
+ ways in which this setting didn't work, providing automated
+ tests, and working with Kinsey on getting this fixed. (closes
+ issue ASTERISK-21013) Reported by: wdoekes Tested by: wdoekes,
+ kmoore patches: AST-2013-003-1.8 uploaded by kmoore, wdoekes
+ (License 6273, 5674) AST-2013-003-10 uploaded by kmoore, wdoekes
+ (License 6273, 5674) AST-2013-003-11 uploaded by kmoore, wdoekes
+ (License 6273, 5674)
+
+ * main/http.c: AST-2013-002: Prevent denial of service in HTTP
+ server AST-2012-014, fixed in January of this year, contained a
+ fix for Asterisk's HTTP server for a remotely-triggered crash.
+ While the fix put in place fixed the possibility for the crash to
+ be triggered, a denial of service vector still exists with that
+ solution if an attacker sends one or more HTTP POST requests with
+ very large Content-Length values. This patch resolves this by
+ capping the Content-Length at 1024 bytes. Any attempt to send an
+ HTTP POST with Content-Length greater than this cap will not
+ result in any memory allocation. The POST will be responded to
+ with an HTTP 413 "Request Entity Too Large" response. This issue
+ was reported by Christoph Hebeisen of TELUS Security Labs (closes
+ issue ASTERISK-20967) Reported by: Christoph Hebeisen patches:
+ AST-2013-002-1.8.diff uploaded by mmichelson (License 5049)
+ AST-2013-002-10.diff uploaded by mmichelson (License 5049)
+ AST-2013-002-11.diff uploaded by mmichelson (License 5049)
+
+ * res/res_format_attr_h264.c: AST-2013-001: Prevent buffer overflow
+ through H.264 format negotiation The format attribute resource
+ for H.264 video performs an unsafe read against a media attribute
+ when parsing the SDP. The value passed in with the format
+ attribute is not checked for its length when parsed into a fixed
+ length buffer. This patch resolves the vulnerability by only
+ reading as many characters from the SDP value as will fit into
+ the buffer. (closes issue ASTERISK-20901) Reported by: Ulf
+ Harnhammar patches: h264_overflow_security_patch.diff uploaded by
+ jrose (License 6182)
+
+2013-03-26 02:28 +0000 [r383840-383878] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Resolve deadlock between SIP registration
+ and channel based functions In r373424, several reentrancy
+ problems in chan_sip were addressed. As a result, the SIP channel
+ driver is now properly locking the channel driver private
+ information in certain operations that it wasn't previously. This
+ exposed two latent problems either in register_verify or by
+ functions called by register_verify. This includes: * Holding the
+ private lock while calling sip_send_mwi_to_peer. This can create
+ a new sip_pvt via sip_alloc, which will obtain the channel
+ container lock. This is a locking inversion, as any channel
+ related lock must be obtained prior to obtaining the SIP channel
+ technology private lock. Note that this issue was already fixed
+ in Asterisk 11. * Holding the private lock while calling
+ sip_poke_peer. In the same vein as sip_send_mwi_to_peer,
+ sip_poke_peer can create a new SIP private, causing the same
+ locking inversion. Note that this locking inversion typically
+ occured when CLI commands were run while a SIP REGISTER request
+ was being processed, as many CLI commands (such as 'sip show
+ channels', 'core show channels', etc.) have to obtain the channel
+ container lock. (issue ASTERISK-21068) Reported by: Nicolas
+ Bouliane (issue ASTERISK-20550) Reported by: David Brillert
+ (issue ASTERISK-21314) Reported by: Badalian Vyacheslav (issue
+ ASTERISK-21296) Reported by: Gabriel Birke ........ Merged
+ revisions 383863 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/cdr.c, /: Resolve deadlock between pending CDR and batch CDR
+ locks r375757 attempted to resolve a race condition between
+ multiple submissions of CDRs while in batch mode from attempting
+ to destroy the scheduled batch submission by extending the batch
+ CDR lock. Unfortunately, this causes a deadlock between the
+ pending CDR lock and the batch CDR lock. This patch resolves the
+ intent of r375757 by simply providing a new lock that protects
+ the scheduling of the batches. The original batch CDR lock is
+ kept to protect manipulation of the batch CDR settings, but has
+ been placed such that it is not held when the pending lock is
+ held. Thanks to Chase Venters for providing lock analysis on the
+ issue. (issue ASTERISK-21162) Reported by: Chase Venters ........
+ Merged revisions 383839 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-26 01:36 +0000 [r383836] Russell Bryant <russell@russellbryant.com>
+
+ * /, apps/app_meetme.c: Fix multi-station answer race condition.
+ When an SLA trunk is ringing (inbound call on the trunk) Asterisk
+ will make outbound calls to the stations that have that trunk. If
+ more than one station answers the call at the same time, all
+ channels other than the first one to answer are left in a bad
+ state. The channel gets leaked, is not connected to anything, and
+ there's no way to get rid of it. We now properly clean up these
+ losing channels by hanging up on them. Since they lost the race,
+ as we process their answer, there is no ringing trunk for them to
+ answer. ........ Merged revisions 383835 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-25 23:24 +0000 [r383798] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Set the CALLERID(dnid-num-plan) for
+ incoming ISDN calls. The CALLEDTON channel variable is set for
+ incoming ISDN calls to the lower 7 bits of the Q.931
+ type-of-number/numbering-plan octet. The CALLERID(dnid-num-plan)
+ should have the same value. (closes issue ASTERISK-21248)
+ Reported by: rmudgett ........ Merged revisions 383796 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-25 12:36 +0000 [r383668] Sean Bright <sean@malleable.com>
+
+ * res/res_config_curl.c, /: Properly delimit post data in
+ res_config_curl. ........ Merged revisions 383667 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-22 20:41 +0000 [r383631] Michael L. Young <elgueromexicano@gmail.com>
+
+ * apps/app_mixmonitor.c: Fix StopMixMonitor Hanging Up When Unable
+ To Stop MixMonitor On A Channel A regression was accidentally
+ introduced when allowing an optional ID to be used when calling
+ StopMixMonitor. When we are unable to stop MixMonitor on a
+ channel, -1 is being returned which triggers the hangup of the
+ channel. This patch restores the prior behavior by returning 0
+ whether we were successful or not. It also allows the call from
+ the manager to use the return code when the action fails. (closes
+ issue ASTERISK-21294) Reported by: daroz Tested by: daroz
+ Patches: asterisk-21294-stop_mixmonitor_hangingup.diff Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2404/
+
+2013-03-20 20:25 +0000 [r383457-383461] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * funcs/func_curl.c, /: Have func_curl log a warning when a curl
+ request fails. Review: https://reviewboard.asterisk.org/r/2403/
+ ........ Merged revisions 383460 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * funcs/func_curl.c: Minor cleanup in func_curl near hashcompat
+ code. Review: https://reviewboard.asterisk.org/r/2402/
+
+2013-03-19 15:58 +0000 [r383341-383342] David M. Lee <dlee@digium.com>
+
+ * codecs/Makefile: Remove codecs/speex/*.i on make clean
+
+ * codecs/Makefile, /: Removed codecs/g722/*.i on make clean
+ ........ Merged revisions 383340 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-16 15:14 +0000 [r383266] Joshua Colp <jcolp@digium.com>
+
+ * res/res_xmpp.c: Fix a bug where resources were not found due to
+ hashing on the priority itself.
+
+2013-03-15 12:51 +0000 [r383166] Kinsey Moore <kmoore@digium.com>
+
+ * main/tcptls.c, main/manager.c, /, channels/chan_sip.c,
+ main/http.c: tcptls: Prevent unsupported options from being set
+ AMI, HTTP, and chan_sip all support TLS in some way, but none of
+ them support all the options that Asterisk's TLS core is capable
+ of interpreting. This prevents consumers of the TLS/SSL layer
+ from setting TLS/SSL options that they do not support. This also
+ gets tlsverifyclient closer to a working state by requesting the
+ client certificate when tlsverifyclient is set. Currently, there
+ is no consumer of main/tcptls.c in Asterisk that supports this
+ feature and so it can not be properly tested. Review:
+ https://reviewboard.asterisk.org/r/2370/ Reported-by: John
+ Bigelow Patch-by: Kinsey Moore (closes issue AST-1093) ........
+ Merged revisions 383165 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-15 01:34 +0000 [r383121-383125] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: When a session timer expires during a
+ T.38 call, re-invite with correct SDP When a session timer
+ expires during a dialog that has re-negotiated to T.38 and
+ Asterisk is the refresher, Asterisk will send a re-INVITE with an
+ SDP containing audio media only. This causes some hilarity with
+ the poor fax session under weigh. This patch corrects that by
+ sending T.38 parameters if we are in the middle of a T.38
+ session. (closes issue ASTERISK-21232) Reported by: Nitesh Bansal
+ patches:
+ dont-send-audio-reinvite-for-sess-timer-in-t38-call.patch
+ uploaded by nbansal (License 6418) ........ Merged revisions
+ 383124 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * pbx/pbx_spool.c, /: Fix processing of call files when using
+ KQueue on OS X In certain situations, call files are not
+ processed when using KQueue with pbx_spool. Asterisk was sending
+ an invalid timeout value when the spool directory is empty,
+ causing the call to kevent to error immediately. This can create
+ a tight loop, increasing the CPU load on the system. (closes
+ issue ASTERISK-21176) Reported by: Carlton O'Riley patches:
+ kqueue_osx.patch uploaded by coriley (License 6473) ........
+ Merged revisions 383120 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-14 16:57 +0000 [r383062] Jason Parker <jparker@digium.com>
+
+ * autoconf/ast_ext_lib.m4, /: Fix whitespace in AST_EXT_LIB_CHECK
+ macro. ........ Merged revisions 383061 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-12 21:17 +0000 [r382940-382943] Michael L. Young <elgueromexicano@gmail.com>
+
+ * addons/res_config_mysql.c, /: Fix Sorting Order For Parking Lots
+ Stored In Static Realtime When retrieving the parking lots from a
+ MySQL database table, the current order is "filename, cat_metric
+ desc, var_metric asc, category". If there are multiple parking
+ lots with the same cat_metric but different categories,
+ everything is being sorted on cat_metric first resulting in
+ errors when loading the parking lots. This patch fixes the
+ problem by sorting on the category field first, then the
+ cat_metric field. (closes issue ASTERISK-21035) Reported by: Alex
+ Epshteyn Patches: asterisk-21035-orderby.diff Michael L. Young
+ (license 5026) ........ Merged revisions 382942 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * contrib/realtime/mysql/sippeers.sql, /,
+ contrib/realtime/postgresql/realtime.sql: Update Contributed
+ Realtime Schema Files - IPv6 Addresses This commit updates some
+ fields in the contributed realtime schema files to handle IPv6
+ addresses. (closes issue ASTERISK-21173) Reported by: Torrey
+ Searle Patches: realtime_sql.patch Torrey Searle (license 5334)
+ asterisk-21173-update-ip-fields.diff Michael L. Young (license
+ 5026) ........ Merged revisions 382939 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-12 20:06 +0000 [r382923] Joshua Colp <jcolp@digium.com>
+
+ * res/res_xmpp.c: Fix a crash when res_xmpp is configured using a
+ username without a domain. (closes issue ASTERISK-21156) Reported
+ by: amsoft2001
+
+2013-03-12 16:23 +0000 [r382848] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c, UPGRADE.txt: Include the Username field
+ in SIP Registry events when Status is registered In
+ ASTERISK-17888, the AMI Registry event during SIP registrations
+ was supposed to include the Username field. Somehow, one of the
+ events was missed. This patch corrects that - the Username field
+ should be included in all AMI Registry events involving SIP
+ registrations. (issue ASTERISK-17888) (closes issue
+ ASTERISK-21201) Reported by: Dmitriy Serov patches:
+ chan_sip.c.diff uploaded by Dmitriy Serov (license 6479) ........
+ Merged revisions 382847 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-12 08:53 +0000 [r382827] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * channels/chan_unistim.c: Fix core dump on CLI usage Fix issue
+ with 'unistim show info' CLI command when device connected not
+ configured
+
+2013-03-08 20:16 +0000 [r382739] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c: chan_sip: Update the via header when
+ relaying SMS MESSAGE Prior to this change, certain conditions for
+ sending the message would result in an address of '(null)' being
+ used in the via header of the SIP message because a NULl value of
+ pvt->ourip was used when initially generating the via header.
+ This is fixed by adding a call to build_via when the address is
+ set before sending the message. (closes issue ASTERISK-21148)
+ Reported by: Zhi Cheng Patches: 700-sip_msg_send_via_fix.patch
+ uploaded by Zhi Cheng (license 6475)
+
+2013-03-07 17:57 +0000 [r382617] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_voicemail.c: Let vm_mailbox_snapshot combine "Urgent"
+ when no folder is specified r381835 fixed a bug in
+ vm_mailbox_snapshot where combining INBOX and Old forgot that
+ Urgent also "counts" as new messages. This fixed the problem when
+ any of the three folders was specified and the combine option was
+ used. It missed the case where the folder isn't specified and we
+ build a snapshot of all folders. This patch corrects that.
+
+2013-03-07 15:08 +0000 [r382574] Kinsey Moore <kmoore@digium.com>
+
+ * main/logger.c: Ensure that logmsgs are freed properly Messages
+ sent while the logger thread is shutting down will now have their
+ associated callid freed properly.
+
+2013-03-07 14:58 +0000 [r382573] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_rtp_asterisk.c: Add a 'secret' probation strictrtp mode
+ to handle delayed changes in RTP source Often, Asterisk may
+ realize that a change in the source of an RTP stream is about to
+ occur and ask that the RTP engine reset it's lock on the current
+ RTP source. In certain scenarios, it may take awhile for the new
+ remote system to send RTP packets, while the old remote system
+ may continue providing RTP during that time period. This causes
+ Asterisk to re-lock onto the old source, thereby rejecting the
+ new source when the old source stops sending RTP and the new
+ source begins. This patch prevents that by having a constant
+ secondary, 'secret' probation mode enabled when an RTP source has
+ been chosen. RTP packets from other sources are always
+ considered, but never chosen unless the current RTP source stops
+ sending RTP. Review: https://reviewboard.asterisk.org/r/2364
+ (closes issue AST-1124) Reported by: John Bigelow Tested by: John
+ Bigelow (closes issue AST-1125) Reported by: John Bigelow Tested
+ by: John Bigelow
+
+2013-03-06 18:28 +0000 [r382514] Kinsey Moore <kmoore@digium.com>
+
+ * /: Recorded merge of revisions 382513 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Correct app_page documentation The 'A' and 'n' options for Page()
+ mention that the announcement will be played simultaneously. This
+ is not necessarily the case.
+
+2013-03-05 03:51 +0000 [r382410] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * channels/chan_unistim.c, /: Fix several unreleased mutex locks
+ that cause problem with processing calls Reported by: Daniel
+ Bohling Tested by: Daniel Bohling (Closes issue ASTERISK-21119)
+ ........ Merged revisions 382409 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-04 21:12 +0000 [r382390] Jason Parker <jparker@digium.com>
+
+ * /, main/event.c: Fix comparison of presence state in event
+ subsystem. Several new IEs were not given types (or names),
+ causing the comparison function to improperly succeed. This adds
+ those. (closes issue AST-1128)
+
+2013-03-04 20:03 +0000 [r382385] kharwell <kharwell@localhost>:
+
+ * apps/app_confbridge.c: Confbridge CLI new record file name check.
+ This fix checks to make sure that if a confbridge record start
+ command is issued from the CLI it will always use the file name
+ given on the CLI even if it changes between start/stop records
+ for a conference. Previously it had been reusing the same file
+ between start/stops even if a new filename was given. (issue
+ AST-1088) Reported by: John Bigelow
+
+2013-03-01 04:28 +0000 [r382322] Michael L. Young <elgueromexicano@gmail.com>
+
+ * contrib/realtime/mysql/sippeers.sql, channels/chan_sip.c,
+ contrib/realtime/postgresql/realtime.sql, CHANGES: Fix / Clean Up
+ Some Items To Handle The New auto_* NAT Options The original
+ report had to do with a realtime peer behind NAT being pruned and
+ the peer's private address being used instead of its external
+ address. Upon debugging, it was discovered that this was being
+ caused by the addition of the auto_force_rport and auto_comedia
+ settings. This patch does the following: * Adds a missing note to
+ the CHANGES file indicating that the default global nat setting
+ is auto_force_rport * Constify the 'req' parameter for
+ check_via() * Add calls to check_via() in a couple of places in
+ order for the auto_* settings to do their job in attempting to
+ determine if NAT is involved * Set the flags SIP_NAT_FORCE_RPORT
+ and SIP_PAGE2_SYMMETRICRTP if the auto_* settings are in use
+ where it was needed * Moves the copying of peer flags up in
+ build_peer() to before they are used; this fixes the realtime
+ prune issue * Update the contrib/realtime schemas to allow the
+ nat column to handle the different nat setting combinations we
+ have This patch received a review and "Ship It!" on the issue
+ itself. (closes issue ASTERISK-20904) Reported by: JoshE Tested
+ by: JoshE, Michael L. Young Patches:
+ asterisk-20904-nat-auto-and-rt-peersv2.diff Michael L. Young
+ (license 5026)
+
+2013-02-28 21:58 +0000 [r382296-382298] Joshua Colp <jcolp@digium.com>
+
+ * res/res_rtp_asterisk.c: While the ICE negotiation is occurring
+ leave strictrtp in an open state, media can and will come from
+ different places.
+
+ * res/res_rtp_asterisk.c: Fix a bug with ICE and strictrtp where
+ media could get dropped. If the end result of the ICE negotiation
+ resulted in the path for media changing it was possible for the
+ strictrtp code to discard the RTP packets. This change causes
+ strictrtp to enter learning mode once again when the ICE
+ negotiation has completed successfully.
+
+2013-02-28 17:16 +0000 [r382230-382234] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_iax2.c: Prevent deadlock in chan_iax2 when
+ attempting to set caller ID A deadlock can occur in chan_iax2
+ when it attempts to set the caller ID, as it already holds the
+ iax2 private lock and improperly fails to obtain the channel lock
+ before calling ast_set_callerid. By not safely obtaining the
+ channel lock, a locking inversion can take place, causing a
+ deadlock. This patch solves this by calling the required deadlock
+ avoidance functions that obtain the channel lock before setting
+ the caller ID. Thanks to Pavel for fixing my syntax errors and
+ testing this patch out. (closes issue ASTERISK-21128) Reported
+ by: Pavel Troller Tested by: Pavel Troller patches:
+ ASTERISK-21128-1.8.diff uploaded by mjordan (license 6283)
+ ASTERISK-21128-modified-1.8.diff uploaded by Pavel Troller
+ (license 6302) ........ Merged revisions 382233 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, apps/app_meetme.c, UPGRADE.txt: Let channels joining a MeetMe
+ conference opt out of the denoiser For some channel drivers,
+ specifically those that have a varying rate in the number of
+ audio samples, the audio quality for a MeetMe conference can be
+ exceedingly poor. This is due to a unilateral application of the
+ DENOISE function in func_speex to channels joining the
+ conference. The denoiser function in the speex library is
+ initialized with the number of audio samples in each sample that
+ will be provided to it. If the number of audio samples changes,
+ the denoiser has to be thrown away and re-initialized. While this
+ could be worked around by removing func_speex, that doesn't help
+ if you actually use the denoiser with other channels on the
+ system. This patches does the following: * Checks for the
+ presence of func_speex as opposed to codec_speex when determining
+ if the DENOISE function is present (which is where the function
+ is actually implemented) * Adds an option to MeetMe 'n' that
+ causes the denoiser to not be applied to a channel when it joins.
+ This keeps the current behavior the default, but let's users
+ disable the denoiser if it causes problems on their system.
+ Review: https://reviewboard.asterisk.org/r/2358 (closes issue
+ AST-1062) Reported by: Thomas Arimont ........ Merged revisions
+ 382227 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-02-27 16:17 +0000 [r382151-382174] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Relax dialog checking in
+ get_sip_pvt_byid_locked so it works when the dialog is forked.
+ (closes issue ASTERISK-20638) Reported by: eelcob Patches:
+ pedantic-call-pickup-from-tag.patch uploaded by eelcob (license
+ 6442) ........ Merged revisions 382171 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * configure, include/asterisk/autoconfig.h.in: Regenerate the
+ configure script. The one in the tree was not working for me at
+ all.
+
+2013-02-26 19:45 +0000 [r382111] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * /, configure, configure.ac: Consider linux-gnuspe as linux-gnu *
+ The powerpcspe Linux port uses linux-gnuspe as the OS string. *
+ Our build system shouldn't really care for that, so just call it
+ linux-gnu. * Original report: Roland Stigge ,
+ http://bugs.debian.org/701505 Review:
+ https://reviewboard.asterisk.org/r/2357/ ........ Merged
+ revisions 382110 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-02-26 19:34 +0000 [r382108] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, channels/chan_sip.c: Correct RPID parsing for unquoted
+ display-name. Parsing Remote-Party-ID will now succeed if
+ display-name is of the *(token LWS) kind and not just the
+ quoted-string kind. Review:
+ https://reviewboard.asterisk.org/r/2341/ ........ Merged
+ revisions 382107 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-02-26 19:19 +0000 [r382096] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * /, main/Makefile: Remove unneeded linux-gnueabi* As of r380521
+ the configure scripts converts the value of linux-gnueabi* of
+ OSARCH to "linux-gnu". So no point in testing for those values.
+ ........ Merged revisions 382087 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-02-26 15:38 +0000 [r382066-382069] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_confbridge.c: Fix typo in r382068 Well, that was
+ embarrassing. Removed an '-l' that somehow got in there.
+
+ * apps/app_confbridge.c: Clean up ConfBridge commands to account
+ for wait_marked users When ConfBridge was refactored to better
+ handle the concept of marked, wait_marked, and normal users
+ co-existing in a conference (thereby implementing a state machine
+ for the conference), the wait_marked users were put into their
+ own list of conference participants, separate from the active
+ users. This list is used for wait_marked users when they are
+ waiting in a conference but no marked user has joined; normal
+ users may have joined at this point however. There are several
+ AMI/CLI commands that affect conference users that were not
+ checking the wait_marked users list: * CLI/AMI commands that
+ mute/unmute a participant. In this case, wait_marked users have
+ to remain in their particular state and should not be affected -
+ however, the commands would return "Channel not found" as opposed
+ to the appropriate error condition. * CLI/AMI commands that kick
+ a participant. An admin should always be able to kick a
+ participant out of the conference. This patch fixes both sets of
+ commands, and cleans up the CLI commands slightly by allowing
+ them to complete a participant name (this was supposed to have
+ been added, but the function call was commented out and wasn't
+ implemented). Review: https://reviewboard.asterisk.org/r/2346/
+ (closes issue AST-1114) Reported by: John Bigelow Tested by: John
+ Bigelow
+
+ * apps/confbridge/conf_config_parser.c,
+ configs/confbridge.conf.sample: Ensure that the default
+ bridge/user profiles are always available ConfBridge and Page
+ require that there always be a default bridge and user profile
+ available. While properties of the default profiles can be
+ overriden in the configuration file, removing them can create
+ situations where neither application can function properly. This
+ patch ensures that if an administrator removes the profiles from
+ the confbridge.conf configuration file, the profiles are added
+ upon load. Documentation clarifying this has been added to the
+ confbridge.conf.sample file. Review:
+ https://reviewboard.asterisk.org/r/2356/ (closes issue AST-1115)
+ Reported by: John Bigelow Tested by: John Bigelow
+
+2013-02-25 12:50 +0000 [r381917-382022] Matthew Jordan <mjordan@digium.com>
+
+ * addons/res_config_mysql.c, /: Clean up use of va_end/va_args in
+ res_config_mysql There were several problems using variadic
+ argument macros in res_config_mysql. * Improper use of va_end.
+ Multiple calls to va_end were possible resulting in an unbalanced
+ matching of va_start/va_end. * Calls to va_arg after a possible
+ encounter of a SENTINEL value. This patch corrects those errors.
+ (closes issue ASTERISK-19451) Reported by: wdoekes patches:
+ ASTERISK-19451-1.8--2.diff uploaded by wdoekes (License 5674)
+ ........ Merged revisions 382021 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_jingle.c, /: Set the sin_family on the bind address
+ socket during initialization Somehow, chan_jingle has managed to
+ operate for years without setting the sin_family on its bindaddr
+ socket. This patch properly sets the field during initial module
+ load to AF_INET. Note that the patch on the issue was modified
+ slightly to change the initialization of the socket from
+ allocation of a chan_jingle private to the module initialization,
+ as the bindaddr object (which is static) only needs to have the
+ address set once. (closes issue ASTERISK-19341) Reported by:
+ andre valentin patches: 0105-chan_jingle.patch uploaded by
+ avalentin (License 6064) ........ Merged revisions 381975 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/manager.c, /: Don't display the AMI ALL class authorization
+ for users if they don't have it When converting AMI class
+ authorizations to a string representation, the method always
+ appends the ALL class authorization. This is especially important
+ for events, as they should always communicate that class
+ authorization - even if the event itself does not specify ALL as
+ a class authorization for itself. (Events have always assumed
+ that the ALL class authorization is implied when they are raised)
+ Unfortunately, this did mean that specifying a user with
+ restricted class authorizations would show up in the 'manager
+ show user' CLI command as having the ALL class authorization.
+ Rather then modifying the existing string manipulation function,
+ this patch adds a function that will only return a string if the
+ field being compared explicitly matches class authorization field
+ it is being compared against. This prevents ALL from being
+ returned unless it is actually specified for the user. (closes
+ issue ASTERISK-20397) Reported by: Johan Wilfer ........ Merged
+ revisions 381939 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_parkandannounce.c, /: Make ParkAndAnnounce return to
+ priority + 1 when return context is not defined The
+ ParkAndAnnounce application documentation for the optional
+ return_context parameter states the following: return_context The
+ goto-style label to jump the call back into after timeout.
+ Default 'priority+1'. Unfortunately, the application was sending
+ the channel back into the dialplan at 'priority', which is the
+ ParkAndAnnounce application call. This causes an infinite loop of
+ the channel constantly being parked, announced, timed out,
+ parked, announced, timed out... while fun, especially for those
+ callers you wish to drive to the end of madness, this was not the
+ intent of the application. (closes issue ASTERISK-20113) Reported
+ by: serginuez patches: app_parkandannounce.diff uploaded by
+ serginuez (License 6405) ........ Merged revisions 381916 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-02-22 19:38 +0000 [r381893] Michael L. Young <elgueromexicano@gmail.com>
+
+ * res/res_agi.c: Fix FastAGI To Properly Check For A Connection
+ When IPv6 support was added to FastAGI, the intent was to have
+ the ability to check all addresses resolved for a host since we
+ might receive an IPv4 address and an IPv6 address. The problem
+ with the current code, is that, since we are doing O_NONBLOCK, we
+ get EINPROGRESS when calling ast_connect() but are ignoring this
+ instead of handling it. We break out of the loop and continue on.
+ When we later call ast_poll(), it succeeds but we never check if
+ we have a connection or not on the socket level. We then attempt
+ to send data to the host address that we think is setup and it
+ fails. We then check the errno and see that we have "connection
+ refused" and then return with agi failed. This patch does the
+ following: * Handles EINPROGRESS by creating the function
+ handle_connection() - ast_poll() was moved into this function -
+ This function checks the results of the connection on the socket
+ level after calling ast_poll() * Continues to the next address if
+ the above fails to create a connection * Once all addresses
+ resolved are tried and we still are unable to establish a
+ connection, then we return that the FastAGI call failed (closes
+ issue ASTERISK-21065) Reported by: Jeremy Kister Tested by:
+ Jeremy Kister, Michael L. Young Patches:
+ asterisk-21065_poll_correctly_v4.diff Michael L. Young (license
+ 5026) Review: https://reviewboard.asterisk.org/r/2330/
+
+2013-02-22 15:41 +0000 [r381880] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_dial.c: app_dial: Honor the 'c' flag when the calling
+ party hangs up Apparently this feature became broken in 11,
+ probably as a result of the Hangup Cause project. (closes issue
+ ASTERISK-21113) Reprted by: Heiko Wundram Patches: app_dial.patch
+ uploaded by Heiko Wundram (license 5822)
+
+2013-02-21 22:48 +0000 [r381848] Matthew Jordan <mjordan@digium.com>
+
+ * /, configure, configure.ac: Properly detect launchd Asterisk was
+ a little too pro-active in claiming that it found launchd. On
+ systems without launchd - such as FreeBSD - this resulted in
+ certain items in Asterisk that conflict with launchd to not be
+ selectable, such as res_timing_kqueue. (closes issue
+ ASTERISK-20749) Reported by: Oleg Baranov ........ Merged
+ revisions 381847 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-02-20 19:14 +0000 [r381835] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_voicemail.c: Let vm_mailbox_snapshot_create's combine
+ option apply to "Urgent" as well The vm_mailbox_snapshot_create
+ function has an option that combines the contents of INBOX and
+ Old into a single snapshot. The intent of this is that both 'new'
+ messages and 'deleted' messages are given in a single snapshot,
+ as some applications prefer this view of the voicemail world.
+ Unfortunately, the initial implementation ignored the "Urgent"
+ folder. The "Urgent" folder is a pseudo-INBOX, in that new
+ messages left with the 'U' flag will be placed in that folder as
+ opposed to INBOX. Thus, the option failed the intent with which
+ it was added. This patch makes it so that the "Urgent" folder is
+ included in the snapshot when that option is used.
+
+2013-02-19 19:44 +0000 [r381702-381791] kharwell <kharwell@localhost>:
+
+ * /, main/features.c: Write the correct callid to the data1 field
+ in queue_log for transfer events. The incorrect callid was being
+ written to the "data1" field in queue_log table for transfer
+ events. The callid of the queue was being written instead of the
+ transfer target's callid. This now gets the correct "transfer to"
+ number and places that in the "data1" field of the queue_log
+ table when a transfer event is triggered. (closes issue
+ ASTERISK-19960) Reported by: vladimir shmagin ........ Merged
+ revisions 381770 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_confbridge.c: Confbridge channels staying active when
+ all participants leave. If you started/stopped recording of a
+ conference multiple times channels would remain active even when
+ all participants left the conference. This was due to the fact
+ that a reference to the confbridge was being added every time a
+ start record command was issued, but when the recording was
+ stopped there was no matching de-reference thus keeping the
+ conference alive. Made sure only a single reference is added for
+ the record thread no matter how many times recording is
+ started/stopped. A de-reference is issued upon thread ending.
+ Note, this issue is being fixed under AST-1088 since it relates
+ to it and should have been corrected along with those
+ modifications. (issue AST-1088) Reported by: John Bigelow
+
+ * apps/app_confbridge.c: Fixed Confbridge file recording deadlock
+ and appending. A deadlock occurred after starting/stopping and
+ then restarting a confbridge recording. Upon starting a recording
+ a record thread is created that holds a lock until just before
+ exiting. Stopping the recording does not stop/exit the thread or
+ release the lock. The thread waits until recording begins again.
+ Starting a stopped recording signals the thread to continue and
+ start recording again. However restarting the recording also
+ created another record thread resulting in a deadlock. The fix
+ was to make sure the record thread was only created once. Also it
+ was noted that filenames for the recordings were being
+ concatenated for each start/stop. This was fixed by creating a
+ new file for each conference session and appending the actual
+ recorded data within the file (e.g. passing the 'a' option to
+ MixMonitor). (issue AST-1088) Reported by: John Bigelow Review:
+ http://reviewboard.digium.internal/r/374/
+
+2013-02-18 20:30 +0000 [r381669] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, configs/sip.conf.sample: Remove "registertrying" and add
+ "rtp_engine" from/to sip.conf.sample The "registertrying" option
+ was removed in r343220. The "rtp_engine" option was added in
+ r186078 but erroneously named "engine" in the sample. Note that
+ there is no global sip setting for a different engine. ........
+ Merged revisions 381668 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-02-18 19:43 +0000 [r381655] Jonathan Rose <jrose@digium.com>
+
+ * funcs/func_presencestate.c: PRESENCE_STATE: Provide better
+ documentation for the 'e' option. Notes that the 'e' option
+ actually decodes data when used as a write function such as with
+ the SET application while it encodes data when used to read.
+ Review: https://reviewboard.asterisk.org/r/2335/
+
+2013-02-16 16:22 +0000 [r381594-381613] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_sip.c: Don't send presencestate information if the
+ state is invalid Previously, presencestate information was sent
+ whenever the state was not NOT_SET. When r381594 actually
+ returned INVALID presence state in all the places it was supposed
+ to, it caused chan_sip to start adding presence state information
+ to NOTIFY requests that it previously would not have added.
+ chan_sip shouldn't be adding presence state information when the
+ provider is in an invalid state; users can't set the state to
+ invalid and an invalid state always implies that the provider is
+ in an error condition. (issue AST-1084)
+
+ * main/presencestate.c, funcs/func_presencestate.c, main/manager.c:
+ Fix crash in PresenceState AMI action when specifying an invalid
+ provider This patch fixes a crash in Asterisk that could be
+ caused by using the PresenceState AMI action while providing an
+ invalid provider. This patch also adds some additional warnings
+ when a user attempts to provide the PresenceState action with
+ invalid data, and removes some NOTICE statements that were still
+ lurking in the code from testing. (closes issue AST-1084)
+ Reported by: John Bigelow Tested by: John Bigelow
+
+2013-02-15 18:42 +0000 [r381566] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix a crash that occurred when a BYE was
+ received on a replaced dialog. Reference counting for the channel
+ and its tech_pvt got messed up at some point between 1.8 and 11.
+ The result was that if a BYE for a dialog that had been replaced
+ (via an INVITE with Replaces) was received, Asterisk would crash
+ due to trying to access data on a channel that was no longer
+ there. The fix I introduced is to remove code that both unrefs
+ the sip_pvt and sets the channel's tech_pvt to NULL when an
+ INVITE with Replaces is handled. This way when a BYE is received,
+ the tech_pvt will be non-NULL and so the BYE can be processed and
+ not cause a crash. (closes issue ASTERISK-20929) reported by
+ Kristopher Lalletti patches: ASTERISK-20929.patch uploaded by
+ Mark Michelson (License #5049)
+
+2013-02-15 17:17 +0000 [r381554] kharwell <kharwell@localhost>:
+
+ * include/asterisk/logger.h, main/autoservice.c, main/logger.c:
+ Stopped spamming of debug messages during attended transfer.
+ While autoservice is running and servicing a channel the callid
+ is being stored and removed in the thread's local storage for
+ each iteration of the thread loop. If debug was set to a
+ sufficient level the log file would be spammed with callid thread
+ local storage debug messages. Added a new function that checks to
+ see if the callid to be stored is different than what is already
+ contained (if anything). If it is different then store/replace
+ and log, otherwise just leave as is. Also made it so all logging
+ of debug messages pertaining to the callid thread storage outputs
+ only when TEST_FRAMEWORK is defined. (issue ASTERISK-21014)
+ (closes issue ASTERISK-21014) Report by: Rusty Newton Review:
+ https://reviewboard.asterisk.org/r/2324/
+
+2013-02-15 17:12 +0000 [r381553] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c: chan_sip: Use video and text crypto
+ attributes to append RTP profiles to SDP Some bad copy/pasting
+ resulted in using the audio crypto attribute for both text and
+ video RTP. Also the audio crypto isn't set until after these, so
+ it was really just bad all around. (closes ASTERISK-20905)
+ Reported by: Kristopher Lalletti patches:
+ rtp_crypto_video_text.diff uploaded by Jonathan Rose (license
+ 6182)
+
+2013-02-14 19:44 +0000 [r381467] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: End stuck DTMF if AST_SOFTHANGUP_ASYNCGOTO
+ because it isn't a real hangup. It doesn't hurt to check
+ AST_SOFTHANGUP_UNBRIDGE either, but it should not be set outside
+ of a bridge. (issue ASTERISK-20492) ........ Merged revisions
+ 381466 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-02-14 03:48 +0000 [r381365] Matthew Jordan <mjordan@digium.com>
+
+ * /, apps/app_db.c: Don't throw a spurious error when using
+ DBdeltree The function call ast_db_deltree returns the number of
+ row deleted, or a negative number if it failed. DBdeltree was
+ treating any non-zero return as an error, causing a spurious
+ verbose error message to be displayed. This patch handles the
+ return code of ast_db_deltree correctly. (closes issue
+ ASTERISK-21070) Reported by: ianc patches: dbdeltree.diff
+ uploaded by ianc (License #5955) ........ Merged revisions 381364
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-02-12 20:31 +0000 [r381306] Mark Michelson <mmichelson@digium.com>
+
+ * main/rtp_engine.c, /: Do not allow native RTP bridging if
+ packetization of media streams differs. The RTP engine will no
+ longer allow for local and remote native RTP bridges if
+ packetization of streams differs. Allowing native bridging in
+ this scenario has been known to cause FAX failures. (closes
+ ASTERISK-20650) Reported by: Maciej Krajewski Patches:
+ ASTERISK-20659.patch uploaded by Mark Michelson (License #5049)
+ Review: https://reviewboard.asterisk.org/r/2319 ........ Merged
+ revisions 381281 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-02-12 20:16 +0000 [r381282] Kinsey Moore <kmoore@digium.com>
+
+ * channels/sip/include/sip.h, channels/chan_sip.c,
+ channels/sip/security_events.c: Fix some more REF_DEBUG-related
+ build errors When sip_ref_peer and sip_unref_peer were exported
+ to be usable in channels/sip/security_events.c, modifications to
+ those functions when building under REF_DEBUG were not taken into
+ account. This change moves the necessary defines into sip.h to
+ make them accessible to other parts of chan_sip that need them.
+
+2013-02-11 20:55 +0000 [r381217] kharwell <kharwell@localhost>:
+
+ * apps/app_playback.c, /: Properly load say.conf upon reload of
+ module app_playback. If say.conf did not exists prior to
+ originally loading module app_playback it would not load on
+ subsequent reloads of the module once it had been created. This
+ occurred because upon reload of the app_playback module it would
+ only load a new configuration if an old one had previously
+ existed. This fix simply removed the association between checking
+ if an old configuration existed and the loading of the new one.
+ (closes issue ASTERISK-20800) Reported by: pgoergler ........
+ Merged revisions 381216 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-02-11 15:03 +0000 [r381159] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_xmpp.c: Fix crash in res_xmpp when deleting pubsub node
+ from CLI An error existed in res_xmpp where it would attempt to
+ delete attributes from a node that itself was also deleted. Per
+ the iksemel documentation, attributes added using iks_insert are
+ copied to the parent node's stack, and will be reclaimed when
+ that node is itself destroyed. (closes issue ASTERISK-20982)
+ Reported by: marcelloceschia patches: delete-node-fix.diff
+ uploaded by marcelloceschia (License 6036)
+
+2013-02-08 17:29 +0000 [r381067] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_confbridge.c: app_confbridge: Fix crash from receiving
+ an AMI action after ConfBridge unloaded. Unloading ConfBridge
+ caused the next AMI action received to crash Asterisk. * Add the
+ missing unregister of AMI action ConfbridgeSetSingleVideoSrc when
+ ConfBridge is unloaded. (closes issue ASTERISK-20994) Reported
+ by: Jeremy Kister Patches: jira_asterisk_20994_v11.patch (license
+ #5621) patch uploaded by rmudgett Tested by: Rusty Newton, Jeremy
+ Kister
+
+2013-02-06 20:14 +0000 [r380974] David M. Lee <dlee@digium.com>
+
+ * /, channels/chan_sip.c: Fixed failing test from r380696. When I
+ added my extensive suite of session timer unit tests, apparently
+ one of them was failing and I never noticed. If neither Min-SE
+ nor Session-Expires is set in the header, it was responding with
+ a Session-Expires of the global maxmimum instead of the
+ configured max for the endpoint. (issue ASTERISK-20787) ........
+ Merged revisions 380973 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-02-06 08:42 +0000 [r380926-380942] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Fix reload skinny with active devices.
+ Patch ensures that d->activeline and l->activesub are moved over
+ to the new device and line so that on callend the appropriate
+ subs can be found to complete hangup before device resets.
+ (closes issue ASTERISK-16610) Reported by: wedhorn Tested by:
+ snuffy, myself Patches: skinny-reloadactive01.diff uploaded by
+ wedhorn (license 5019)
+
+ * channels/chan_skinny.c: Reset skinny vmexten on reload. Make
+ skinny reset vmexten '\0' on reload to ensure that it is set to
+ '\0' if the appropriate item is removed/commented in skinny.conf.
+ part of ASTERISK-21037 Reported by: snuffy Tested by: snuffy,
+ myself Patches: part of immed_dial_fix.diff uploaded by snuffy
+ (license 5024)
+
+2013-02-05 19:09 +0000 [r380854-380894] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_page.c, apps/app_confbridge.c: app_page and
+ app_confbridge: Fix custom announcement on entering conference.
+ The Page and ConfBridge custom announcement did not play when
+ users entered the conference. * Fix the
+ CONFBRIDGE(user,announcement) file not getting played. The code
+ to do this got removed accidentally when the ConfBridge code was
+ restructured to be more state machine like. * Fixed
+ play_prompt_to_user() doxygen comments. * Fixed the Page A(x) and
+ n options for the caller. The caller never played the
+ announcement file and totally ignored the n option. The code to
+ do this was lost when the application was converted to use
+ ConfBridge. * Factored out setup_profile_bridge(),
+ setup_profile_paged(), and setup_profile_caller() routines to
+ setup ConfBridge profiles. Made each profile setup routine use
+ the default template if one has not already been setup by
+ dialplan. (closes issue ASTERISK-20990) Reported by: Jeremy
+ Kister Tested by: rmudgett
+
+ * apps/confbridge/conf_state_multi_marked.c: app_confbridge: Fix
+ error messages on exiting conference. A marked user ending a
+ conference with only end_marked users generates error messages:
+ ERROR[0000][C-00000000]: confbridge/conf_state.c:47
+ conf_invalid_event_fn: Invalid event for confbridge user '' * The
+ MULTI_MARKED state was doing too much when it was kicking out the
+ end_marked users from the conference. The kicked out users will
+ clean up after themselves when they exit the conference. (closes
+ issue ASTERISK-20991) Reported by: Jeremy Kister Tested by:
+ rmudgett
+
+ * apps/app_page.c: app_page: Fixup application XML documentation
+ typos and inaccuracies.
+
+ * apps/confbridge/conf_config_parser.c: Because the compiler can
+ check types with a struct copy and memcpy() cannot.
+
+ * main/dial.c, /: Separate option_types[] from the struct
+ definition. Updated the option_types[] doxygen comment. ........
+ Merged revisions 380853 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-02-04 19:50 +0000 [r380816] Jason Parker <jparker@digium.com>
+
+ * res/pjproject/aconfigure, res/pjproject/build/os-auto.mak.in,
+ Makefile, res/pjproject/aconfigure.ac, res/Makefile,
+ res/pjproject/build/common.mak: Fix how we build pjproject. Allow
+ parallel builds, better tolerate failures, build faster. This
+ also stops running dependencies before top-level configure has
+ been run. (closes issue ASTERISK-20815) Review:
+ https://reviewboard.asterisk.org/r/2292/
+
+2013-01-31 21:42 +0000 [r380735-380736] Jason Parker <jparker@digium.com>
+
+ * res/pjproject/pjlib/include/pj/config_site.h: Ignore warnings
+ caused by PJ_TODO()s in pjproject.
+
+ * res/pjproject/pjlib/src/pj/ssl_sock_ossl.c,
+ res/pjproject/pjlib/src/pj/log.c,
+ res/pjproject/pjlib/src/pj/pool_buf.c,
+ res/pjproject/pjsip-apps/src/samples/icedemo.c,
+ res/pjproject/pjmedia/src/test/test.c: Fix a few compiler
+ warnings.
+
+2013-01-31 20:10 +0000 [r380698] David M. Lee <dlee@digium.com>
+
+ * /, channels/chan_sip.c: Process session timers, even if
+ Session-Expires header is missing Previously, Asterisk only
+ processed session timer information if both the 'Supported:
+ timer' and 'Session-Expires' headers were present. However, the
+ Session-Expires header is optional. If we were to receive a
+ request with a Min-SE greater than our configured
+ session-expires, we would respond with a 'Session-Expires' header
+ that was too small. This patch cleans the situation up a bit,
+ always processing timer information if the 'Supported: timer'
+ header is present. (closes issue ASTERISK-20787) Reported by:
+ Mark Michelson Review: https://reviewboard.asterisk.org/r/2299/
+ ........ Merged revisions 380696 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-31 19:03 +0000 [r380671-380673] Jason Parker <jparker@digium.com>
+
+ * res/pjproject/pjsip/build/Makefile,
+ res/pjproject/pjsip-apps/build/Makefile,
+ res/pjproject/pjmedia/build/Makefile,
+ res/pjproject/pjlib-util/build/Makefile,
+ res/pjproject/pjlib/build/Makefile,
+ res/pjproject/pjnath/build/Makefile: Add support for parallel
+ builds of pjproject. Also adds proper dependency checking, and
+ direct .a file targets. We don't take advantage of this
+ currently, but we will soon. (issue ASTERISK-20815)
+
+ * res/pjproject/aconfigure, res/pjproject/aconfigure.ac: Always
+ check for libm, regardless of configure options.
+
+ * res/pjproject/aconfigure, res/pjproject/aconfigure.ac,
+ res/pjproject/build/cc-auto.mak.in,
+ res/pjproject/build/rules.mak: Remove a cross-compile workaround.
+ ar and ranlib can be easily detected with autoconf.
+
+2013-01-31 00:30 +0000 [r380575-380612] Richard Mudgett <rmudgett@digium.com>
+
+ * /, include/asterisk/channel.h: Make CHECK_BLOCKING() debug
+ message more useful. Change the displayed pthread value to hex
+ format so it can be easily matched with CLI core show threads or
+ gdb. ........ Merged revisions 380611 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_dahdi.c, /: chan_dahdi: Fix "dahdi show channels
+ group" for groups greater than 31. The variable type used was not
+ large enough to hold a group bit field. ........ Merged revisions
+ 380572 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-27 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 11.3.0-rc2 Released.
+
+ * app_confbridge: Fix error messages on exiting conference.
+
+ A marked user ending a conference with only end_marked users
+ generates error messages:
+ ERROR[0000][C-00000000]: confbridge/conf_state.c:47
+ conf_invalid_event_fn: Invalid event for confbridge user ''
+
+ The MULTI_MARKED state was doing too much when it was kicking out
+ the end_marked users from the conference. The kicked out users
+ will clean up after themselves when they exit the conference.
+
+ * app_page and app_confbridge: Fix custom announcement on entering
+ conference.
+
+ The Page and ConfBridge custom announcement did not play when users
+ entered the conference.
+
+ Fix the CONFBRIDGE(user,announcement) file not getting played. The
+ code to do this got removed accidentally when the ConfBridge code
+ was restructured to be more state machine like.
+
+ Fixed play_prompt_to_user() doxygen comments.
+
+ Fixed the Page A(x) and n options for the caller. The caller never
+ played the announcement file and totally ignored the n option. The
+ code to do this was lost when the application was converted to use
+ ConfBridge.
+
+ Factored out setup_profile_bridge(), setup_profile_paged(), and
+ setup_profile_caller() routines to setup ConfBridge profiles. Made
+ each profile setup routine use the default template if one has not
+ already been setup by dialplan.
+
+ * app_confbridge: Fix crash from receiving an AMI action after
+ ConfBridge unloaded.
+
+ Unloading ConfBridge caused the next AMI action received to crash
+ Asterisk. Add the missing unregister of AMI action
+ ConfbridgeSetSingleVideoSrc when ConfBridge is unloaded.
+
+ * Fixed Confbridge file recording deadlock and appending.
+
+ A deadlock occurred after starting/stopping and then restarting a
+ confbridge recording. Upon starting a recording a record thread is
+ created that holds a lock until just before exiting. Stopping the
+ recording does not stop/exit the thread or release the lock. The
+ thread waits until recording begins again. Starting a stopped
+ recording signals the thread to continue and start recording
+ again. However restarting the recording also created another
+ record thread resulting in a deadlock. The fix was to make sure
+ the record thread was only created once.
+
+ * Confbridge channels staying active when all participants leave.
+
+ If you started/stopped recording of a conference multiple times
+ channels would remain active even when all participants left the
+ conference. This was due to the fact that a reference to the
+ confbridge was being added every time a start record command was
+ issued, but when the recording was stopped there was no matching
+ de-reference thus keeping the conference alive. Made sure only a
+ single reference is added for the record thread no matter how
+ many times recording is started/stopped. A de-reference is
+ issued upon thread ending.
+
+ * Let vm_mailbox_snapshot_create's combine option apply to "Urgent"
+ as well
+
+ The vm_mailbox_snapshot_create function has an option that combines
+ the contents of INBOX and Old into a single snapshot. The intent
+ of this is that both 'new' messages and 'deleted' messages are given
+ in a single snapshot, as some applications prefer this view of the
+ voicemail world. Unfortunately, the initial implementation ignored the
+ "Urgent" folder. The "Urgent" folder is a pseudo-INBOX, in that new
+ messages left with the 'U' flag will be placed in that folder as
+ opposed to INBOX. Thus, the option failed the intent with which it
+ was added.
+
+ * Fix comparison of presence state in event subsystem.
+
+ Several new IEs were not given types (or names), causing the
+ comparison function to improperly succeed. This adds those.
+
+ * Let vm_mailbox_snapshot combine "Urgent" when no folder is specified
+
+ r381835 fixed a bug in vm_mailbox_snapshot where combining INBOX and
+ Old forgot that Urgent also "counts" as new messages. This fixed the
+ problem when any of the three folders was specified and the combine
+ option was used. It missed the case where the folder isn't specified
+ and we build a snapshot of all folders. This patch corrects that.
+
+ * Do not allow native RTP bridging if packetization of media streams
+ differs.
+
+ The RTP engine will no longer allow for local and remote native RTP
+ bridges if packetization of streams differs. Allowing native bridging
+ in this scenario has been known to cause FAX failures.
+
+ * Resolve deadlock between pending CDR and batch CDR locks
+
+ r375757 attempted to resolve a race condition between multiple
+ submissions of CDRs while in batch mode from attempting to destroy the
+ scheduled batch submission by extending the batch CDR lock. Unfortunately,
+ this causes a deadlock between the pending CDR lock and the batch CDR lock.
+ This patch resolves the intent of r375757 by simply providing a new lock
+ that protects the scheduling of the batches. The original batch CDR lock
+ is kept to protect manipulation of the batch CDR settings, but has been
+ placed such that it is not held when the pending lock is held.
+
+ Thanks to Chase Venters for providing lock analysis on the issue.
+
+ * Resolve deadlock between SIP registration and channel based
+ functions
+
+ In r373424, several reentrancy problems in chan_sip were addressed. As
+ a result, the SIP channel driver is now properly locking the channel
+ driver private information in certain operations that it wasn't previously.
+ This exposed two latent problems either in register_verify or by functions
+ called by register_verify. This includes:
+ * Holding the private lock while calling sip_send_mwi_to_peer. This
+ can create a new sip_pvt via sip_alloc, which will obtain the channel
+ container lock. This is a locking inversion, as any channel related lock
+ must be obtained prior to obtaining the SIP channel technology private
+ lock.
+ * Holding the private lock while calling sip_poke_peer. In the same vein as
+ sip_send_mwi_to_peer, sip_poke_peer can create a new SIP private, causing
+ the same locking inversion.
+
+ Note that this locking inversion typically occured when CLI commands were run
+ while a SIP REGISTER request was being processed, as many CLI commands (such
+ as 'sip show channels', 'core show channels', etc.) have to obtain the channel
+ container lock.
+
+ * AST-2013-001: Prevent buffer overflow through H.264 format negotiation
+
+ The format attribute resource for H.264 video performs an unsafe read
+ against a media attribute when parsing the SDP. The value passed in with
+ the format attribute is not checked for its length when parsed into a fixed
+ length buffer. This patch resolves the vulnerability by only reading
+ as many characters from the SDP value as will fit into the buffer.
+
+ * AST-2013-002: Prevent denial of service in HTTP server
+
+ AST-2012-014, fixed in January of this year, contained a fix for
+ Asterisk's HTTP server for a remotely-triggered crash. While the fix put in
+ place fixed the possibility for the crash to be triggered, a denial of
+ service vector still exists with that solution if an attacker sends one or
+ more HTTP POST requests with very large Content-Length values. This patch
+ resolves this by capping the Content-Length at 1024 bytes. Any attempt to send
+ an HTTP POST with Content-Length greater than this cap will not result in any
+ memory allocation. The POST will be responded to with an HTTP 413 "Request
+ Entity Too Large" response.
+
+ This issue was reported by Christoph Hebeisen of TELUS Security Labs
+
+ * AST-2013-003: Prevent username disclosure in SIP channel driver
+
+ When authenticating a SIP request with alwaysauthreject enabled,
+ allowguest disabled, and autocreatepeer disabled, Asterisk discloses whether
+ a user exists for INVITE, SUBSCRIBE, and REGISTER transactions in
+ multiple ways. The information is disclosed when:
+ * A "407 Proxy Authentication Required" response is sent instead of a
+ "401 Unauthorized" response
+ * The presence or absence of additional tags occurs at the end of
+ "403 Forbidden" (such as "(Bad Auth)")
+ * A "401 Unauthorized" response is sent instead of "403 Forbidden"
+ response after a retransmission
+ * Retransmission are sent when a matching peer did not exist, but not
+ when a matching peer did exist.
+ This patch resolves these various vectors by ensuring that the responses sent
+ in all scenarios is the same, regardless of the presence of a matching peer.
+
+ This issue was reported by Walter Doekes, OSSO B.V. A substantial portion of
+ the testing and the solution to this problem was done by Walter as well - a
+ huge thanks to his tireless efforts in finding all the ways in which this
+ setting didn't work, providing automated tests, and working with Kinsey on
+ getting this fixed.
+
+ * Fix white noise on SRTP decryption
+
+ When res_rtp_asterisk.c was altered to avoid attempting to apply
+ unprotect algorithms to non-audio RTP packets, the test used was
+ incorrect. This caused the audio packets to not be decrypted and
+ resulted in loud white noise on the other endpoint (or both endpoints
+ depending on the call legs involved). The test now properly checks the
+ version field in the RTP header to ensure that RTP and RTCP are
+ decrypted while other types of packets are not.
+
+2013-01-30 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 11.3.0-rc1 Released.
+
+2013-01-30 17:46 +0000 [r380452-380521] Matthew Jordan <mjordan@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Support building Asterisk for Raspberry Pi/Raspbian with
+ hard-float support Building Asterisk on Raspbian with hard-float
+ support fails as it uses the string 'linux-gnueabihf' for host
+ os, as opposed to 'linux-gnueabi'. This patch modifies the
+ configure script for Asterisk such that it will match on any
+ string beginning with 'linux-gnueabi', as opposed to requiring an
+ explicit match. (closes issue ASTERISK-21006) Reported by:
+ Christian Hesse Tested by: Christian Hesse patches:
+ linux-gnueabihf.patch uploaded by Christian Hesse (license 6459)
+ linux-gnueabihf-autoconf.patch uploaded by Christian Hesse
+ (license 6459) ........ Merged revisions 380520 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_sip.c: Unregister SIP provider API if module load
+ is declined A user in #asterisk ran into a problem where a
+ configuration error prevented the chan_sip module from being
+ loaded. Upon fixing their configuratione error, they could no
+ longer load the chan_sip module. This was because the
+ configuration checking happened after the SIP provider was
+ registered with the Asterisk core, and subsequent attempts to
+ load the SIP module failed as the provider was already
+ registered. Since we want to detect any failure in registering
+ chan_sip as early as possible (as that could be emblematic of a
+ deeper mismatch between module and Asterisk core), this patch
+ does not change the registration location, but does ensure that
+ if a module load is declined, we unregister the module as the SIP
+ api provider.
+
+ * /, channels/chan_sip.c: Perform case insensitive comparisons for
+ T.38 attributes RFC5347 section 2.5.2 states the following: ...
+ The attribute "T38MaxBitRate" was once incorrectly registered
+ with IANA as "T38maxBitRate" (lower-case "m"). In accordance with
+ T.38 examples and common implementation practice, the form
+ "T38MaxBitRate" SHOULD be generated by implementations conforming
+ to this package. In general, it is RECOMMENDED that
+ implementations of this package accept lowercase, uppercase, and
+ mixed upper/lowercase encodings of all the T.38 attributes. ...
+ Asterisk currently does not perform case insensitive matching on
+ the T.38 attributes. This causes the T38MaxBitRate attribute to
+ be negotiated at 2400 baud instead of 14400 (or whatever value
+ you actually wanted). This patch makes it so that when we compare
+ T.38 attributes, we do so in a case insensitive fashion. Note
+ that while the issue reporter did not directly write the patch,
+ they contributed to it (and would have provided one themselves if
+ the license had gone through a tad faster), and hence get
+ attribution for it. Review:
+ https://reviewboard.asterisk.org/r/2298/ (closes issue
+ ASTERISK-20897) Reported by: Eric Hill Tested by: Eric Hill
+ patches: -- uploaded by Eric Hill ........ Merged revisions
+ 380458 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * res/res_calendar_icalendar.c, /: Fix memory leak in
+ res_calendar_icalendar The ICalendar module had a systemic memory
+ leak on each fetch of data from the ICalendar source. The
+ previous fetched data was not being properly disposed. This patch
+ makes it so that before each fetch of data, we dispose of the
+ previously fetched data. (closes issue ASTERISK-21012) Reported
+ by: Joel Vandal Tested by: Joel Vandal ........ Merged revisions
+ 380451 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-29 17:54 +0000 [r380384] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_agent.c: chan_agent: Prevent multiple channels
+ from logging in as the same agent. Multiple channels logging in
+ as the same agent can result in dead channels waiting for a
+ condition signal that will never come because another channel
+ thread stole it. A symptom is chan_sip repeatedly generating
+ warning messages about rescheduling autodestruction of dialogs
+ with an agent channel owner. * Made only login_exec() (the app
+ AgentLogin) clear the agent_pvt->chan pointer to prevent multiple
+ channels from logging in as the same agent. agent_read(),
+ agent_call(), and agent_set_base_channel() no longer disconnect
+ the agent channel from the agent_pvt. This also eliminates the
+ need to keep checking for agent_pvt->chan being NULL. * Made
+ agent_hangup() not wake up the AgentLogin agent thread until it
+ is done. * Made agent_request() not able to get the agent until
+ he has logged in and any wrapup time has expired. * Made
+ agent_request() use ast_hangup() instead of agent_hangup() to
+ correctly dispose of a channel. * Removed
+ agent_set_base_channel(). Nobody calls it and it is a bad thing
+ in general. * Made only agent_devicestate() determine the current
+ device state of an agent. Note: Agent group device states have
+ never been supported. Review:
+ https://reviewboard.asterisk.org/r/2260/ ........ Merged
+ revisions 380364 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-29 17:14 +0000 [r380350] David M. Lee <dlee@digium.com>
+
+ * channels/sip/sdp_crypto.c, /: Corrected crypto tag in SDP ANSWER
+ for SRTP. (again) The original fix (r380043) for getting Asterisk
+ to respond with the correct tag overlooked some corner cases, and
+ the fact that the same code is in 1.8. This patch moves the
+ building of the crypto line out of sdp_crypto_process(). Instead,
+ it merely copies the accepted tag. The call to sdp_crypto_offer()
+ will build the crypto line in all cases now, using a tag of "1"
+ in the case of sending offers. (closes issue ASTERISK-20849)
+ Reported by: José Luis Millán Review:
+ https://reviewboard.asterisk.org/r/2295/ ........ Merged
+ revisions 380347 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-29 17:05 +0000 [r380348] Jonathan Rose <jrose@digium.com>
+
+ * main/features.c: call_parking: Make sure fallbacks are used when
+ lacking a flat channel exten A regression was introduced which
+ removed automatic fallback behavior from the PBX. This behavior
+ was used by call parking (or at least documented as how the
+ feature works) in order to select an extension when the flat
+ channel extension wasn't available from the comebackcontext.
+ Parking now handles the fallbacks internally in order to keep
+ behavior matching with how it is documented. (closes issue
+ ASTERISK-20716) Reported by: Chris Gentle Review:
+ https://reviewboard.asterisk.org/r/2296/
+
+2013-01-29 14:45 +0000 [r380298-380331] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_sip.c: Ensure that a declined media stream is
+ terminated with a '\r\n' In r369028, chan_sip's processing of
+ media streams in an SDP was modified to better handle multiple
+ offered media streams. Part of that change modified how streams
+ were declined. Previously, declined media streams were not
+ handled in an RFC compliant manner; now, we set the port number
+ to 0 in the media stream definition and proceed on with the next
+ media stream. Unfortunately, the formatting of the declined media
+ stream forgot to append a '\r\n' to the end of the media stream.
+ This is normally added to the accepted media streams later on in
+ the processing of the SDP. Since the declined media stream uses a
+ different buffer than the accepted media streams (and is a
+ malloc'd buffer as opposed to a struct ast_str), it's easier to
+ just slap the '\r\n' on the declined media stream buffer rather
+ than attempt to append it later on. So, that's what we do. And
+ now some devices (and probably some providers) will be a bit
+ happier (but probably not terribly happy, since we just rejected
+ something they offered). Review:
+ https://reviewboard.asterisk.org/r/2297/ (closes issue
+ ASTERISK-20908) Reported by: Dennis DeDonatis Tested by: Dennis
+ DeDonatis
+
+ * autoconf/ast_check_pwlib.m4, /, configure: Update configure
+ script to be compatible with ptlib 2.10.9 With ptlib 2.10.9, the
+ configure script fails due to grep returning multiple matches for
+ the pattern it searches for. This patch updates the pattern
+ matching to return only the actual version for the symbol
+ searched for, PTLIB_VERSION. (closes issue ASTERISK-20980)
+ Reported by: Stefan Reuter patches: ASTERISK-20980-1.patch
+ uploaded by Stefan Reuter (license 5339) ........ Merged
+ revisions 380297 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-28 21:08 +0000 [r380255] Sean Bright <sean@malleable.com>
+
+ * /, channels/iax2.h, channels/chan_iax2.c: Correct the number of
+ available call numbers in IAX2. There is currently an edge case
+ where call number 32768 might be allocated for a call, even
+ though the IAX2 protocol requires call numbers be only 15 bits.
+ This resulted in some unpredictable behavior when call number
+ 32678 is chosen. This patch was mostly written by Richard Mudgett
+ via ReviewBoard. I'm just committing it. Review:
+ https://reviewboard.asterisk.org/r/2293/ ........ Merged
+ revisions 380254 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-28 01:57 +0000 [r380211] Russell Bryant <russell@russellbryant.com>
+
+ * /, main/file.c: Change cleanup ordering in filestream destructor.
+ This patch came about due to a problem observed where wav files
+ had an empty header. The header is supposed to be updated in
+ wav_close(). It turns out that this was broken when the
+ cache_record_files option from asterisk.conf was enabled. The
+ cleanup code was moving the file to its final destination
+ *before* running the close() method of the file destructor, so
+ the header didn't get updated. Another problem here is that the
+ move was being done before actually closing the FILE *. Finally,
+ the last bug fixed here is that I noticed that wav_close() checks
+ for stream->filename to be non-NULL. In the previous cleanup
+ order, it's checking a pointer to freed memory. This doesn't
+ actually cause anything to break, but it's treading on dangerous
+ waters. Now the free() of stream->filename is happening after the
+ format module's close() method gets called, so it's safer.
+ Review: https://reviewboard.asterisk.org/r/2286/ ........ Merged
+ revisions 380210 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-27 20:31 +0000 [r380193] Michael L. Young <elgueromexicano@gmail.com>
+
+ * apps/confbridge/conf_config_parser.c: Fix Some Configured
+ Conference Bridge Sounds Not Being Set The "sound_only_one" sound
+ was not being set even though it was configured. In looking into
+ this, I found that the "join" and "leave" prompts were not being
+ set either. (closes issue ASTERISK-20898) Reported by: Stephan
+ Tested by: Stephan Patches:
+ asterisk-20898-custom-sounds-ignored.diff uploaded by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2289/
+
+2013-01-24 16:39 +0000 [r380043] David M. Lee <dlee@digium.com>
+
+ * channels/sip/sdp_crypto.c: Corrected crypto tag in SDP ANSWER for
+ SRTP. When Asterisk responds with an SDP ANSWER for SRTP, it had
+ the code to correctly fill in the crypto data, which was
+ overwritten by a call to sdp_crypto_offer. Corrected the
+ situation by changing sdp_crypto_offer to not replacing crypto
+ data if it already exists. (closes issue ASTERISK-20849) Reported
+ by: José Luis Millán Tested by: Iñaki Baz Castillo Patches:
+ fix_sdp_crypto_tags.diff uploaded by Pedro Kiefer (license 6407)
+
+2013-01-24 04:01 +0000 [r380028] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_confbridge.c: Correct documentation for ConfbridgeList
+ AMI action The documentation for ConfbridgeList states that the
+ Conference field is optional. That's not really the case: if you
+ fail to provide a Conference number, the command will kick back
+ an error. (closes issue AST-1090) Reported by: John Bigelow
+
+2013-01-23 00:23 +0000 [r379964] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/astobj2.c: Attempt to be more helpful when using a bad
+ ao2 object pointer. Put the external obj pointer in the message
+ instead of the internal version. ........ Merged revisions 379963
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-22 22:05 +0000 [r379892-379949] Jonathan Rose <jrose@digium.com>
+
+ * res/res_fax_spandsp.c: res_fax_spandsp: fix t38 transmission bug
+ caused by not returning success This patch fixes the problem, but
+ the issue includes a test which is still being considered for the
+ automated test suite. (issue ASTERISK-20919) Reported by: NITESH
+ BANSAL Patches: patch_ast_fax_spandsp.patch uploaded by NITESH
+ BANSAL (license 6418)
+
+ * /, apps/app_meetme.c, sounds/Makefile: app_meetme: Use new
+ prompts for administrator menu The old prompts for the
+ administrator menu were inadequate. They didn't mention that the
+ menu had additional options through the 8 key and pressing the 8
+ key wouldn't reveal what those options were. This patch fixes all
+ of that while also organizing code pertaining to each individual
+ menu type which was previously all stored in one gigantic
+ function along with many of the basic conference functions.
+ (closes issue AST-996) Reported by: John Bigelow Review:
+ http://reviewboard.digium.internal/r/360/ ........ Merged
+ revisions 379885 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-22 14:51 +0000 [r379826] Matthew Jordan <mjordan@digium.com>
+
+ * /, apps/app_meetme.c: Fix station ringback; trunk hangup issues
+ in SLA This patch fixes two bugs: * If an outbound call is made
+ from a SLA phone using SLAStation, then there is no ringtone
+ audible to the phone that originates the call. The indication of
+ the ringing was not being passed to the SLA station; this patch
+ fixes that by passing through the progress indications. * If an
+ SLA station hangs up before the called party answers, then the
+ channel to the called party continues to ring until a timeout
+ occurs. If the called party manages to answer, Asterisk attempts
+ to connect the called party to a non-existant MeetMe room. This
+ patch corrects the behavior by abandoning the call attempt if it
+ detects that the SLA station is no longer in use while attempting
+ to call the called party. Review:
+ https://reviewboard.asterisk.org/r/2275/ (closes issue
+ ASTERISK-20462) Reported by: dkerr patches:
+ asterisk-11-bugid20440+20462.patch uploaded by dkerr (license
+ 5558) asterisk-11-bugid20462.patch uploaded by dkerr (license
+ 5558) (closes issue ASTERISK-20440) Reported by: dkerr patches:
+ asterisk-11-bugid20440.patch uploaded by dkerr (license 5558)
+ asterisk-11-bugid20440+20462.patch uploaded by dkerr (license
+ 5558) ........ Merged revisions 379825 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-22 00:35 +0000 [r379808] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_bridge.c, apps/app_confbridge.c: confbridge: Minor
+ fixes playing user counts to the conference. * Generate a warning
+ message if sound files do not exist when trying to play the user
+ count to the conference. Use the new helper routine
+ sound_file_exists() for consistency. * Put the new user into
+ autoservice when playing user counts to the conference. * Check
+ the return value of ast_bridge_impart().
+
+2013-01-21 20:40 +0000 [r379790] Matthew Jordan <mjordan@digium.com>
+
+ * contrib/scripts/safe_asterisk, main/asterisk.c,
+ contrib/init.d/rc.suse.asterisk,
+ contrib/init.d/rc.mandriva.asterisk,
+ contrib/init.d/rc.debian.asterisk, /,
+ contrib/init.d/rc.redhat.asterisk, UPGRADE.txt,
+ contrib/init.d/rc.gentoo.asterisk,
+ contrib/init.d/rc.slackware.asterisk,
+ contrib/init.d/rc.archlinux.asterisk: Update init.d scripts to
+ handle stderr; readd splash screen for remote consoles When
+ r376428 was commited to re-order start up sequences to be more
+ tolerant of forking with thread primitives, a few items were
+ changed that caused changes in behavior on some distros. This
+ includes: * Not displaying the splash screen on a remote console.
+ * Displaying an error message on stderr when a remote console
+ cannot connect to a running instance of Asterisk. In the first
+ case, the splash screen was re-added (thanks to Michael L.
+ Young). In the second case, the various init.d scripts were
+ modified to pipe stderr to /dev/null, as the error message is
+ useful - if you execute a remote console or a remote console
+ command execution and it fail, it should tell you. Note that the
+ error message was always present, it just failed to be printed
+ prior to r376428. Much thanks to the folks who quickly reported
+ this problem, provided solutions, and promptly tested the various
+ init.d scripts on a variety of distros. (closes issue
+ ASTERISK-20945) Reported by: Warren Selby Tested by: Michael L.
+ Young, Jamuel Starkey, kaldemar, Danny Nicholas, mjordan patches:
+ asterisk-20945-remote-intro-msg.diff uploaded by elguero (license
+ 5026) ASTERISK-20945-1.8-mjordan.diff uploaded by mjordan
+ (license 6283) ........ Merged revisions 379760 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 379777 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2013-01-21 18:33 +0000 [r379719] Kinsey Moore <kmoore@digium.com>
+
+ * /, codecs/codec_ilbc.c: Prevent segfault for interpolated iLBC
+ frames When iLBC is being used with a jitter buffer and the jb
+ has to interpolate frames, it generates frames with a null
+ pointer and a non-zero datalen. This is now handled properly.
+ (closes issue ASTERISK-20914) Reported By: John McEleney Patches:
+ ASTERISK-20914-1.8.diff uploaded by Matt Jordan (license 6283)
+ ........ Merged revisions 379718 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-21 06:27 +0000 [r379677] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Fix device call logging issues in skinny
+ Skinny device call logging (ie missed, place and received calls)
+ has issues because the incorrect sequence of callstates is/can be
+ sent to the device. This patch removes some extra callstate
+ updates driven by forces external to skinny and ensures the
+ needed intermediary callstate messages are sent. (closes issue
+ ASTERISK-20964) Reported by: wedhorn Tested by: snuffy, myself
+ Patches: ast11-skinny-calllog01.diff uploaded by wedhorn (license
+ 5019)
+
+2013-01-21 04:39 +0000 [r379643] Andrew Latham <lathama@gmail.com>
+
+ * contrib/scripts/install_prereq: Add LDAP libraries to install
+ script Add LDAP dev package to Debian/Ubuntu install list.
+ Existed in Redhat already. (issue ASTERISK-20886)
+
+2013-01-21 04:07 +0000 [r379609] Matthew Jordan <mjordan@digium.com>
+
+ * /, apps/app_minivm.c: Fix crash in app_minivm when mime encoding
+ string An incorrect string initializations was left in
+ ast_str_encode_mime from the patch that converted string
+ manipulations to use ast_str strings (r191140). The string
+ initialization causes a crash when ast_str_set is called on the
+ string later on in the function. (closes issue ASTERISK-18697)
+ Reported by: Chris Boot patches:
+ minivm-null-pointer-dereference-fix.patch uploaded by bootc
+ (license 6309) (issue ASTERISK-20854) Reported by: Chris Warr
+ Tested by: Chris Warr ........ Merged revisions 379608 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-20 02:53 +0000 [r379582] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Fix issues with skinny sessions Fixes a
+ couple of issues with the way skinny handles sessions by ensuring
+ sessions aren't used after being freed. Some other minor changes.
+ Review: https://reviewboard.asterisk.org/r/2272/
+
+2013-01-19 20:49 +0000 [r379548] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, configure, include/asterisk/autoconfig.h.in,
+ include/asterisk/compat.h, main/strcompat.c, configure.ac: Add
+ builtin roundf() for systems lacking it. (closes issue
+ ASTERISK-16854) Review: https://reviewboard.asterisk.org/r/2276
+ Reported-by: Ovidiu Sas ........ Merged revisions 379547 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-19 00:17 +0000 [r379513] Matthew Jordan <mjordan@digium.com>
+
+ * main/asterisk.c, /: Fix astcanary startup problem due to wrong
+ pid value from before daemon call When Asterisk forks itself into
+ the background via a call to daemon, it must re-set the pid value
+ of the new process. Otherwise, astcanary gets the pid value of
+ the process before the fork, which prevents it from running.
+ Asterisk eventually starts lowering its priority, as it can no
+ longer communicate with the proverbial canary in the coal mine.
+ This patch ensures that the correct process identifier is used by
+ astcanary. Note that this is getting committed to 10 as a
+ regression fix. (closes issue ASTERISK-20947) Reported by: Jakob
+ Hirsch Tested by: mjordan patches:
+ asterisk-10.12.0.astcanary_ppid.diff uploaded by Jakob Hirsch
+ (license 6113) ........ Merged revisions 379509 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 379510 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2013-01-18 21:46 +0000 [r379478] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_confbridge.c: Fix regression in Confbridge user count
+ When the restructuring work got committed to Confbridge in
+ r375470 to fix many open issues, it caused a regression in the
+ reported count of users when conference information was requested
+ via CLI or manager. This corrects the user count and user
+ information displayed when listing conference information from
+ the CLI and manager. (closes issue ASTERISK-20938) Reported By:
+ Timo Teras Patches: confbridge-list.patch uploaded by Timo Teras
+ (license 5409)
+
+2013-01-18 21:10 +0000 [r379475] David M. Lee <dlee@digium.com>
+
+ * Makefile, configure, include/asterisk/autoconfig.h.in,
+ main/Makefile, configure.ac, UPGRADE.txt, makeopts.in: Specify
+ the -rpath linker flag when prefix != /usr. This allows Asterisk
+ to start without having to specify the LD_LIBRARY_PATH. This can
+ be disabled by passing --disable-rpath to configure. (closes
+ issue ASTERISK-20407) Reported by: David M. Lee Review:
+ https://reviewboard.asterisk.org/r/2132/
+
+2013-01-18 18:13 +0000 [r379460] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_voicemail.c: app_voicemail: Improve msg_id handling
+ app_voicemail will no longer issue error messages when it
+ retrieves an msg_id with a NULL value from realtime and will
+ instead simply populate the msg_id field with a newly generated
+ msg_id. In addition, this patch changes the way msg_ids are
+ generated to eliminate certain causes of duplicate IDs appearing
+ within a single system. In addition, when messages are copied,
+ they will now receive a new msg_id. (closes issue ASTERISK-20717)
+ Reported by: Alec Davis Review:
+ https://reviewboard.asterisk.org/r/2220/
+
+2013-01-18 05:26 +0000 [r379393] David M. Lee <dlee@digium.com>
+
+ * channels/sip/include/reqresp_parser.h, /, channels/chan_sip.c,
+ channels/sip/reqresp_parser.c: Fix Record-Route parsing for large
+ headers. Record-Route parsing copied the header into a char[256]
+ array, which can be a problem if the header is longer than that.
+ This patch parses the header in place, without the copy, avoiding
+ the issue. In addition to the original patch, I added a unit test
+ for the new get_in_brackets_const function. (closes issue
+ ASTERISK-20837) Reported by: Corey Farrell Patches:
+ chan_sip-build_route-optimized-rev1.patch uploaded by Corey
+ Farrell (license 5909) (with minor changes by dlee) ........
+ Merged revisions 379392 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-17 02:30 +0000 [r379343] Matthew Jordan <mjordan@digium.com>
+
+ * /, addons/chan_mobile.c: Fix issue where chan_mobile fails to
+ bind to first available port Per the bluez API, in order to bind
+ to the first available port, the rc_channel field of the socket
+ addressing structure used to bind the socket should be set to 0.
+ Previously, Asterisk had set the rc_channel field set to 1,
+ causing it to connect to whatever happens to be on port 1. We
+ could probably not explicitly set rc_channel to 0 since we memset
+ the struct earlier, but explicitly setting it will hopefully
+ prevent someone from coming in and setting it to some explicit
+ port in the future. (closes issue ASTERISK-16357) Reported by:
+ challado Tested by: Alexander Heinz, Nikolay Ilduganov, benjamin,
+ eliafino, David van Geyn patches: ASTERISK-16357.diff uploaded by
+ Nikolay Ilduganov (license 6253) ........ Merged revisions 379342
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-16 22:49 +0000 [r379311] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c, /: Further fix misinformation in the description
+ of manager MailboxStatus command. The description still claimed
+ that it returned the number of messages rather than whether there
+ were messages waiting. ........ Merged revisions 379310 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-16 21:13 +0000 [r379277] Jason Parker <jparker@digium.com>
+
+ * contrib/scripts/install_prereq, /: Reduce number of packages
+ install_prereq installs on Debian systems. 'search' will look for
+ any package containing the name provided, so we need to force a
+ more exact search. ........ Merged revisions 379276 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-16 18:08 +0000 [r379230-379232] Richard Mudgett <rmudgett@digium.com>
+
+ * main/logger.c: Reduce call-id logging resource usage. Since there
+ is no need for the call-id logging ao2 object to have a lock,
+ don't create it with one.
+
+ * channels/chan_misdn.c, /: chan_misdn: Fix compile error. (issue
+ ASTERISK-15456) ........ Merged revisions 379226 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-16 17:45 +0000 [r379146-379228] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_xmpp.c, res/res_jabber.c, doc/appdocsxml.dtd: Let
+ documentation reference links specify which module they're
+ linking to Again, since res_jabber/res_xmpp have duplicate APIs,
+ their documentation ref links have to specify which reference
+ they're referring to. The various documentation parsers can
+ interpret the module attribute however they want in order to
+ construct the appropriate links.
+
+ * doc/appdocsxml.dtd: Update the dtd to actually *support* the
+ module attribute in all elements Mea culpa.
+
+ * res/res_xmpp.c, res/res_jabber.c: Add module tags to
+ documentation for res_jabber/res_xmpp Since res_jabber/res_xmpp
+ provide the same APIs (app/func/manager/etc.), the XML
+ documentation for each needs to call out which module is
+ providing the documentation. The module attribute has been added
+ to the various XML fragments for this purpose.
+
+ * /, addons/chan_mobile.c: Fix parsing SMSSRC for SMS messages The
+ parser for SMS messages would incorrectly parse out the from
+ number. The parsing would incorrectly start scanning for the from
+ number at the same index as the first double quote ("); this
+ would inadvertently cause it to treat the first double quote as
+ the terminating double quote for the from number as well. The
+ SMSSRC should now populate correctly. (closes issue
+ ASTERISK-16822) Reported by: menschentier Tested by: Jonas Falck
+ patches: fixSMSSRC.patch uploaded by jonax (license 6320) (closes
+ issue ASTERISK-19153) Reported by: Panos Gkikakis patches:
+ sms-sender-fix.diff uploaded by roeften (license 5884) ........
+ Merged revisions 379178 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_misdn.c, /: Set the INVALID_EXTEN channel variable
+ when chan_misdn forces the 'i' extension The chan_misdn channel
+ driver will send a channel with an invalid destination to the 'i'
+ extension itself if said extension can be reached. It forgot,
+ however, to set the INVALID_EXTEN channel variable when it
+ bounces the channel to this extension. Dialplan writers
+ everywhere moaned at yet another inconsistency. This is yet
+ another example of why duplicating logic in multiple places
+ results in bugs that stick around in Jira for just under three
+ years. Yes: ASTERISK-15456 was created on January 18th, 2010.
+ Patch committed on January 15th, 2013. Ouch. (closes issue
+ ASTERISK-15456) Reported by: Thomas Omerzu patches:
+ chan_misdn_invalid.patch2 uploaded by Thomas Omerzu (license
+ 5927) ........ Merged revisions 379145 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-14 15:27 +0000 [r379020] David M. Lee <dlee@digium.com>
+
+ * /, channels/chan_sip.c: Fix XML encoding of 'identity display' in
+ NOTIFY messages, continued. When r378933 was merged into 1.8, it
+ should have also escaped remote_display, since it will have the
+ same XML encoding problem when the caller/callee roles are
+ reversed. (closes issue ABE-2902) Reported by: Guenther Kelleter
+ ........ Merged revisions 379001 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-13 21:44 +0000 [r378984] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_rtp_asterisk.c, /: Reset RTP timestamp; sequence number
+ on SSRC change In r370252 for ASTERISK-18404, Asterisk's handling
+ of RTP was modified to better account for out of order RTP
+ packets. This was accomplished by using the RTP timestamp and
+ sequence number to check for out of order packets. However, when
+ a SSRC change occurs, the timestamp and sequence number will no
+ longer have any relation to the previously received packets. The
+ variables tracking the timestamp and sequence number therefore
+ have to be reset. (closes issue ASTERISK-20906) Reported by:
+ Eelco Brolman patches: dtmf_on_hold.patch uploaded by Eelco
+ Brolman (license #6442) ........ Merged revisions 378967 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-12 06:36 +0000 [r378934] David M. Lee <dlee@digium.com>
+
+ * include/asterisk/utils.h, /, channels/chan_sip.c,
+ tests/test_xml_escape.c (added), main/utils.c: Fix XML encoding
+ of 'identity display' in NOTIFY messages. XML encoding in
+ chan_sip is accomplished by naively building the XML directly
+ from strings. While this usually works, it fails to take into
+ account escaping the reserved characters in XML. This patch adds
+ an 'ast_xml_escape' function, which works similarly to
+ 'ast_uri_encode'. This is used to properly escape the
+ local_display attribute in XML formatted NOTIFY messages. Several
+ things to note: * The Right Thing(TM) to do would probably be to
+ replace the ast_build_string stuff with building an ast_xml_doc.
+ That's a much bigger change, and out of scope for the original
+ ticket, so I refrained myself. * It is with great sadness that I
+ wrote my own ast_xml_escape function. There's one in libxml2, but
+ it's knee-deep in libxml2-ness, and not easily used to one-off
+ escape a string. * I only escaped the string we know is causing
+ problems (local_display). At least some of the other strings are
+ URI-encoded, which should be XML safe. Rather than figuring out
+ what's safe and escaping what's not, it would be much cleaner to
+ simply build an ast_xml_doc for the messages and let the XML
+ library do the XML escaping. Like I said, that's out of scope.
+ (closes issue ABE-2902) Reported by: Guenther Kelleter Tested by:
+ Guenther Kelleter Review:
+ http://reviewboard.digium.internal/r/365/ ........ Merged
+ revision 378919 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 378933 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-11 23:04 +0000 [r378917] Joshua Colp <jcolp@digium.com>
+
+ * res/res_xmpp.c: Retain XMPP filters across reconnections so
+ external modules continue to function as expected. Previously if
+ an XMPP client reconnected any filters added by an external
+ module were lost. This issue exhibited itself with chan_motif not
+ receiving and reacting to Jingle signaling. (closes issue
+ ASTERISK-20916) Reported by: kuj
+
+2013-01-09 20:29 +0000 [r378734-378780] David M. Lee <dlee@digium.com>
+
+ * main/rtp_engine.c, /: Fix end condition in
+ ast_rtp_lookup_mime_multiple2. The erroneous end condition would
+ never include the AST_RTP_CISCO_DTMF flag in the debug output.
+ (closes issue ASTERISK-20772) Reported by: Xavier Hienne ........
+ Merged revisions 378776 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * include/asterisk/strings.h: Move declaration of
+ ast_regex_string_to_regex_pattern futher down strings.h. The
+ prior location is before the declaration of struct ast_str, which
+ causes compiler warnings. (closes issue ASTERISK-20852) Reported
+ by: Pavel Troller Patches: strings.diff uploaded by Pavel Troller
+ (license 6302)
+
+ * /, include/asterisk/causes.h: Replace errant tabs with spaces in
+ causes.h. (closes issue ASTERISK-20826) Reported by: snuffy
+ Patches: notabs.dif uploaded by snuffy (license 5024) ........
+ Merged revisions 378733 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-09 00:03 +0000 [r378687-378690] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_queue.c: app_queue: Fix incorrect assertion. (issue
+ ASTERISK-16115) ........ Merged revisions 378689 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, configs/queues.conf.sample, UPGRADE.txt, CHANGES,
+ apps/app_queue.c: app_queue: Fix multiple calls to a queue member
+ that is in only one queue. When ringinuse=no queue members can
+ receive more than one call if these calls happen at nearly the
+ same time. * Fix so a queue member does not receive more than one
+ call from a queue. NOTE: This fix does not prevent multiple calls
+ to a member if the member is in more than one queue. * Did some
+ refactoring to eliminate some code redundancy. (issue
+ ASTERISK-16115) Reported by: nik600 Patches:
+ jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch
+ uploaded by rmudgett Modified * Revert the -r341580 and -r341599
+ changes adding the queues.conf check_state_unknown option as it
+ was added in an attempt to fix this problem. The fix did not need
+ to be optional. The fix should not have tried to explicitly set
+ the device state. Setting the device state by something other
+ than the device introduces a race condition. I also could not see
+ how the change would be effective other than delaying the
+ app_queue code long enough for the device state to propagate to
+ app_queue. ........ Merged revisions 378663 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378683 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2013-01-06 20:40 +0000 [r378622] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Rewrite skinny dialing to remove threaded
+ simpleswitch This rewrite changes skinny dialing from the
+ threaded simpleswitch to a scheduled timeout approach. There were
+ some underlying issues with the threaded simple switch with
+ occasional corruption and possible segfaults. Review:
+ https://reviewboard.asterisk.org/r/2240/
+
+2013-01-04 23:04 +0000 [r378592] Jonathan Rose <jrose@digium.com>
+
+ * res/res_srtp.c, /: res_srtp: Prevent a crash from occurring due
+ to srtp_create failures in srtp_create Under some circumstances,
+ libsrtp's srtp_create function deallocates memory that it wasn't
+ initially responsible for allocating. Because we weren't
+ initially aware of this behavior, this memory was still used in
+ spite of being unallocated during the course of the
+ srtp_unprotect function. A while back I made a patch which would
+ set this value to NULL, but that exposed a possible condition
+ where we would then try to check a member of the struct which
+ would cause a segfault. In order to address these problems,
+ ast_srtp_unprotect will now set an error value when it ends
+ without a valid SRTP session which will result in the caller of
+ srtp_unprotect observing this error and hanging up the relevant
+ channel instead of trying to keep using the invalid session
+ address. (closes issue ASTERISK-20499) Reported by: Tootai
+ Review:
+ https://reviewboard.asterisk.org/r/2228/diff/#index_header
+ ........ Merged revisions 378591 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-04 22:18 +0000 [r378582] Kinsey Moore <kmoore@digium.com>
+
+ * res/pjproject/aconfigure, res/pjproject/aconfigure.ac,
+ res/pjproject/build/common.mak: Fix pjproject compilation in
+ certain circumstances On a fresh checkout of Asterisk 11, running
+ make before ./configure could cause the pjproject subdirectory to
+ get in an odd state that would prevent compilation. This patch by
+ Tilghman prevents that from occurring. (closes issue
+ ASTERISK-20681) Reported by: Dinesh Ramjuttun Tested by: danilo
+ borges, Steve Lang patches: 20121208__ccar_solved.diff.txt
+ uploaded by Tilghman Lesher (license 5003)
+
+2013-01-04 21:18 +0000 [r378559] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, channels/chan_sip.c: Fix SIP Notify Messages To Have The
+ Proper IP Address In The FROM Field On a multihomed server when
+ sending a NOTIFY message, we were not figuring out which network
+ should be used to contact the peer. This patch fixes the problem
+ by calling ast_sip_ouraddrfor() and then build_via() so that our
+ NOTIFY message contains the correct IP address. Also, a debug
+ message is being added to help follow the call-id changes that
+ occur. This was helpful for confirming that the IP address was
+ set properly since the call-id contains the IP address. It also
+ will be helpful for troubleshooting purposes when following a
+ call in the debug logs. (closes issue ASTERISK-20805) Reported
+ by: Bryan Hunt Tested by: Bryan Hunt, Michael L. Young Patches:
+ asterisk-20805-notify-ip-v2.diff uploaded by Michael L. Young
+ (license 5026) Review: https://reviewboard.asterisk.org/r/2255/
+ ........ Merged revisions 378554 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-04 21:16 +0000 [r378555] Joshua Colp <jcolp@digium.com>
+
+ * res/res_rtp_asterisk.c, /: Don't pass STUN packets through the
+ SRTP unprotect function. (closes issue AST-1036) Reported by:
+ jbigelow ........ Merged revisions 378553 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-03 22:12 +0000 [r378515] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, apps/app_queue.c: Fix Queue Log Reporting Every Call
+ COMPLETECALLER With "h" Extension Present When the "h" extension
+ is present within the context of the queue, all calls are being
+ reported COMPLETECALLER even when the agent is hanging up the
+ call. This patch checks to see if the agent hung-up or not
+ instead of only relying on checking if the queue (caller) channel
+ hung-up or not. It would appear that having the h extension in
+ the mix, the pbx goes to the h extension, "hanging-up" the queue
+ channel and triggering the reporting of COMPLETECALLER. (closes
+ issue ASTERISK-20743) Reported by: call Tested by: call, Michael
+ L. Young Patches: asterisk-20743-q-cmplt-caller.diff uploaded by
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2256/ ........ Merged
+ revisions 378514 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-03 19:41 +0000 [r378487] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_agent.c: chan_agent: Fix wrapup time wait
+ response. * Made agent_cont_sleep() and agent_ack_sleep() stop
+ waiting if the wrapup time expires. agent_cont_sleep() had tried
+ but returned the wrong value to stop waiting. * Made
+ agent_ack_sleep() take a struct agent_pvt pointer instead of a
+ void pointer for better type safety. ........ Merged revisions
+ 378486 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-03 18:48 +0000 [r378459] Kinsey Moore <kmoore@digium.com>
+
+ * main/channel.c, /: Add missing test event This test event was
+ missing from channel.c causing the dial_LS_options test to fail
+ intermittently because of a race condition where most code paths
+ emitted the test event but this one did not. The dial_LS_options
+ test should stop bouncing now. ........ Merged revisions 378455
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-03 18:44 +0000 [r378428-378457] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_agent.c: chan_agent: Misc code cleanup. * Fix
+ off-nominal path resource cleanup in agent_request(). * Create
+ agent_pvt_destroy() to eliminate inlined versions in many places.
+ * Pull invariant code out of loop in add_agent(). * Remove
+ redundant module user references in login_exec(). * Remove unused
+ struct agent_pvt logincallerid[] member. * Remove some redundant
+ code in agent_request(). ........ Merged revisions 378456 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_agent.c: chan_agent: Fix agent_indicate()
+ locking. Avoid deadlock potential with local channels and
+ simplify the locking. ........ Merged revisions 378427 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-03 15:38 +0000 [r378411] Joshua Colp <jcolp@digium.com>
+
+ * res/res_xmpp.c: Prevent exhaustion of system resources through
+ exploitation of event cache This patch changes res_xmpp to no
+ longer cache events under certain circumstances. (issue
+ ASTERISK-20175) Reported by: Russell Bryant, Leif Madsen, Joshua
+ Colp Tested by: kmoore
+
+2013-01-03 15:36 +0000 [r378376-378409] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_xmpp.c: Prevent crashes in res_xmpp when receiving large
+ messages Similar to r378287, res_xmpp was marshaling data read
+ from an external source onto the stack. For a sufficiently large
+ message, this could cause a stack overflow. This patch modifies
+ res_xmpp in a similar fashion to res_jabber by removing the stack
+ allocation, as it was unnecessary. (issue ASTERISK-20658)
+ Reported by: wdoekes
+
+ * main/config.c, funcs/func_realtime.c, /: Prevent crashes from
+ occurring when reading from data sources with large values When
+ reading configuration data from an Asterisk .conf file or when
+ pulling data from an Asterisk RealTime backend, Asterisk was
+ copying the data on the stack for manipulation. Unfortunately, it
+ is possible to read configuration data or realtime data from some
+ data source that provides a large blob of characters. This could
+ potentially cause a crash via a stack overflow. This patch
+ prevents large sets of data from being read from an ARA backend
+ or from an Asterisk conf file. (issue ASTERISK-20658) Reported
+ by: wdoekes Tested by: wdoekes, mmichelson patches: *
+ issueA20658_dont_process_overlong_config_lines.patch uploaded by
+ wdoekes (license 5674) * issueA20658_func_realtime_limit.patch
+ uploaded by wdoekes (license 5674) ........ Merged revisions
+ 378375 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-02 21:17 +0000 [r378358] Richard Mudgett <rmudgett@digium.com>
+
+ * main/manager.c, /, main/features.c, include/asterisk/channel.h:
+ Fix AMI redirect action with two channels failing to redirect
+ both channels. The AMI redirect action can fail to redirect two
+ channels that are bridged together. There is a race between the
+ AMI thread redirecting the two channels and the bridge thread
+ noticing that a channel is hungup from the redirects. * Made the
+ bridge wait for both channels to be redirected before exiting. *
+ Made the AMI redirect check that all required headers are present
+ before proceeding with the redirection. * Made the AMI redirect
+ require that any supplied ExtraChannel exist before proceeding.
+ Previously the code fell back to a single channel redirect
+ operation. (closes issue ASTERISK-18975) Reported by: Ben Klang
+ (closes issue ASTERISK-19948) Reported by: Brent Dalgleish
+ Patches: jira_asterisk_19948_v11.patch (license #5621) patch
+ uploaded by rmudgett Tested by: rmudgett, Thomas Sevestre, Deepak
+ Lohani, Kayode Review: https://reviewboard.asterisk.org/r/2243/
+ ........ Merged revisions 378356 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-02 18:30 +0000 [r378337] Kinsey Moore <kmoore@digium.com>
+
+ * /: Restore branch-1.8-merged on 11 This was accidentally deleted
+ during a merge.
+
+2013-01-02 18:09 +0000 [r378287-378321] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_calendar.c, include/asterisk/devicestate.h,
+ channels/chan_local.c, /, main/ccss.c, channels/chan_sip.c,
+ apps/app_meetme.c, main/channel_internal_api.c,
+ channels/chan_agent.c, main/devicestate.c,
+ include/asterisk/channel.h, res/res_jabber.c, apps/app_queue.c,
+ channels/chan_iax2.c, main/channel.c, channels/chan_dahdi.c,
+ channels/chan_skinny.c, include/asterisk/event_defs.h,
+ main/features.c, main/event.c, apps/app_confbridge.c,
+ apps/confbridge/conf_state_empty.c, funcs/func_devstate.c:
+ Prevent exhaustion of system resources through exploitation of
+ event cache Asterisk maintains an internal cache for devices in
+ the event subsystem. The device state cache holds the state of
+ each device known to Asterisk, such that consumers of device
+ state information can query for the last known state for a
+ particular device, even if it is not part of an active call. The
+ concept of a device in Asterisk can include entities that do not
+ have a physical representation. One way that this occurred was
+ when anonymous calls are allowed in Asterisk. A device was
+ automatically created and stored in the cache for each anonymous
+ call that occurred; this was possible in the SIP and IAX2 channel
+ drivers and through channel drivers that utilized the
+ res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif).
+ These devices are never removed from the system, allowing
+ anonymous calls to potentially exhaust a system's resources. This
+ patch changes the event cache subsystem and device state
+ management to no longer cache devices that are not associated
+ with a physical entity. (issue ASTERISK-20175) Reported by:
+ Russell Bryant, Leif Madsen, Joshua Colp Tested by: kmoore
+ patches: event-cachability-3.diff uploaded by jcolp (license
+ 5000) ........ Merged revisions 378303 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378320 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c, main/http.c,
+ res/res_jabber.c: Resolve crashes due to large stack allocations
+ when using TCP Asterisk had several places where messages
+ received over various network transports may be copied in a
+ single stack allocation. In the case of TCP, since multiple
+ packets in a stream may be concatenated together, this can lead
+ to large allocations that overflow the stack. This patch modifies
+ those portions of Asterisk using TCP to either favor heap
+ allocations or use an upper bound to ensure that the stack will
+ not overflow: * For SIP, the allocation now has an upper limit *
+ For HTTP, the allocation is now a heap allocation instead of a
+ stack allocation * For XMPP (in res_jabber), the allocation has
+ been eliminated since it was unnecesary. Note that the HTTP
+ portion of this issue was independently found by Brandon Edwards
+ of Exodus Intelligence. (issue ASTERISK-20658) Reported by:
+ wdoekes, Brandon Edwards Tested by: mmichelson, wdoekes patches:
+ ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license
+ 5049) issueA20658_http_postvars_use_malloc2.patch uploaded by
+ wdoekes (license 5674) issueA20658_limit_sip_packet_size3.patch
+ uploaded by wdoekes (license 5674) ........ Merged revisions
+ 378269 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 378286 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-31 14:44 +0000 [r378219] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Ensure chan_sip rejects encrypted streams
+ without crypto info This ensures that Asterisk rejects encrypted
+ media streams (RTP/SAVP audio and video) that are missing
+ cryptographic keys and ensures that the incoming SDP is
+ consistent with RFC4568 as far as having a crypto attribute
+ present for any SAVP streams. Review:
+ https://reviewboard.asterisk.org/r/2204/ ........ Merged
+ revisions 378217 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378218 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-20 21:44 +0000 [r378163-378165] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /: Give the causes[] a struct name. ........
+ Merged revisions 378164 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /: Add branch-1.8-merged property to allow direct merging from
+ v1.8
+
+2012-12-18 17:41 +0000 [r378121] Kinsey Moore <kmoore@digium.com>
+
+ * main/channel.c, /: Add test events for time limit-related hangups
+ This patch adds hangup-related test events in order to support
+ testing of time-limited bridges. This aids in testing the S() and
+ L() bridge options. (issue SWP-4713) ........ Merged revisions
+ 378119 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 378120 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-17 23:09 +0000 [r378090-378094] Richard Mudgett <rmudgett@digium.com>
+
+ * main/loader.c, /: Fix potential double free when unloading a
+ module. ........ Merged revisions 378092 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378093 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/chan_local.c, /: Make chan_local module references tied
+ to local_pvt lifetime. The chan_local module references were
+ manually tied to the existence of the ;1 and ;2 channel links. *
+ Made chan_local module references tied to the existence of the
+ local_pvt structure as well as automatically take care of the
+ module references. * Tweaked the wording of the local_fixup()
+ failure warning message to make sense. Review:
+ https://reviewboard.asterisk.org/r/2181/ ........ Merged
+ revisions 378088 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378089 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-17 20:58 +0000 [r378073] Jason Parker <jparker@digium.com>
+
+ * main/Makefile: Make libasteriskssl.so symlink use a relative
+ path. This was causing issues when using DESTDIR, since the path
+ to which the link pointed is not likely to exist (and not useful
+ to exist) on the target system. (issue ASTNOW-284)
+
+2012-12-14 21:32 +0000 [r378038] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_queue.c: app_queue: Revert bad ringinuse=no patch.
+ With the option ringinuse=no set, the patch committed for
+ ASTERISK-16115 causes non-SIP queue members to never be called
+ because the device state is checked after a channel is created to
+ determine if the member is busy. These queue members always get
+ the "Member %s is busy, cannot dial" message. Most channel
+ drivers other than chan_sip use the default device state
+ handling. The default device-state state is considered in use or
+ unknown if the channel exists or not respectively. (closes issue
+ ASTERISK-20801) Reported by: rmudgett Patches:
+ jira_asterisk_16115_revert_r370418_v1.8.patch (license #5621)
+ patch uploaded by rmudgett ........ Merged revisions 378036 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378037 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-14 01:49 +0000 [r378010] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Fix skinny to recognise vmexten in
+ general section of conf Fixup the vmexten so if globally set in
+ general section will be honored by chan_skinny. Also get rid of
+ the 'global_' part of variable name to match regexten. (closes
+ issue ASTERISK-20790) Reported by: snuffy Tested by: snuffy,
+ myself Patches: skinny-vm.diff uploaded by snuffy (license 5024)
+
+2012-12-13 21:04 +0000 [r377993] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/confbridge/conf_state.c, /,
+ apps/confbridge/include/confbridge.h,
+ include/asterisk/bridging.h, apps/app_confbridge.c,
+ apps/confbridge/conf_state_multi_marked.c: confbridge: Fix MOH on
+ simultaneous user entry to a new conference. When two users
+ entered a new conference simultaneously, one of the callers hears
+ MOH. This happened if two unmarked users entered simultaneously
+ and also if a waitmarked and a marked user entered
+ simultaneously. * Created a confbridge internal MOH API to
+ eliminate the inlined MOH handling code. Note that the conference
+ mixing bridge needs to be locked when actually starting/stopping
+ MOH because there is a small window between the conference join
+ unsuspend MOH and actually joining the mixing bridge. * Created
+ the concept of suspended MOH so it can be interrupted while
+ conference join announcements to the user and DTMF features can
+ operate. * Suspend any MOH until the user is about to actually
+ join the mixing bridge of the conference. This way any pre-join
+ file playback does not need to worry about MOH. * Made post-join
+ actions only play deferred entry announcement files. Changing the
+ user/conference state during that time is not protected or
+ controlled by the state machine. (closes issue ASTERISK-20606)
+ Reported by: Eugenia Belova Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/2232/ ........ Merged
+ revisions 377992 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-13 20:03 +0000 [r377985-377991] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Minor fixes for chan_skinny Whitespace,
+ change SUBSTATE_ONHOOK to correct SKINNY_ONHOOK and correct len
+ of 2 strcmp in skinny_setdebug(). (see opticron's review on
+ https://reviewboard.asterisk.org/r/2240/)
+
+ * channels/chan_skinny.c: Fix skinny debug tab completion Review
+ the syntax of the 'skinny debug' command to show more than just
+ 'show' for options to 'skinny debug' command. (closes issue
+ ASTERISK-20789) Reported by: snuffy Tested by: snuffy, myself
+ Patches: skinny-debug.diff uploaded by snuffy (license 5024)
+
+2012-12-13 13:51 +0000 [r377948] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Ensure Min-SE is included in outbound
+ INVITEs Asterisk now includes Min-SE in outbound INVITEs when the
+ value is not 90 (the default) and session timers are not
+ disabled. This has the effect of Asterisk following RFC4028 more
+ closely with regard to 422 responses and preventing situations in
+ which Asterisk would be forced to temporarily accept a call to
+ tear it down based on a Session-Expires below the locally
+ configured Min-SE. (issue SWP-5051) Review:
+ https://reviewboard.asterisk.org/r/2222/ Reported-by: Kinsey
+ Moore Patch-by: Kinsey Moore ........ Merged revisions 377946
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 377947 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-12 22:42 +0000 [r377924] Rusty Newton <rnewton@digium.com>
+
+ * /, sounds/Makefile: Incremented EXTRA_SOUNDS_VERSION in
+ sounds/Makefile to 1.4.12 for new Extra Sounds releases See
+ CHANGES-* files in English extra 1.4.12 tarballs for new sound
+ prompts added. (closes ASTERISK-20328) Reported by: Matt Jordan
+ (closes AST-755) Reported by: John Bigelow ........ Merged
+ revisions 377922 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377923 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-11 23:59 +0000 [r377910] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix a potential deadlock in chan_sip during
+ transfers. The issue comes from the fact that transfers may
+ perform a redirecting update on a channel. The issue is that lock
+ inversion between the channel and its tech_pvt occurs since the
+ channel lock is released during the transfer process. The fix is
+ to move when the redirecting update occurs to a place where
+ neither the tech_pvt or the channel is locked so that the two can
+ be locked in the proper order. (closes issue ASTERISK-20708)
+ reported by Mark Michelson patches: ASTERISK-20708-3.patch
+ uploaded by Mark Michelson (License #5049) Tested by: Tim
+ Ringenbach at Asteria Solutions Group
+
+2012-12-11 22:01 +0000 [r377849-377883] Richard Mudgett <rmudgett@digium.com>
+
+ * main/timing.c, main/channel.c, main/data.c, main/stun.c, /,
+ main/file.c, main/http.c, main/aoc.c, main/image.c, main/cel.c:
+ Cleanup CLI commands on exit for several files. (issue
+ ASTERISK-20649) Reported by: Corey Farrell Patches:
+ unregister-cli-multiple-all.patch (license #5909) patch uploaded
+ by Corey Farrell ........ Merged revisions 377881 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377882 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/udptl.c, /: Cleanup udptl on exit. * Cleanup CLI commands on
+ exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches:
+ udptl-shutdown-1_8-10.patch (license #5909) patch uploaded by
+ Corey Farrell udptl-shutdown-11-trunk.patch (license #5909) patch
+ uploaded by Corey Farrell Modified ........ Merged revisions
+ 377847 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 377848 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-11 20:51 +0000 [r377843] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_clialiases.c, /: Fix crash that can occur if CLI
+ registration fails for an aliased command. A recent memory leak
+ fix in main/cli.c causes an ast_cli_entry's command field to be
+ freed and NULLed if ast_cli_register() fails. res_clialiases was
+ ignoring the return value of ast_cli_register() and was then
+ passing the NULL command off to a a hash function. This resulted
+ in a crash. The fix is not to ignore the erroneous return value.
+ If ast_cli_register() fails, then we do not continue trying to
+ process the current alias. ........ Merged revisions 377840 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377842 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-11 20:45 +0000 [r377706-377839] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/taskprocessor.c: Cleanup taskprocessor on exit. * Cleanup
+ CLI commands on exit. (issue ASTERISK-20649) Reported by: Corey
+ Farrell Patches: taskprocessor-cleanup-1_8-11-trunk.patch
+ (license #5909) patch uploaded by Corey Farrell
+ taskprocessor-cleanup-10-only.patch (license #5909) patch
+ uploaded by Corey Farrell Modified ........ Merged revisions
+ 377837 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 377838 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/pbx.c, /: Cleanup pbx on exit. * Cleanup CLI commands on
+ exit. * Unreference hints and statecbs containers on exit. (issue
+ ASTERISK-20649) Reported by: Corey Farrell Patches:
+ pbx-cleanup-1_8.patch (license #5909) patch uploaded by Corey
+ Farrell pbx-cleanup-10.patch (license #5909) patch uploaded by
+ Corey Farrell pbx-cleanup-11-trunk.patch (license #5909) patch
+ uploaded by Corey Farrell Modified ........ Merged revisions
+ 377806 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 377807 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, main/logger.c: Cleanup logger on exit. * Cleanup CLI commands,
+ destroy verbosers and logchannels lists on exit. (issue
+ ASTERISK-20649) Reported by: Corey Farrell Patches:
+ logger-cleanup-all.patch (license #5909) patch uploaded by Corey
+ Farrell Modified ........ Merged revisions 377771 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377772 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, main/indications.c: Cleanup indications on exit. * Made
+ ast_unregister_indication_country() unlink the found tone zone
+ before selecting a new default_tone_zone to make it impossible to
+ select the tone zone being unregistered again. * Ringcadence is
+ no longer parsed twice in store_config_tone_zone(). * Cleanup CLI
+ commands and destroy default_tone_zone on exit. (issue
+ ASTERISK-20649) Reported by: Corey Farrell Patches:
+ indications-cleanup-all.patch (license #5909) patch uploaded by
+ Corey Farrell Modified ........ Merged revisions 377740 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377741 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, main/event.c: Cleanup event on exit. * Cleanup CLI commands on
+ exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches:
+ event_shutdown-10-only.patch (license #5909) patch uploaded by
+ Corey Farrell event_shutdown-1_8-11-trunk.patch (license #5909)
+ patch uploaded by Corey Farrell ........ Merged revisions 377708
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 377709 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/dnsmgr.c, /: Cleanup dnsmgr on exit. * Cleanup dnsmgr thread
+ and CLI commands on exit. (issue ASTERISK-20649) Reported by:
+ Corey Farrell Patches: dnsmgr-cleanup-1_8.patch (license #5909)
+ patch uploaded by Corey Farrell dnsmgr-cleanup-10-11-trunk.patch
+ (license #5909) patch uploaded by Corey Farrell Modified ........
+ Merged revisions 377704 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377705 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-10 16:55 +0000 [r377625-377657] Kinsey Moore <kmoore@digium.com>
+
+ * /, res/res_fax.c: Ensure ReceiveFax provides a CED tone via T.38
+ When using res_fax_digium, the T.38 CED tone was not being
+ provided properly which would cause some incoming faxes to fail.
+ This was not an issue with res_fax_spandsp since it does not
+ strictly honor the send_ced flag and sends the CED tone whenever
+ receiving a T.38 fax. (closes issue FAX-343) Reported-by:
+ Benjamin Tietz Patch-by: Kinsey Moore ........ Merged revisions
+ 377655 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 377656 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, channels/chan_sip.c: Handle Session-Expires less than local
+ Min-SE in 200 OK Ensure that a call is immediately torn down if a
+ Session-Expires value received in a 200 OK is less than the local
+ Min-SE. This also prevents Asterisk from allowing calls with
+ Session-Expires below the RFC4028-mandated minimum (90s). (closes
+ issue ASTERISK-20653) Review:
+ https://reviewboard.asterisk.org/r/2237/ Patch-by: Kinsey Moore
+ ........ Merged revisions 377623 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377624 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-10 06:49 +0000 [r377577-377593] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * channels/chan_unistim.c, /: Fix codec mismatch Fix code to send
+ in both rx and tx open stream messages correct codecs. Found that
+ on phase 0/1 phones wrong codecs cause to no audio in some
+ situations. (issue ASTERISK-20183) ........ Merged revisions
+ 377591 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 377592 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/chan_unistim.c: Remove trailing whitespaces in number
+ from incoming redial list. Reported by: Igor Olhovskiy
+
+2013-01-14 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 11.2.0 Released.
+
+2013-01-09 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 11.2.0-rc2 Released.
+
+ * Fix pjproject compilation in certain circumstances.
+
+ On a fresh checkout of Asterisk 11, running make before ./configure
+ could cause the pjproject subdirectory to get in an odd state that
+ would prevent compilation. This patch by Tilghman prevents that from
+ occurring.
+
+ (closes issue ASTERISK-20681)
+ Patch-by: Tilghman Lesher
+
+ * AST-2012-014: Resolve crashes due to large stack allocations when
+ using TCP
+
+ Asterisk had several places where messages received over various
+ network transports may be copied in a single stack allocation. In
+ the case of TCP, since multiple packets in a stream may be
+ concatenated together, this can lead to large allocations that
+ overflow the stack.
+
+ This patch modifies those portions of Asterisk using TCP to either
+ favor heap allocations or use an upper bound to ensure that the
+ stack will not overflow:
+ * For SIP, the allocation now has an upper limit
+ * For HTTP, the allocation is now a heap allocation instead of a
+ stack allocation
+ * For XMPP, the allocation has been eliminated since it was
+ unnecessary.
+
+ This patch contains the fix for both res_jabber and res_xmpp.
+
+ * AST-2012-015: Prevent exhaustion of system resources through
+ exploitation of event cache
+
+ Asterisk maintains an internal cache for devices in the event
+ subsystem. The device state cache holds the state of each device
+ known to Asterisk, such that consumers of device state information
+ can query for the last known state for a particular device, even if
+ it is not part of an active call. The concept of a device in
+ Asterisk can include entities that do not have a physical
+ representation. One way that this occurred was when anonymous calls
+ are allowed in Asterisk. A device was automatically created and
+ stored in the cache for each anonymous call that occurred; this was
+ possible in the SIP and IAX2 channel drivers and through channel
+ drivers that utilized the res_jabber/res_xmpp resource modules (Gtalk,
+ Jingle, and Motif). These devices are never removed from the system,
+ allowing anonymous call to potentially exhaust a system's resources.
+
+ This patch changes the event cache subsystem and device state
+ management to no longer cache devices that are not associated with a
+ physical entity.
+
+ * Revert bad ringinuse=no patch.
+
+ With the option ringinuse=no set, the patch committed previous for
+ ASTERISK-16115 causes non-SIP queue members to never be called
+ because the device state is checked after a channel is created to
+ determine if the member is busy. These queue members always get the
+ "Member %s is busy, cannot dial" message.
+
+ Most channel drivers other than chan_sip use the default device
+ state handling. The default device state is considered in use or
+ unknown if the channel exists or not, respectively.
+
+ * Fix multiple calls to a queue member that is only in queue.
+
+ When ringinuse=no queue members can receive more than one call if
+ these calls happen at nearly the same time. This patch fixes it so a
+ queu member does not receive more than one call from a queue. note
+ that this fix does not prevent multiple calls to a member if hte
+ member is in more than one queue (see ASTERISK-16115).
+
+2012-12-10 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 11.2.0-rc1 Released.
+
+2012-12-10 01:41 +0000 [r377505-377511] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/xmldoc.c, /: Improve documentation by making all of the
+ colors used readable, no matter what the background color is.
+ Dark blue on a black background is unreadable, as is yellow on a
+ light background. This patch turns on the bright attribute for
+ colors when on a dark background and turns *off* the bright
+ attribute when the -W command line option is used (indicating a
+ _light_ background). This ensures that text is readable in both
+ cases. Patch by: tilghman Review:
+ https://reviewboard.asterisk.org/r/2224 ........ Merged revisions
+ 377509 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 377510 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, addons/cdr_mysql.c: Remove some dead code and additionally
+ handle a case that wasn't handled. ........ Merged revisions
+ 377487 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 377504 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-09 01:22 +0000 [r377462] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_motif.c: Add missing support for "who hung up" to
+ chan_motif. (closes issue ASTERISK-20671) Reported by: Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/2208/
+
+2012-12-08 00:29 +0000 [r377401-377433] Richard Mudgett <rmudgett@digium.com>
+
+ * contrib/realtime/mysql/sippeers.sql, /: Fix order of SIP
+ allow/disallow in MySQL contrib script. Using the contrib
+ sippeers.sql script to create the sippeers MySQL table would
+ result in being unable to place calls if you set the disallow
+ value to all. (closes issue ASTERISK-20756) Reported by: Andre
+ Luis Patches: sippeers.patch patch uploaded by Andre Luis
+ ........ Merged revisions 377431 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377432 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, main/astmm.c: MALLOC_DEBUG: Only wait if we want atexit
+ allocation dumps. ........ Merged revisions 377398 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377399 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-07 22:02 +0000 [r377383] Kinsey Moore <kmoore@digium.com>
+
+ * /, codecs/codec_dahdi.c: codec_dahdi: Fix output of "transcoder
+ show" CLI command. In r306010 "Asterisk media architecture
+ conversion - no more format bitfields", the logic for
+ incrementing encoders and decoders when opening transcoder
+ channels was changed without making the corresponding change when
+ decrementing encoder / decoder channels. The result being that
+ when a channel was destroyed, codec_dahdi couldn't properly tell
+ if it was an encoder or decoder, and the default case is to
+ assume it was a decoder. This could result in negative numbers
+ for decoders in use like in: VOIP6*CLI> transcoder show 2/-2
+ encoders/decoders of 92 channels are in use. (closes issue
+ ASTERISK-19921) Patch-by: Shaun Ruffell ........ Merged revisions
+ 377382 from http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-06 23:58 +0000 [r377355] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/confbridge/conf_config_parser.c, /, apps/app_confbridge.c:
+ confbridge: Fix some resource leaks on conference teardown. *
+ Made destroy_conference_bridge() destroy a missed ast_mutex_t and
+ ast_cond_t. * Made join_conference_bridge() init the
+ ast_mutex_t's and ast_cond_t so destroy_conference_bridge() can
+ destroy them unconditionally. * Made join_conference_bridge()
+ abort if the new conference could not be added to the conferences
+ container. * Made leave_conference() discard any post-join
+ actions if join_conference_bridge() had to abort early. * Made
+ the join_conference_bridge() diagnostic messages better describe
+ what happened. * Renamed leave_conference_bridge() to
+ leave_conference() and made it only take a conference user
+ pointer. The conference pointer was redundant. * Made
+ conf_bridge_profile_copy() use struct copy instead of memcpy(). *
+ No need to lock the conference in start_conf_record_thread()
+ since all of the callers already have it locked. ........ Merged
+ revisions 377354 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-06 17:28 +0000 [r377340] Russell Bryant <russell@russellbryant.com>
+
+ * main/named_acl.c: Add CLI tab completion to 'acl show'. The 'acl
+ show' CLI command allows you to show the details about a specific
+ named ACL in acl.conf. This patch adds tab completion to the
+ command. Review: https://reviewboard.asterisk.org/r/2230/
+
+2012-12-06 14:11 +0000 [r377319] Matthew Jordan <mjordan@digium.com>
+
+ * main/manager.c: Fix memory leak in 'manager show event' when
+ command entered incorrectly When the CLI command 'manager show
+ event' was run incorrectly and its usage instructions returned, a
+ reference to the event container was leaked. This would prevent
+ the container from being reclaimed when Asterisk exits. We now
+ properly decrement the count on the ao2 object using the nifty
+ RAII_VAR macro. Thanks to Russell for helping me stumble on this,
+ and Terry for writing that ridiculously helpful macro.
+
+2012-12-05 17:08 +0000 [r377262] Jonathan Rose <jrose@digium.com>
+
+ * res/res_srtp.c, /: res_srtp: Fix a crash caused by srtp_dealloc
+ on an already dealloced session When srtp_create fails, the
+ session may be dealloced or just not alloced. At the same time
+ though, the session pointer might not be set to NULL in this
+ process and attempting to srtp_dealloc it again will cause a
+ segfault. This patch checks for failure of srtp_create and sets
+ the session pointer to NULL if it fails. (closes issue
+ ASTERISK-20499) Reported by: tootai Review:
+ https://reviewboard.asterisk.org/r/2228/ ........ Merged
+ revisions 377256 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377261 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-05 16:50 +0000 [r377259] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Fix a SIP request memory leak with TLS
+ connections. During the TLS re-work in chan_sip some TLS specific
+ code was moved into a separate function. This function operates
+ on a copy of the incoming SIP request. This copy was never
+ deinitialized causing a memory leak for each request processed.
+ This function is now given a SIP request structure which it can
+ use to copy the incoming request into. This reduces the amount of
+ memory allocations done since the internal allocated components
+ are reused between packets and also ensures the SIP request
+ structure is deinitialized when the TLS connection is torn down.
+ (closes issue ASTERISK-20763) Reported by: deti ........ Merged
+ revisions 377257 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377258 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-05 02:19 +0000 [r377213-377244] Richard Mudgett <rmudgett@digium.com>
+
+ * main/format.c, /: Fix registering core show codecs/codec CLI
+ commands twice. ........ Merged revisions 377241 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * apps/confbridge/conf_config_parser.c, /: confbridge: Fix several
+ small issues. * Made func_confbridge_helper() allow an empty
+ value when setting options. You previously could not
+ Set(CONFBRIDGE(user,pin)=) and clear the configured pin from the
+ dialplan. * Made func_confbridge_helper() handle its datastore
+ better if multiple threads attempt to set the first CONFBRIDGE
+ option value on the channel. * Made the func_confbridge_helper()
+ only output one diagnostic message concerning the option. * Made
+ the bridge video_mode able to repeatedly change in the config
+ file and CONFBRIDGE dialplan function. The video_mode option
+ values are an enum and not independent of each other. * Made
+ handle_cli_confbridge_show_bridge_profile() better handle the
+ video_mode option. * Simplified datastore handling code in
+ conf_find_user_profile() and conf_find_bridge_profile(). (closes
+ issue ASTERISK-20655) Reported by: Birger "WIMPy" Harzenetter
+ ........ Merged revisions 377227 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, apps/app_confbridge.c: confbridge: Update online XML
+ documentation. ........ Merged revisions 377212 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-04 12:59 +0000 [r377195] Russell Bryant <russell@russellbryant.com>
+
+ * contrib/scripts/install_prereq: Add libuuid to install_prereq for
+ Fedora. I ran this script and my build failed. pjproject requires
+ this.
+
+2012-12-03 22:58 +0000 [r377039-377167] Richard Mudgett <rmudgett@digium.com>
+
+ * main/asterisk.c, /: Cleanup ast_run_atexits() atexits list. *
+ Convert atexits list to a mutex instead of a rd/wr lock. The lock
+ is only write locked. * Move CLI verbose Asterisk ending message
+ to where AMI message is output in really_quit() to avoid further
+ surprises about using stuff already shutdown. (issue
+ ASTERISK-20649) Reported by: Corey Farrell ........ Merged
+ revisions 377165 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377166 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/asterisk.c, /, include/asterisk/_private.h,
+ main/stdtime/localtime.c: Cleanup core main on exit. * Cleanup
+ time zones on exit. * Make exit clean/unclean report consistent
+ for AMI and CLI in really_quit(). (issue ASTERISK-20649) Reported
+ by: Corey Farrell Patches: core-cleanup-1_8-10.patch (license
+ #5909) patch uploaded by Corey Farrell
+ core-cleanup-11-trunk.patch (license #5909) patch uploaded by
+ Corey Farrell Modified ........ Merged revisions 377135 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377136 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/config.c, /: Cleanup config cache on exit. (issue
+ ASTERISK-20649) Reported by: Corey Farrell Patches:
+ config-cleanup-all.patch (license #5909) patch uploaded by Corey
+ Farrell ........ Merged revisions 377104 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377105 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/cli.c, /: Cleanup CLI resources on exit and CLI command
+ registration errors. (issue ASTERISK-20649) Reported by: Corey
+ Farrell Patches: cli-leaks-1_8-10.patch (license #5909) patch
+ uploaded by Corey Farrell cli-leaks-11-trunk.patch (license
+ #5909) patch uploaded by Corey Farrell Modified ........ Merged
+ revisions 377073 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377074 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/cdr.c, /: Cleanup CDR resources on exit. * Simplify
+ do_reload() return handling since it never returned anything
+ other than 0. (issue ASTERISK-20649) Reported by: Corey Farrell
+ Patches: cdr-cleanup-1_8.patch (license #5909) patch uploaded by
+ Corey Farrell cdr-cleanup-10-11-trunk.patch (license #5909) patch
+ uploaded by Corey Farrell Modified ........ Merged revisions
+ 377069 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 377070 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, main/ccss.c: Fix CCSS CLI commands and logger level not
+ unregistered. (issue ASTERISK-20649) Reported by: Corey Farrell
+ Patches: ccss-cleanup-all.patch (license #5909) patch uploaded by
+ Corey Farrell ........ Merged revisions 377037 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 377038 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-03 14:54 +0000 [r377021] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_motif.c: Fix an RTP instance reference count leak
+ in chan_motif. When setting up an RTP instance the RTCP portion
+ of the instance keeps a reference to the instance itself. In
+ order to release this reference and stop RTCP the stop API call
+ must be called before destroying the instance. (closes issue
+ ASTERISK-20751) Reported by: joshoa
+
+2012-12-01 00:46 +0000 [r376983] Joshua Colp <jcolp@digium.com>
+
+ * configs/motif.conf.sample, channels/chan_motif.c: Tweak extension
+ used for incoming calls received on Motif. Based on feedback from
+ numerous individuals this patch tweaks incoming calls to first
+ look for an extension with the name of the endpoint. If no such
+ extension exists the call will silently fall back to the "s"
+ extension as it previously did.
+
+2012-11-30 21:35 +0000 [r376952] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/misdn/isdn_lib.c: chan_misdn: Fix sending
+ RELEASE_COMPLETE in response to SETUP. Fix sending a
+ RELEASE_COMPLETE in response to a SETUP if chan_misdn does not
+ have a B channel available to assign to the call. (closes issue
+ ABE-2869) Reported by: Guenther Kelleter Patches:
+ setup-reject_2.diff (license #6372) patch uploaded by Guenther
+ Kelleter Modified ........ Merged revision 376949 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 376950 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376951 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-30 17:07 +0000 [r376921] Sean Bright <sean@malleable.com>
+
+ * /, funcs/func_volume.c: Minor spelling fix to the VOLUME
+ documentation. ........ Merged revisions 376919 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376920 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-30 16:36 +0000 [r376917] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Fix potential crashes during SIP attended
+ transfers. The principal behind this patch is simple. During a
+ transfer, we manipulate channels that are owned by a separate
+ thread than the one we currently are running in, so it makes
+ sense that we need to grab a reference to the channels so that
+ they cannot disappear out from under us. In the wild, crashes
+ were sometimes seen when the transferring party would hang up the
+ call before the transfer target answered the call. The most
+ common place to see the crash occur was when attempting to send a
+ connected line update to the transferer channel. (closes issue
+ ASTERISK-20226) Reported by Jared Smith Patches:
+ ASTERISK-20226.patch uploaded by Mark Michelson (License #5049)
+ Tested by: Jared Smith ........ Merged revisions 376901 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376916 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-29 22:59 +0000 [r376866-376870] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_local.c, /: chan_local: Fix local_pvt ref leak in
+ local_devicestate(). Regression introduced by ASTERISK-20390 fix.
+ (closes issue ASTERISK-20769) Reported by: rmudgett Tested by:
+ rmudgett ........ Merged revisions 376868 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376869 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, channels/chan_sip.c: Fix compile error. (issue ASTERISK-20724)
+ ........ Merged revisions 376864 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376865 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-29 21:57 +0000 [r376836] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, channels/chan_sip.c: Improve Code Readability And Fix Setting
+ natdetected Flag For 1.8, 10, 11 and trunk we are are improving
+ the code readability. For 11 and trunk, auto nat detection was
+ added. The natdetected flag was being set to 1 when the host
+ address in the VIA header did not specifiy a port. This patch
+ fixes this by setting the port on the temporary sock address used
+ to SIP_STANDARD_PORT in order for the sock address comparison to
+ work properly. (closes issue ASTERISK-20724) Reported by: Michael
+ L. Young Patches: asterisk-20724-set-port-v2.diff uploaded by
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2206/ ........ Merged
+ revisions 376834 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376835 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-29 17:17 +0000 [r376822] Pedro Kiefer <pedro@kiefer.com.br>
+
+ * channels/chan_sip.c: Fix chan_sip websocket payload handling
+ Websocket by default doesn't return an ast_str for the payload
+ received. When converting it to an ast_str on chan_sip the last
+ character was being omitted, because ast_str functions expects
+ that the given length includes the trailing 0x00. payload_len
+ only has the actual string length without counting the trailing
+ zero. For most cases this passed unnoticed as most of SIP
+ messages ends with \r\n. (closes issue ASTERISK-20745) Reported
+ by: Iñaki Baz Castillo Review:
+ https://reviewboard.asterisk.org/r/2219/
+
+2012-11-29 00:46 +0000 [r376760-376790] Richard Mudgett <rmudgett@digium.com>
+
+ * main/asterisk.c, /, main/astmm.c: Add MALLOC_DEBUG atexit
+ unreleased malloc memory summary. * Adds the following CLI
+ commands to control MALLOC_DEBUG reporting of unreleased malloc
+ memory when Asterisk is shut down. memory atexit list on memory
+ atexit list off memory atexit summary byline memory atexit
+ summary byfunc memory atexit summary byfile memory atexit summary
+ off * Made check all remaining allocated region blocks atexit for
+ fence violations. * Increased the allocated region hash table
+ size by about three times. It still isn't large enough
+ considering the number of malloced blocks Asterisk uses. * Made
+ CLI "memory show allocations anomalies" use
+ regions_check_all_fences(). Review:
+ https://reviewboard.asterisk.org/r/2196/ ........ Merged
+ revisions 376788 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376789 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, main/astmm.c: Enhance MALLOC_DEBUG CLI commands. * Fixed CLI
+ "memory show allocations" misspelling of anomalies option. The
+ command will still accept the original misspelling. *
+ Miscellaneous tweaks to CLI "memory show allocations" command
+ output format. * Made CLI "memory show summary" summarize by line
+ number instead of by function if a filename is given. * Made CLI
+ "memory show summary" sort its output by filename or
+ function-name/line-number depending upon request. * Miscellaneous
+ tweaks to CLI "memory show summary" command output format.
+ ........ Merged revisions 376758 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376759 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-28 16:37 +0000 [r376727] Jonathan Rose <jrose@digium.com>
+
+ * main/manager.c, /: manager: Make challenge work with
+ allowmultiplelogin=no Prior to this patch, challenge would yield
+ a multiple logins error if used without providing the username
+ (which isn't really supposed to be an argument to challenge) if
+ allowmultiplelogin was set to no because allowmultiplelogin finds
+ a user with a zero length login name. This check is simply
+ disabled for the challenge action when the username is empty by
+ this patch. (closes issue ASTERISK-20677) Reported by: Vladimir
+ Patches: challenge_action_nomultiplelogin.diff uploaded by
+ Jonathan Rose (license 6182) ........ Merged revisions 376725
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 376726 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-28 00:08 +0000 [r376629-376690] Richard Mudgett <rmudgett@digium.com>
+
+ * main/pbx.c, /, UPGRADE.txt: Fix extension matching with the '-'
+ char. The '-' char is supposed to be ignored by the dialplan
+ extension matching. Unfortunately, it's treatment is not handled
+ consistently throughout the extension matching code. * Made the
+ old exten matching code consistently ignore '-' chars. * Made the
+ old exten matching code consistently handle case in the matching.
+ * Made ignore empty character sets. * Fixed ast_extension_cmp()
+ to return -1, 0, or 1 as documented. The only user of it in
+ pbx_lua.c was testing for -1. It was originally returning the
+ strcmp() value for less than which is not usually going to be -1.
+ * Fix character set sorting if the sets have the same number of
+ characters and start with the same character. Character set [0-9]
+ now sorts before [02-9a] as originally intended. * Updated some
+ extension label and priority already in use warnings to also
+ indicate if the extension is aliased. (closes issue
+ ASTERISK-19205) Reported by: Philippe Lindheimer, Birger "WIMPy"
+ Harzenetter Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/2201/ ........ Merged
+ revisions 376688 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376689 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * addons/res_config_mysql.c, /, apps/app_celgenuserevent.c,
+ pbx/pbx_dundi.c: Remove unnecessary channel module references. *
+ Removed call to ast_module_user_hangup_all() in
+ res_config_mysql.c since it is effectively a noop. No channels
+ can attach a reference to that module. * Removed call to
+ ast_module_user_hangup_all() in app_celgenuserevent.c. The caller
+ of unload_module() has already called it. * Removed redundant
+ channel module references in pbx_dundi.c. The registered dialplan
+ function callback dispatchers for the read/read2/write callbacks
+ already reference the module before calling. * pbx_dundi: Moved
+ unregistering CLI commands, DUNDi switch, and dialplan functions
+ to the first thing the unload_module() does. This will reduce the
+ chance of new channels using DUNDi services while the module is
+ being torn down. ........ Merged revisions 376657 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376658 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, include/asterisk/linkedlists.h: Made AST_LIST_REMOVE() simpler
+ and use better names. * Update doxygen of AST_LIST_REMOVE().
+ ........ Merged revisions 376627 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376628 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-22 23:58 +0000 [r376588] Matthew Jordan <mjordan@digium.com>
+
+ * main/lock.c, /, main/logger.c, include/asterisk/lock.h:
+ Re-initialize logmsgs mutex upon logger initialization to prevent
+ lock errors Similar to the patch that moved the fork earlier in
+ the startup sequence to prevent mutex errors in the recursive
+ mutex surrounding the read/write thread registration lock, this
+ patch re-initializes the logmsgs mutex. Part of the start up
+ sequence before forking the process into the background includes
+ reading asterisk.conf; this has to occur prior to the call to
+ daemon in order to read startup parameters. When reading in a
+ conf file, log statements can be generated. Since this can't be
+ avoided, the mutex instead is re-initialized to ensure a reset of
+ any thread tracking information. This patch also includes some
+ additional debugging to catch errors when locking or unlocking
+ the recursive mutex that surrounds locks when the DEBUG_THREADS
+ build option is enabled. DO_CRASH or THREAD_CRASH will cause an
+ abort() if a mutex error is detected. (issue ASTERISK-19463)
+ Reported by: mjordan Tesetd by: mjordan ........ Merged revisions
+ 376586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 376587 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-20 21:58 +0000 [r376561] David M. Lee <dlee@digium.com>
+
+ * res/res_http_websocket.c: Added missing newlines to websocket
+ ast_logs. Without these newlines, log messages just continue
+ tacking onto the same line, and do not flush immediately.
+
+2012-11-20 18:57 +0000 [r376550] Mark Michelson <mmichelson@digium.com>
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c: Add "Require:
+ timer" to 200 OK responses when appropriate. The method by which
+ the Require header is added to 200 responses is inspired by the
+ method that Olle Johansson uses in his darjeeling-prack branch.
+ (closes issue ASTERISK-20570) Reported by Matt Jordan, at the
+ behest of Olle Johansson Review:
+ https://reviewboard.asterisk.org/r/2172 ........ Merged revisions
+ 376521 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 376522 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-20 17:37 +0000 [r376540] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/chan_sip.c: Reduce CLI spam of "Extension Changed"
+ device state messages. Asterisk 11 follows RFC3265 that states
+ that after every subscribe or resubscribe a notify should be
+ sent. Thus the console if filled continuously with the following
+ after every subscribe; == Extension Changed 8512[phones] new
+ state IDLE for Notify User cisco1 In Asterisk 1.8 only changes
+ would be sent. Thus only when a device state changed was anything
+ emitted to the console. fix: Only print to console when device
+ state isn't forced. (closes issue ASTERISK-20706) Reported by:
+ alecdavis Tested by: alecdavis alecdavis (license 585)
+
+2012-11-19 19:54 +0000 [r376471] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, channels/chan_sip.c, main/security_events.c,
+ main/indications.c: Fix most leftover non-opaque ast_str uses.
+ Instead of calling str->str, one should use ast_str_buffer(str).
+ Same goes for str->used as ast_str_strlen(str) and str->len as
+ ast_str_size(str). Review:
+ https://reviewboard.asterisk.org/r/2198 ........ Merged revisions
+ 376469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 376470 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-18 20:22 +0000 [r376415-376441] Matthew Jordan <mjordan@digium.com>
+
+ * main/asterisk.c, /, main/utils.c: Reorder startup sequence to
+ prevent lockups when process is sent to background Although it is
+ very rare and timing dependent, the potential exists for the call
+ to 'daemon' to cause what appears to be a deadlock in Asterisk
+ during startup. This can occur when a recursive mutex is obtained
+ prior to the daemon call executing. Since daemon uses fork to
+ send the process into the background, any threading primitives
+ are unsafe to re-use after the call. Implementations of pthread
+ recursive mutexes are highly likely to store the thread
+ identifier of the thread that previously obtained the mutex. If
+ the mutex was locked prior to the fork, a subsequent unlock
+ operation will potentially fail as the thread identifier is no
+ longer valid. Since the mutex is still locked, all subsequent
+ attempts to grab the mutex by other threads will block. This
+ behavior exhibited itself most often when DEBUG_THREADS was
+ enabled, as this compile time option surrounds the mutexes in
+ Asterisk with another recursive mutex that protects the storage
+ of thread related information. This made it much more likely that
+ a recursive mutex would be obtained prior to daemon and unlocked
+ after the call. This patch does the following: a) It backports a
+ patch from Asterisk 11 that prevents the spawning of the
+ localtime monitoring thread. This thread is now spawned after
+ Asterisk has fully booted. b) It re-orders the startup sequence
+ to call daemon earlier during Asterisk startup. This limits the
+ potential of threading primitives being accessed by
+ initialization calls before daemon is called. c) It removes calls
+ to ast_verbose/ast_log/etc. prior to daemon being called.
+ Developers should send error messages directly to stderr prior to
+ daemon, as calls to ast_log may access recursive mutexes that
+ store thread related information. d) It reorganizes when thread
+ local storage is created for storing lock information during the
+ creation of threads. Prior to this patch, the read/write lock
+ protecting the list of threads in ast_register_thread would
+ utilize the lock in the thread local storage prior to it being
+ initialized; this patch prevents that. On a very related note,
+ this patch will *greatly* improve the stability of the Asterisk
+ Test Suite. Review: https://reviewboard.asterisk.org/r/2197
+ (closes issue ASTERISK-19463) Reported by: mjordan Tested by:
+ mjordan ........ Merged revisions 376428 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376431 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * apps/confbridge/conf_state.c, /: Add a test event that reports
+ changes in ConfBridge state This patch adds a test event to
+ ConfBridge that reports transitions between states in ConfBridge.
+ This is used by tests in the Asterisk Test Suite that verify
+ state changes based on the entering/leaving of conference
+ participants. ........ Merged revisions 376414 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-16 19:59 +0000 [r376391] Jonathan Rose <jrose@digium.com>
+
+ * res/res_monitor.c, /: monitor: prevent attempts to move/remove
+ recordings skipped with 'i' and 'o'. The i and o options for
+ monitor skip the input and output sides of a recording
+ respectively. This patch addresses a problem in those options
+ when monitor is called without specifying a specific filename
+ where monitor will try to move the recording that was skipped.
+ Since this usually doesn't exist when these options are used, it
+ would produce a warning when it does this in most cases, but it
+ is conceivable that there are use cases where this could result
+ in moving/removing a file unintentionally. (closes issue
+ ASTERISK-20641) Reported by: Jonathan Rose Review:
+ https://reviewboard.asterisk.org/r/2190/ ........ Merged
+ revisions 376389 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376390 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-16 00:09 +0000 [r376339-376343] David M. Lee <dlee@digium.com>
+
+ * /, utils/extconf.c: Fixed extconf.c breakage introduced in
+ r376306. To quote wdoekes: > Note that I'm not confirming
+ legitimacy of having that file in tree at > all. Is anyone using
+ aelparse/conf2ael? ........ Merged revisions 376340 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376342 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * utils/Makefile, tests/test_astobj2_thrash.c (added),
+ utils/utils.xml, /, utils/hashtest.c (removed),
+ tests/test_hashtab_thrash.c (added), utils/hashtest2.c (removed),
+ include/asterisk/hashtab.h: Migrate hashtest/hashtest2 to be unit
+ tests. Both hashtest and hashtest2 are manual testing apps that
+ thrash hash tables (hashtab and ao2 containers, respectively), by
+ spinning up several threads that randomly insert, delete, lookup
+ and iterate over the hash table. If the app doesn't crash, the
+ hash table probably passes the test. Those utils are not a part
+ of the typical Asterisk build, so they do not usually get
+ compiled. This all makes them less that useful. This patch
+ removes those manual test programs and replaces them with
+ Asterisk unit test modules (test_{hashtab,astobj2}_thrash.so). It
+ also attempts to make the tests more deterministic. * Rather than
+ spinning up some number of threads that operate on the hash table
+ randomly, spin up four threads that concurrenly add, remove,
+ lookup and iterate over the hash table. * Each thread checks the
+ state of the hash table both during and after execution, and
+ indicates a test failure if things are not as expected. * Each
+ thread times out after 60 seconds to prevent deadlocking the unit
+ test run. (closes issue ASTERISK-20505) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/2189/ ........ Merged
+ revisions 376306 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376315 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-15 23:03 +0000 [r376310] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_meetme.c: app_meetme: Fix channels lingering when
+ hung up under certain conditions Channels would get stuck and
+ MeetMe would repeatedly display an Unable to write frame to
+ channel error in the conf_run function if hung up during certain
+ sound prompts such as during user count announcements. This patch
+ fixes that by reintroducing a hangup check in the meetme's main
+ loop (also in conf_run). (closes issue ASTERISK-20486) Reported
+ by: Michael Cargile Review:
+ https://reviewboard.asterisk.org/r/2187/ Patches:
+ meetme_hangup_patch_ASTERISK-20486_v3.diff uploaded by Jonathan
+ Rose (license 6182) ........ Merged revisions 376307 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376308 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-15 02:08 +0000 [r376264] Rusty Newton <rnewton@digium.com>
+
+ * apps/app_voicemail.c, /: Patch to play correct sound file when a
+ voicemail's urgent status is removed We were attempting to play
+ "vm-urgent-removed", which didn't exist. Now we play
+ "vm-marked-nonurgent" which exists and is the correct sound file.
+ Previous behavior was silence and a warning on the CLI. (issue
+ ASTERISK-20280) (closes issue ASTERISK-20280) Reported by: Tomo
+ Takebe Tested by: Rusty Newton Patches: asterisk20280.patch
+ uploaded by Rusty Newton (license 5829) ........ Merged revisions
+ 376262 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 376263 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-14 19:53 +0000 [r376234] Richard Mudgett <rmudgett@digium.com>
+
+ * pbx/pbx_spool.c, /: Fix call files when astspooldir is relative.
+ Future dated call files are ignored when astspooldir is relative
+ to the current directory. The queue_file() assumed that the qdir
+ needed to be prepended if the given filename did not start with a
+ '/'. If astspooldir is relative it is not going to start from the
+ root directory obviously so it will not start with a '/'. The
+ filename used in queue_file() ultimately results in qdir
+ prepended multiple times. * Made queue_file() not prepend qdir if
+ the filename contains a '/'. (closes issue ASTERISK-20593)
+ Reported by: James Le Cuirot Patches:
+ 0004-Fix-future-call-files-from-relative-directories.patch
+ (license #6439) patch uploaded by James Le Cuirot ........ Merged
+ revisions 376232 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376233 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-13 18:48 +0000 [r376217] Brent Eagles <beagles@digium.com>
+
+ * main/channel.c, /: Patch to prevent stopping the active generator
+ when it is not the silence generator. This patch introduces an
+ internal helper function to safely check whether the current
+ generator is the one that is expected before deactivating it. The
+ current externally accessible ast_channel_stop_generator()
+ function has been modified to be implemented in terms of the new
+ function. (closes issue ASTERISK-19918) Reported by: Eduardo Abad
+ ........ Merged revisions 376199 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376208 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-12 20:45 +0000 [r376168] Joshua Colp <jcolp@digium.com>
+
+ * main/pbx.c, /: Properly check if the "Context" and "Extension"
+ headers are empty in a ShowDialPlan action. The code which
+ handles the ShowDialPlan action wrongly assumed that a non-NULL
+ return value from the function which retrieves headers from an
+ action indicates that the header has a value. This is incorrect
+ and the contents must be checked to see if they are blank.
+ (closes issue ASTERISK-20628) Reported by: jkroon Patches:
+ asterisk-showdialplan-incorrect-error.patch uploaded by jkroon
+ ........ Merged revisions 376166 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376167 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-12 20:16 +0000 [r376144] Michael L. Young <elgueromexicano@gmail.com>
+
+ * main/pbx.c, /: Fix Dynamic Hints Variable Substition - Underscore
+ Problem When adding a dynamic hint, if an extension contains an
+ underscore no variable subsitution is being performed. This patch
+ changes from checking if the extension contains an underscore to
+ checking if the extension begins with an underscore. (closes
+ issue ASTERISK-20639) Reported by: Steven T. Wheeler Tested by:
+ Steven T. Wheeler, Michael L. Young Patches:
+ asterisk-20639-dynamic-hint-underscore.diff uploaded by Michael
+ L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2188/ ........ Merged
+ revisions 376142 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376143 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-11 17:08 +0000 [r376130] Joshua Colp <jcolp@digium.com>
+
+ * res/res_rtp_asterisk.c, channels/chan_sip.c,
+ configs/sip.conf.sample: Remove a fixed size limitation for
+ producing SDP and change how ICE support is disabled by default.
+ With ICE support enabled in chan_sip and a large number of
+ interfaces on the system it was possible for the produced SDP to
+ be truncated due to some fixed size buffers. These buffers have
+ now been changed so they will dynamically grow as needed. ICE
+ support is now also enabled by default in res_rtp_asterisk to
+ provide a smoother experience for chan_motif users where it is
+ required. To maintain the previous behavior in chan_sip it is no
+ longer enabled by default there. (closes issue ASTERISK-20643)
+ Reported by: coopvr
+
+2012-11-08 22:08 +0000 [r376089] Mark Michelson <mmichelson@digium.com>
+
+ * /, res/res_fax.c: Fix a "set but not used" warning on newer gccs.
+ Turns out the "helpful" setting of ms and res in this macro is
+ completely useless after the timeout antipattern fix. If you're a
+ new guy looking to write code, don't write a macro like this one.
+ ........ Merged revisions 376087 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376088 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-08 21:10 +0000 [r376048-376060] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_ss7.c, /: chan_dahdi/SS7: Made reject incoming call
+ for an in-alarm or blocked channel. If a SS7 call comes in
+ requesting a CIC that is in-alarm, the call is accepted and
+ connects if the extension exists in the dialplan. The call does
+ not have any audio. * Made release the call immediately with
+ circuit congestion cause. (closes issue ASTERISK-20204) Reported
+ by: Tuan Le Patches: jira_asterisk_20204_v1.8.patch (license
+ #5621) patch uploaded by rmudgett ........ Merged revisions
+ 376058 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 376059 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/asterisk.c, include/asterisk/utils.h,
+ include/asterisk/astmm.h, /, main/utils.c, main/astmm.c: Add
+ MALLOC_DEBUG enhancements. * Makes malloc() behave like calloc().
+ It will return a memory block filled with 0x55. A nonzero value.
+ * Makes free() fill the released memory block and boundary
+ fence's with 0xdeaddead. Any pointer use after free is going to
+ have a pointer pointing to 0xdeaddead. The 0xdeaddead pointer is
+ usually an invalid memory address so a crash is expected. * Puts
+ the freed memory block into a circular array so it is not reused
+ immediately. * When the circular array rotates out a memory block
+ to the heap it checks that the memory has not been altered from
+ 0xdeaddead. * Made the astmm_log message wording better. * Made
+ crash if the DO_CRASH menuselect option is enabled and something
+ is found. * Fixed a potential alignment issue on 64 bit systems.
+ struct ast_region.data[] should now be aligned correctly for all
+ platforms. * Extracted region_check_fences() from
+ __ast_free_region() and handle_memory_show(). * Updated
+ handle_memory_show() CLI usage help. Review:
+ https://reviewboard.asterisk.org/r/2182/ ........ Merged
+ revisions 376029 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 376030 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-07 19:03 +0000 [r376014] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/time.h, apps/app_jack.c, apps/app_dial.c,
+ main/pbx.c, main/rtp_engine.c, /, apps/app_meetme.c,
+ res/res_fax.c, apps/app_record.c, channels/chan_agent.c,
+ main/utils.c, include/asterisk/channel.h, apps/app_queue.c,
+ channels/sig_pri.c, channels/chan_iax2.c, main/channel.c,
+ channels/chan_dahdi.c, apps/app_waitforring.c,
+ channels/sig_analog.c: Multiple revisions 375993-375994 ........
+ r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov
+ 2012) | 30 lines Fix misuses of timeouts throughout the code.
+ Prior to this change, a common method for determining if a
+ timeout was reached was to call a function such as
+ ast_waitfor_n() and inspect the out parameter that told how many
+ milliseconds were left, then use that as the input to
+ ast_waitfor_n() on the next go-around. The problem with this is
+ that in some cases, submillisecond timeouts can occur, resulting
+ in the out parameter not decreasing any. When this happens
+ thousands of times, the result is that the timeout takes much
+ longer than intended to be reached. As an example, I had a
+ situation where a 3 second timeout took multiple days to finally
+ end since most wakeups from ast_waitfor_n() were under a
+ millisecond. This patch seeks to fix this pattern throughout the
+ code. Now we log the time when an operation began and find the
+ difference in wall clock time between now and when the event
+ started. This means that sub-millisecond timeouts now cannot play
+ havoc when trying to determine if something has timed out. Part
+ of this fix also includes changing the function ast_waitfor() so
+ that it is possible for it to return less than zero when a
+ negative timeout is given to it. This makes it actually possible
+ to detect errors in ast_waitfor() when there is no timeout.
+ (closes issue ASTERISK-20414) reported by David M. Lee Review:
+ https://reviewboard.asterisk.org/r/2135/ ........ r375994 |
+ mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3
+ lines Remove some debugging that accidentally made it in the last
+ commit. ........ Merged revisions 375993-375994 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375995 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-06 18:59 +0000 [r375966] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/features.h, main/channel.c, /,
+ main/channel_internal_api.c, main/features.c,
+ include/asterisk/channel.h: Fix stuck DTMF when bridge is broken.
+ When a bridge is broken by an AMI Redirect action or the
+ ChannelRedirect application, an in progress DTMF digit could be
+ stuck sending forever. * Made simulate a DTMF end event when a
+ bridge is broken and a DTMF digit was in progress. (closes issue
+ ASTERISK-20492) Reported by: Jeremiah Gowdy Patches:
+ bridge_end_dtmf-v3.patch.txt (license #6358) patch uploaded by
+ Jeremiah Gowdy Modified to jira_asterisk_20492_v1.8.patch
+ jira_asterisk_20492_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/2169/ ........ Merged
+ revisions 375964 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375965 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-12-10 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 11.1.0 Released.
+
+2012-12-06 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 11.1.0-rc3 Released.
+
+ * chan_local: Fix local_pvt ref leak in local_devicestate().
+
+ Regression introduced by ASTERISK-20390 fix.
+
+ (closes issue ASTERISK-20769)
+ Reported by: rmudgett
+
+2012-12-05 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 11.1.0-rc2 Released.
+
+ * Fix a SIP request memory leak with TLS connections.
+
+ During the TLS re-work in chan_sip some TLS specific code was moved
+ into a separate function. This function operates on a copy of the
+ incoming SIP request. This copy was never deinitialized causing a
+ memory leak for each request processed.
+
+ This function is now given a SIP request structure which it can use
+ to copy the incoming request into. This reduces the amount of memory
+ allocations done since the internal allocated components are reused
+ between packets and also ensures the SIP request structure is
+ deinitialized when the TLS connection is torn down.
+
+ (closes issue ASTERISK-20763)
+ Reported by: deti
+
+2012-11-06 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 11.1.0-rc1 Released.
+
+2012-11-06 12:09 +0000 [r375925] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_motif.c: Fix a bug where our Motif ICE candidates
+ were not quite proper, and make us more forgiving. An issue was
+ reported on the mailing list where calling would result in an
+ "Incomplete ICE-UDP candidate received on session" error message.
+ This is the result of the ICE-UDP candidate code not placing a
+ "network" attribute within the candidates. This is now done. To
+ increase compatibility though I have removed the requirement for
+ the "network" attribute to exist within ICE-UDP candidates that
+ are received since we don't actually require the value. Reported
+ on the mailing list by Jean-Denis Girard.
+
+2012-11-05 23:09 +0000 [r375895] Matthew Jordan <mjordan@digium.com>
+
+ * main/timing.c, main/channel.c, /, res/res_timing_pthread.c,
+ res/res_timing_dahdi.c, res/res_timing_timerfd.c,
+ bridges/bridge_softmix.c, funcs/func_jitterbuffer.c,
+ include/asterisk/timing.h, res/res_musiconhold.c,
+ channels/chan_iax2.c, res/res_fax_spandsp.c,
+ res/res_timing_kqueue.c: Refactor ast_timer_ack to return an
+ error and handle the error in timer users Currently, if an
+ acknowledgement of a timer fails Asterisk will not realize that a
+ serious error occurred and will continue attempting to use the
+ timer's file descriptor. This can lead to situations where errors
+ stream to the CLI/log file. This consumes significant resources,
+ masks the actual problem that occurred (whatever caused the timer
+ to fail in the first place), and can leave channels in odd
+ states. This patch propagates the errors in the timing resource
+ modules up through the timer core, and makes users of these
+ timers handle acknowledgement failures. It also adds some
+ defensive coding around the use of timers to prevent using bad
+ file descriptors in off nominal code paths. Note that the patch
+ created by the issue reporter was modified slightly for this
+ commit and backported to 1.8, as it was originally written for
+ Asterisk 10. Review: https://reviewboard.asterisk.org/r/2178/
+ (issue ASTERISK-20032) Reported by: Jeremiah Gowdy patches:
+ jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license
+ 6358) ........ Merged revisions 375893 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375894 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-05 21:41 +0000 [r375864] Richard Mudgett <rmudgett@digium.com>
+
+ * main/loader.c, /: Add safety NULL pointer check in module user
+ references. Made __ast_module_user_remove() check for NULL
+ pointers. ........ Merged revision 375860 from C.3 ........
+ Merged revisions 375862 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375863 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-05 17:59 +0000 [r375847] Jonathan Rose <jrose@digium.com>
+
+ * /, UPGRADE.txt: chan_sip: Document a change to user-field
+ encoding introduced with r303509 The change in question was added
+ to improve compliance with RFC3261, but at the time of commit, it
+ wasn't adequately documented in the UPGRADE notes. (closes issue
+ ASTERISK-20561) Reported by: Deniz Review:
+ https://reviewboard.asterisk.org/r/2177/ ........ Merged
+ revisions 375846 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-04 03:09 +0000 [r375729-375802] Matthew Jordan <mjordan@digium.com>
+
+ * main/manager.c, /: Don't attempt to purge sessions when no
+ sessions exist Manager's tcp/tls objects have a periodic function
+ that purge old manager sessions periodically. During shutdown,
+ the underlying container holding those sessions can be disposed
+ of and set to NULL before the tcp/tls periodic function is
+ stopped. If the periodic function fires, it will attempt to
+ iterate over a NULL container. This patch checks for whether or
+ not the sessions container exists before attempting to purge
+ sessions out of it. If the sessions container is NULL, we simply
+ return. Note that this error was also caught by the Asterisk Test
+ Suite. ........ Merged revisions 375800 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375801 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, res/res_fax.c: Only deref a reserved gateway session if we
+ actually reserved one Its perfectly acceptable to have a gateway
+ session unreserved when we go to first allocate one. Unreffing
+ the reserved gateway session - when its NULL - will result in an
+ assertion error. This problem was caught by the Asterisk Test
+ Suite (once we had enough of the debugging flags enabled)
+ ........ Merged revisions 375797 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/manager.c, /: Properly clean up manager resources on exit
+ This patch does two things: 1) It properly unregisters the
+ manager CLI commands 2) It cleans up AMI users on exit. Prior to
+ this patch, the AMI users were not being disposed of properly,
+ resulting in a memory leak. (closes issue ASTERISK-20646)
+ Reported by: Corey Farrell patches: manager_shutdown.patch
+ uploaded by Corey Farrell (license 5909) ........ Merged
+ revisions 375793 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375794 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/db.c, /: Properly finalize prepared SQLite3 statements to
+ prevent memory leak The AstDB uses prepared SQLite3 statements to
+ retrieve data from the SQLite3 database. These statements should
+ be finalized during Asterisk shutdown so that the SQLite3
+ database can be properly closed. Failure to finalize the
+ statements results in a memory leak and a failure when closing
+ the database. This patch fixes those issues by ensuring that all
+ prepared statements are properly finalized at shutdown. (closes
+ issue ASTERISK-20647) Reported by: Corey Farrell patches:
+ astdb-sqlite3_close.patch uploaded by Corey Farrell (license
+ 5909) ........ Merged revisions 375761 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/xmldoc.c: Fix memory leaks in XML documentation This patch
+ fixes two memory leaks: 1) When building XML documentation items,
+ the 'name' attribute was extracted from XML elements but not
+ properly freed after being copied into the item being built. 2)
+ When unloading XML documentation, the doctree container objects
+ were not properly freed. This patch corrects these memory leaks.
+ Note that this patch was modified slightly for this commmit, as
+ the case where the 'name' attribute doesn't exist also wasn't
+ handled in the item construction. This patch also checks for that
+ attribute not existing. (closes issue ASTERISK-20648) Reported
+ by: Corey Farrell Tested by: mjordan patches:
+ xmldoc-memory_leak.patch uploaded by Corey Farrell (license 5909)
+
+ * main/cdr.c, /: Prevent multiple CDR batches from conflicting when
+ scheduling the CDR write The Asterisk Test Suite caught an error
+ condition where a scheduled CDR batch write can be deleted twice
+ if two channels attempt to post their CDRs at the same time. The
+ batch CDR mutex is locked while the CDRs are appended to the
+ current batch list; however, it is unlocked prior to actually
+ scheduling the CDR write. As such, two threads can attempt to
+ remove the currently scheduled batch write at the same time,
+ resulting in an assertion error. This patch extends the time that
+ the mutex is locked to encompass actually scheduling the write.
+ This prevents two threads from unscheduling the currently
+ scheduled write at the same time. ........ Merged revisions
+ 375727 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 375728 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-03 03:17 +0000 [r375702] Andrew Latham <lathama@gmail.com>
+
+ * README, include/asterisk/doxyref.h: Doxygen Updates Replace links
+ to missing text files removed in the 1.6.x series with links to
+ the wiki. Doxygen can handle URLs fine, don't atempt to quote
+ them. Also update the wiki link in the Readme to get everyone on
+ the same page. (issue ASTERISK-20259) ........ Merged revisions
+ 375698 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 375699 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-02 20:59 +0000 [r375661] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, channels/chan_misdn.c, /, main/ccss.c,
+ main/format_pref.c: Things don't need to be that const. ........
+ Merged revisions 375658 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375659 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-02 20:56 +0000 [r375660] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Fix for chan_skinny leaving RTP ports
+ open Skinny wasn't closing RTP sockets. This patch includes
+ ast_rtp_instance_stop before ast_rtp_instance_destroy which fixes
+ the problem. Also add destroy for VRTP (which I believe is
+ unused, but exists). Review:
+ https://reviewboard.asterisk.org/r/2176/
+
+2012-11-02 18:44 +0000 [r375627] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.h, /, channels/misdn/isdn_lib.c: Multiple
+ revisions 375519-375524 ........ r375519 | rmudgett | 2012-10-30
+ 16:06:15 -0500 (Tue, 30 Oct 2012) | 11 lines chan_misdn: Timer
+ primitives must be handled first. The frm->addr is a different
+ "address space" than the stack/instance address of other Lx
+ primitives. The test for B channel instance address could fail.
+ Patches: patch01_timers.diff (license #6372) patch uploaded by
+ Guenther Kelleter JIRA ABE-2888 ........ r375520 | rmudgett |
+ 2012-10-30 16:14:58 -0500 (Tue, 30 Oct 2012) | 10 lines
+ chan_misdn: Free memory in error paths and other memory leaks.
+ The one line commented with BUG is not easily fixable because
+ there is no de-init function one can call. Patches:
+ patch02_memory.diff (license #6372) patch uploaded by Guenther
+ Kelleter JIRA ABE-2888 ........ r375521 | rmudgett | 2012-10-30
+ 16:38:41 -0500 (Tue, 30 Oct 2012) | 14 lines chan_misdn: ISDN NT
+ L2 de-establish/establish * An NT-PTMP cannot de/establish L2
+ since it doesn't know the TEIs. * On NT-PTP L2 is started when L1
+ is finally active in handle_l1. * L2 deactivation logging
+ cleanup. * L2 aggregate link status is unknown for NT-PTMP, show
+ as "UNKN". * Removed unused functions and code for L2 handling.
+ Patches: patch03_L2estab.diff (license #6372) patch uploaded by
+ Guenther Kelleter Modified JIRA ABE-2888 ........ r375522 |
+ rmudgett | 2012-10-30 16:56:14 -0500 (Tue, 30 Oct 2012) | 22
+ lines chan_misdn: Fix broken upper_id/lower_id usage. Sending PH
+ prim via lower_id layer (3 or 1) simply does not work. For TE (3)
+ it returns an error (len=-6) which is not evaluated by
+ handle_l1(), so the L1 layer status ends up wrong. Instead PH
+ must be sent via L4, only then does it reach L1 without an error
+ message. And NT PH prims only reach L1 when they are sent to
+ layer 2 id. --> use upper_id to send PH primitives. * Check for
+ errors in PH_(DE)ACTIVATE | CONFIRM. * Debug messages are
+ improved. * The lower_id is now not used for anything, except:
+ Why is lower_id layer deleted when it wasn't created? I removed
+ this code since it looks very wrong. Patches:
+ patch04_l1activation.diff (license #6372) patch uploaded by
+ Guenther Kelleter JIRA ABE-2888 ........ r375523 | rmudgett |
+ 2012-10-30 17:29:15 -0500 (Tue, 30 Oct 2012) | 31 lines
+ chan_misdn: Fix loss of B channels if L1 is down. If you make 2
+ calls out an NT PTMP port which is not connected to any phone,
+ the B channel associated with that call becomes unusable until
+ Asterisk is restarted. The problem is the EVENT_SETUP is queued
+ when L1 is not up in misdn_lib_send_event(). If L1 cannot be
+ activated the event won't be dequeued. It gets even worse when
+ the call is hung up. The queued EVENT_SETUP will be overwritten
+ by an EVENT_DISCONNECT. The reserved B channel then will never be
+ freed. If later someone connects a phone to the port, L1 will
+ eventually activate and the queued EVENT_DISCONNECT is sent down
+ the stack. However, it is ignored because it is the wrong call
+ state. The real fix would be that activation and queueing for a
+ new SETUP is done by the NT stack. But since it doesn't, the
+ workaround must be removed because it doesn't always work. Fix:
+ The event is no longer queued but immediately sent to the stack.
+ If L1 cannot be activated, the L3 state machine that was started
+ by the EVENT_SETUP will do its work, i.e. a timeout will release
+ the B channel properly. The SETUP possibly cannot be sent the
+ first time but is resent by T303 in case L1 could be activated.
+ Patches: patch05_bchan-loss.diff (license #6372) patch uploaded
+ by Guenther Kelleter Modified JIRA ABE-2888 ........ r375524 |
+ rmudgett | 2012-10-30 18:26:05 -0500 (Tue, 30 Oct 2012) | 13
+ lines chan_misdn: Remove some calls to exit(). Try proper cleanup
+ when something goes wrong in misdn_lib_init(). Especially do not
+ call exit()! * Fix memory leak because stack_destroy() does not
+ free the stack struct. Patches: patch06_cleanup-init.diff
+ (license #6372) patch uploaded by Guenther Kelleter Modified JIRA
+ ABE-2888 ........ Merged revisions 375519-375524 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 375625 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375626 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-02 17:24 +0000 [r375613] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, channels/chan_sip.c: Fix Wrong Result In Debug Message For SDP
+ Origin Processing While looking at some debug logs, I noticed
+ that it was being reported that the SDP origin line was
+ unsupported or failed. Upon looking into this on my local
+ machine, I found that I too was getting this debug message yet
+ everything seemed to be getting processed properly. What was
+ discovered is, that, the variable to determine what is displayed
+ in the debug message for the SDP line that was processed, was not
+ being set for the origin line when the result was successful.
+ This patch fixes this and was tested on local machine. ........
+ Merged revisions 375594 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375601 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-01 14:52 +0000 [r375575] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: chan_sip: Fix a bug
+ causing SIP reloads to remove all entries from the registry A
+ regression was introduced in chan_sip by changes to sip reload
+ introduced by r349097. That patch moved peer purging from the
+ beginning of the reload to after the general configuration was
+ finished. This patch fixes that by undoing the repositioning of
+ the original peer purging code and using a similar function after
+ performing general configuration that purges only autocreated
+ peers that were created when persist mode isn't enabled. (closes
+ issue ASTERISK-20611) Reported by: Alisher Review:
+ https://reviewboard.asterisk.org/r/2171/
+
+2012-10-31 18:00 +0000 [r375559] Joshua Colp <jcolp@digium.com>
+
+ * res/res_http_websocket.exports.in: Fix an issue with
+ res_http_websocket where the chan_sip WebSocket handler could not
+ be registered. On some systems the optional API support uses the
+ GCC compiler attribute "weakref" to provide its functionality.
+ This code changes the function names and prefixes "__" to the
+ front. The res_http_websocket exports file did not take this into
+ account, thereby not allowing those functions to be global and
+ ultimately found. (closes issue ASTERISK-20631) Reported by:
+ danjenkins
+
+2012-10-31 14:49 +0000 [r375532] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_calendar_ews.c, /: Properly extract the Body information
+ of an EWS calendar item Unlike all other calendar modules,
+ res_calendar_ews fails to extract the Body information for a
+ calendar item. This is due, in part, to a quirk in the schema in
+ the XML - not only does a CalendarItem contain a Body element,
+ but the CalendarItem exists as a descendant of a different Body
+ element. The neon parser was erroneously skipping all Body
+ elements. This patch fixes that by bypassing Body elements that
+ are not a child of CalendarItem, and parsing the Body element out
+ if it is a child. Note that the original patch by Terry Wilson
+ only needed slight modifications to make it properly pull the
+ Body information out; as such, while I've linked to the patch
+ that I uploaded for Dmitry, I've attributed the patch to Terry.
+ (closes issue ASTERISK-19738) Reported by: Dmitry Burilov Tested
+ by: Dmitry Burilov patches: calendar_ews_body_2012_10_29.diff
+ uploaded by Terry Wilson (license 6283) ........ Merged revisions
+ 375528 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 375531 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-30 19:23 +0000 [r375506] Richard Mudgett <rmudgett@digium.com>
+
+ * /, bridges/bridge_softmix.c: Fix ConfBridge crash if no timing
+ module loaded. (closes issue ASTERISK-19448) Reported by: feyfre
+ Patches: smfix.patch (license #6099) patch uploaded by feyfre
+ Modified for coding guidelines. ........ Merged revisions 375496
+ from http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-30 19:09 +0000 [r375471-375486] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_mixmonitor.c: mixmonitor: Add a test event This test
+ event is being used to fix the mixmonitor_audiohook_inherit test.
+ ........ Merged revisions 375484 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375485 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, apps/app_confbridge.c: confbridge: Fix a bug which made
+ conferences not record with AMI/CLI commands When confbridge was
+ changed to handle conference status with a state machine in
+ r374658. The function responsible for starting recording for a
+ conference was refactored with the function actually responsible
+ for launching the recording thread being split into a function
+ with another name. The old function name was still used for
+ manually started recordings through AMI or CLI. This patch fixes
+ that by switching which function is used to start recording the
+ conference. (closes issue ASTERISK-20601) Reported by: Vilius
+ Patches: confbridge_mixmonitor.diff uploaded by Jonathan Rose
+ (license 6182) ........ Merged revisions 375470 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-30 02:22 +0000 [r375469] Matthew Jordan <mjordan@digium.com>
+
+ * /, apps/app_queue.c: Ensure that the Queue application tracks
+ busy members in off nominal situations There are a few code paths
+ where the Queue application fails to count a paused or in use
+ queue member as being 'busy'. This can cause callers to get stuck
+ in the Queue until a paused agent unpauses themselves. (closes
+ issue ASTERISK-20623) Reported by: Bryan Walters patches:
+ app_queue.patch uploaded by Bryan Walters (license 5851) ........
+ Merged revisions 375450 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375451 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-29 21:23 +0000 [r375437] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Prevent resetting of NATted realtime peer
+ address on reload. If a "sip reload" is issued for a SIP peer,
+ then his IP address will be cleared, thus resulting in forgetting
+ the public IP address. Asterisk will then attempt to route SIP
+ traffic to the private IP address. The fix here is to make "sip
+ reload" ignore realtime peers when "host = dynamic" is spotted.
+ Realtime peers can now only have their IP address reset if they
+ have gone from being not dynamic to being dynamic. (closes issue
+ ASTERISK-18203) reported by daren ferreira (closes issue
+ ASTERISK-20572) reported by JoshE Patches: fix_nat_realtime.diff
+ uploaded by JoshE (license #6075) ........ Merged revisions
+ 375415 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 375417 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-29 19:29 +0000 [r375363-375390] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Fix the Park 'r' option when a channel parks
+ itself. When a channel uses the Park appliation to park itself
+ with the 'r' option, the channel hears music-on-hold instead of
+ the requested ringing. * Added a missing check for the 'r' option
+ when a channel parks itself. (closes issue ASTERISK-19382)
+ Reported by: James Stocks Patches by: dsessions Review:
+ https://reviewboard.asterisk.org/r/2148/ ........ Merged
+ revisions 375388 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375389 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/chan_dahdi.c, /: chan_dahdi: Fix segfault dereferencing
+ a NULL tech_pvt. The tech support customer was using the AMI
+ Redirect action shortly after a call was placed. While the
+ channel tried to do an ast_read(), the masquerade resulting from
+ the channel redirect took place. The masquerade in the middle of
+ the ast_read() resulted in the segfault. (closes issue AST-1025)
+ Reported by: Trey Blancher Patches: jira_ast_1025_v1.8_v2.patch
+ (license #5621) patch uploaded by rmudgett ........ Merged
+ revisions 375361 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375362 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-23 16:22 +0000 [r375288-375327] Jonathan Rose <jrose@digium.com>
+
+ * contrib/scripts/ast_tls_cert, /: ast_tls_cert script: Better
+ response for various exit conditions to openssl (closes issue
+ ASTERISK-20260) Reported by: Daniel O'Connor Patches:
+ ast_tls_cert-update.diff uploaded by Daniel O'Connor (license
+ 6419) ........ Merged revisions 375325 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375326 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, main/app.c: core: Fix a memory leak in app.c from an early
+ return ast_app_group_match_get_count allocates memory with the
+ regcomp function and we previously forgot to free it when bailing
+ out due to a regex compilation failure against category. (closes
+ issue AST-1018) Reported by: Guenther Kelleter Patches:
+ regcomp_memleak.diff uploaded by Guenther Kelleter (license 6372)
+ ........ Merged revisions 375299 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375300 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, codecs/gsm/src/code.c: GSM: Fix encoding problems with GSM
+ (closes issue ASTERISK-20457) Reported by: Richard Miller
+ Patches: code.patch uploaded by Richard Miller (license 5685)
+ ........ Merged revisions 375272 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375273 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-18 21:44 +0000 [r375219-375247] Jonathan Rose <jrose@digium.com>
+
+ * UPGRADE.txt: app_queue: add upgrade notes for 375216 Adds UPGRADE
+ notes describing behavioral changes to rrmemory strategy caused
+ by 375216 (issue AST-989) Reported by: Thomas Arimont
+
+ * /, apps/app_queue.c: app_queue: Make ordering of
+ rrmemory/rrordered persist over add/remove members Prior to this
+ patch, adding, removing or reloading members to rrmemory would
+ cause the order to become completely jumbled. Now it behaves more
+ or less like rrordered other than the fact that it stores the
+ members on a hash table rather than a linked list. This patch
+ also prevents removal of members and member reloads from jumbling
+ rrordered queues. (issue AST-989) Reported by: Thomas Arimont
+ Review: https://reviewboard.asterisk.org/r/2164/ ........ Merged
+ revisions 375216 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375217 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-18 20:02 +0000 [r375191] Richard Mudgett <rmudgett@digium.com>
+
+ * Makefile, /, build_tools/make_version, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, makeopts.in:
+ build_tools: Allow Asterisk to report git SHAs in version string.
+ Make git more attractive for managing work-in-progress.
+ Especially convenient when a potential patch set needs to be
+ tested on multiple platforms since one can use git to keep all
+ the test environments in sync independent of a subversion server.
+ Now the Asterisk version will show the exact git SHA5 that was
+ used when building (still appended by "M" if there are local
+ modifications) from a git clone of the Asterisk repository so the
+ developer can more easily know what is actually under test. You
+ will now get this: $ asterisk -V Asterisk GIT-1698298 Instead of
+ this: $ asterisk -V Asterisk UNKNOWN__and_probably_unsupported
+ This has zero impact for those not using git with the exception
+ of an extra test in the configure script to gather git's path.
+ This is necessary to prevent "sudo make install" from failing
+ since git may not be in the path in make's shell environment.
+ (closes issue ASTERISK-20483) Reported by: Shaun Ruffell Patches:
+ 0001-build_tools-Allow-Asterisk-to-report-git-SHAs-in-ver.patch
+ (license #5417) patch uploaded by Shaun Ruffell Modified ........
+ Merged revisions 375189 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375190 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-17 19:00 +0000 [r375148] Kinsey Moore <kmoore@digium.com>
+
+ * main/tcptls.c, /: Ensure Asterisk fails TCP/TLS SIP calls when
+ certificate checking fails When placing a call to a TCP/TLS SIP
+ endpoint whose certificate is not signed by a configured CA
+ certificate, Asterisk would issue a warning and continue to
+ process the call as if there was not an issue with the
+ certificate. Asterisk now properly fails the call if the
+ certificate fails verification or if the certificate does not
+ exist when certificate checking is enabled (the default
+ behavior). (closes issue ASTERISK-20559) Reported by: kmoore
+ Review: https://reviewboard.asterisk.org/r/2163/ ........ Merged
+ revisions 375146 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375147 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-16 21:44 +0000 [r375079-375113] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, channels/chan_sip.c: Fixes to the fd-oriented SIP TCP reads.
+ Don't crash on large user input. Allow SIP headers without space.
+ Optimize code a bit. Review:
+ https://reviewboard.asterisk.org/r/2162 ........ Merged revisions
+ 375111 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 375112 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, channels/chan_sip.c: Update sip_request_call SIP dial string
+ documentation. This was missed when merging review r1859.
+ ........ Merged revisions 375074 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375078 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-16 14:08 +0000 [r375051] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: Remove a log message that was left in
+ accidentally from call-id logging development.
+
+2012-10-15 21:15 +0000 [r375027] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, /, main/ccss.c, include/asterisk/strings.h,
+ channels/chan_iax2.c: Fix some potential misuses of ast_str in
+ the code. Passing an ast_str pointer by value that then calls
+ ast_str_set(), ast_str_set_va(), ast_str_append(), or
+ ast_str_append_va() can result in the pointer originally passed
+ by value being invalidated if the ast_str had to be reallocated.
+ This fixes places in the code that do this. Only the example in
+ ccss.c could result in pointer invalidation though since the
+ other cases use a stack-allocated ast_str and cannot be
+ reallocated. I've also updated the doxygen in strings.h to
+ include notes about potential misuse of the functions mentioned
+ previously. Review: https://reviewboard.asterisk.org/r/2161
+ ........ Merged revisions 375025 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375026 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-15 08:11 +0000 [r375016] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * channels/chan_unistim.c: Fix underscreen buttons warnings apeared
+ while transfer process
+
+2012-10-14 11:57 +0000 [r374995] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * config.guess, config.sub, /: Update config.guess and config.sub:
+ 2012-10-10 Update config.guess and config.sub to revision
+ fb456b34ef4aa02b95dc6be69aaa66fa94a844fb from the
+ savannah.gnu.org git repo. Adds support for e.g. aarch64 (ARM
+ 64bit). config.guess:timestamp='2012-09-25'
+ config.sub:timestamp='2012-10-10' ........ Merged revisions
+ 374977 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 374991 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-12 21:57 +0000 [r374932] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_voicemail.c: Avoid a segfault on invalid format names If
+ a format name was not found by ast_getformatbyname, a NULL
+ pointer would be passed into ast_format_rate and immediately
+ dereferenced. This ensures that a valid pointer is used since the
+ structure is already allocated on the stack. (closes issue
+ DPH-523) Reported-by: Steve Pitts
+
+2012-10-12 16:20 +0000 [r374914] Mark Michelson <mmichelson@digium.com>
+
+ * main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h:
+ Do not use a FILE handle when doing SIP TCP reads. This is used
+ to solve an issue where a poll on a file descriptor does not
+ necessarily correspond to the readiness of a FILE handle to be
+ read. This change makes it so that for TCP connections, we do a
+ recv() on the file descriptor instead. Because TCP does not
+ guarantee that an entire message or even just one single message
+ will arrive during a read, a loop has been introduced to ensure
+ that we only attempt to handle a single message at a time. The
+ tcptls_session_instance structure has also had an overflow buffer
+ added to it so that if more than one TCP message arrives in one
+ go, there is a place to throw the excess. Huge thanks goes out to
+ Walter Doekes for doing extensive review on this change and
+ finding edge cases where code could fail. (closes issue
+ ASTERISK-20212) reported by Phil Ciccone Review:
+ https://reviewboard.asterisk.org/r/2123 ........ Merged revisions
+ 374905 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 374906 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-11 21:18 +0000 [r374850-374877] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_motif.c: Fix a bug where audio on Google Voice
+ would not work due to ignoring candidates. Instead of ignoring
+ parts of the message that are not known just ignore the ones we
+ know may be present and that would cause a problem.
+
+ * res/res_rtp_asterisk.c: Remove code that should not have gotten
+ in. (issue ASTERISK-20554)
+
+ * res/res_rtp_asterisk.c, channels/chan_motif.c: Fix an issue where
+ outgoing calls would fail to establish audio due to ICE
+ negotiation failures. This change removes the requirement for
+ ufrag and pwd in the transport stanza and also makes us the
+ controlling agent. (closes issue ASTERISK-20554) Reported by:
+ mmichelson
+
+2012-10-11 15:44 +0000 [r374845] Matthew Jordan <mjordan@digium.com>
+
+ * main/cdr.c, /: Fix incorrect billing duration reported when batch
+ mode is enabled Similar to r369351, the billing duration can be
+ skewed when batch mode is enabled. This happened much more rarely
+ than the duration, as it only occured when the call was answered
+ (thereby indicating an actual answer time) and immediately hung
+ up on (indicating a billsec of 0). Since a billing time of '0'
+ can either mean that the call immediately ended or that the CDR
+ was improperly answered, we have to use additional information to
+ know whether or not we can trust the CDR billsec value. Prior to
+ this patch, we looked to see if we had a valid answer time. If we
+ did, and billsec was zero, we used the current time to calculate
+ what billsec value we could from the CDR being written. If batch
+ mode is enabled, this will incorrectly report a billsec value
+ being much greater than the actual duration of the call. Instead
+ of relying on the presence of an answer time to know whether or
+ not we can re-calculate the billsec for the CDR, we now also use
+ the presence of the CDR's end time to know if we need to
+ re-calculate or whether we can trust the billsec value that we
+ have. This prevents erroneous jumps in the billsec value, while
+ still making sure that in the worst case, some billing time will
+ be calculated. (closes issue AST-1016) Reported by: Thomas
+ Arimont Tested by: Thomas Arimont ........ Merged revisions
+ 374843 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 374844 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-11 15:31 +0000 [r374842] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, include/asterisk/sip_api.h,
+ channels/chan_sip.exports.in (removed), main/sip_api.c (added):
+ Don't make chan_sip export global symbols. During testing, it was
+ discovered that having chan_sip export global symbols was
+ problematic. The biggest problem was that load order was
+ affected. Trying to use realtime could be problematic since in
+ all likelihood the necessary realtime driver(s) would not be
+ loaded before chan_sip. In addition, it was found that it was
+ impossible to use the Digium Phone Module for Asterisk since it
+ must be loaded before chan_sip since it must hook into chan_sip's
+ configuration parsing. The solution is to use a virtual table in
+ the same manner that other modules in Asterisk do, like
+ app_voicemail. (closes issue ASTERISK-20545) Reported by: kmoore
+
+2012-10-11 13:33 +0000 [r374833] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_motif.c: Consider the Google Talk content stanza
+ name (jin:content) valid.
+
+2012-10-10 21:03 +0000 [r374804] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_queue.c: app_queue: Made pass connected line updates
+ from the caller to ringing queue members. Party A calls Party B
+ Party B puts Party A on hold. Party B calls a queue. Ringing
+ queue member D sees Party B identification. Party B transfers
+ Party A to the queue. Queue member D does not get a connected
+ line update for Party A. Queue member D answers the call and
+ still sees Party B information. However, if Party A later
+ transfers the call to Party C then queue member D gets a
+ connected line update for Party C. * Made pass connected line
+ updates from the caller to queue members while the queue members
+ are ringing. (closes issue AST-1017) Reported by: Thomas Arimont
+ (closes issue ABE-2886) Reported by: Thomas Arimont Tested by:
+ rmudgett ........ Merged revisions 374801 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 374802 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374803 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-10 13:35 +0000 [r374792] Kinsey Moore <kmoore@digium.com>
+
+ * main/manager.c: Fix segfault regression from r370681 Due to usage
+ of ast_hook_send_action, AMI action handling code should be able
+ to handle a NULL mansession->session. This would cause a crash on
+ NULL dereference if action_originate was called from
+ ast_hook_send_action. (closes issue ASTERISK-20544)
+
+2012-10-09 22:21 +0000 [r374771] Richard Mudgett <rmudgett@digium.com>
+
+ * main/pbx.c, /: Fix execution of 'i' extension due to
+ uninitialized variable. The fix for ASTERISK-18243 added code
+ that could potentially use dst_exten[] uninitialized. As a result
+ the 'i' exten may not be executed when it should. (closes issue
+ ASTERISK-20455) Reported by: Richard Miller Patches:
+ pbx-1.8.16.0.diff (license #5685) patch uploaded by Richard
+ Miller Made some cosmetic modifications. ........ Merged
+ revisions 374758 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374763 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-09 21:34 +0000 [r374755-374756] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Improve logging for DTLS-SRTP failure
+ situations. (closes issue ASTERISK-20487) Reported by: mjordan
+
+ * channels/chan_sip.c: Add a log message for when DTLS-SRTP is
+ requested and the underlying engine does not support it. (closes
+ issue ASTERISK-20487) Reported by: mjordan
+
+2012-10-08 22:30 +0000 [r374708-374729] Richard Mudgett <rmudgett@digium.com>
+
+ * configs/chan_dahdi.conf.sample, /: dahdi.conf.sample: Add
+ description for "buffers" setting. This contains an edited
+ version of the patch originally created by John Bigelow. (closes
+ issue ASTERISK-14435) Reported by: John Bigelow Patches:
+ buffers.patch (license #5091) patch uploaded by John Bigelow
+ 0001-dahdi.conf.sample-Add-description-for-buffers-settin.patch
+ (license #5417) patch uploaded by Shaun Ruffell Modified ........
+ Merged revisions 374727 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374728 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * pbx/pbx_spool.c, /: Fix deletion of unopenable spool files. If
+ scan_service() cannot open the spool file, it logs a message
+ saying that it will delete the file and calls remove_from_queue()
+ to do it. However, remove_from_queue() fails to delete the spool
+ file because struct outgoing has not yet been fully initialized.
+ * Merged allocating a new struct outgoing and init_outgoing()
+ into new_outgoing(). Allocation is initialization. * Made
+ apply_outgoing() not initialize the spool filename in struct
+ outgoing. * Made apply_outgoing() call ast_trim_blanks() and
+ ast_skip_blanks() rather than manually inlining them. * Reduced
+ indentation levels in apply_outgoing(). * Fixed a garbled comment
+ in remove_from_queue(). * Reworked scan_service() to simplify it.
+ (closes issue ASTERISK-17231) Reported by: David Chappell
+ Patches: spool_open_failure.diff (license #4997) patch uploaded
+ by David Chappell Started with this patch. ........ Merged
+ revisions 374686 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 * Fixed some
+ memory leaks on off nominal paths in init_outgoing() when merging
+ into the new_outgoing() function dealing with o->capabilities.
+ ........ Merged revisions 374695 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-25 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 11.0.0 Released.
+
+2012-10-17 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 11.0.0-rc2 Released.
+
+ * [r374792] Fix segfault regression from r370681
+
+ Due to usage of ast_hook_send_action, AMI action handling code should
+ be able to handle a NULL mansession->session. This would cause a
+ crash on NULL dereference if action_originate was called from
+ ast_hook_send_action.
+
+ (closes issue ASTERISK-20544)
+
+ * [r374842] Don't make chan_sip export global symbols.
+
+ During testing, it was discovered that having chan_sip export global
+ symbols was problematic.
+
+ The biggest problem was that load order was affected.
+ Trying to use realtime could be problematic since in
+ all likelihood the necessary realtime driver(s) would
+ not be loaded before chan_sip.
+
+ In addition, it was found that it was impossible to
+ use the Digium Phone Module for Asterisk since it
+ must be loaded before chan_sip since it must hook
+ into chan_sip's configuration parsing.
+
+ The solution is to use a virtual table in the same
+ manner that other modules in Asterisk do, like
+ app_voicemail.
+
+ (closes issue ASTERISK-20545)
+ Reported by: kmoore
+
+ * [r374850] Fix an issue where outgoing calls would fail to establish
+ audio due to ICE negotiation failures.
+
+ This change removes the requirement for ufrag and pwd in the transport
+ stanza and also makes us the controlling agent.
+
+ (closes issue ASTERISK-20554)
+ Reported by: mmichelson
+
+ * [r374851] Remove code that should not have gotten in (r374850)
+
+ (issue ASTERISK-20554)
+
+ * [r374877] Fix a bug where audio on Google Voice would not work due to
+ ignoring candidates.
+
+ Instead of ignoring parts of the message that are not known just
+ ignore the ones we know may be present and that would cause a problem.
+
+ * [r375148] Ensure Asterisk fails TCP/TLS SIP calls when certificate
+ checking fails
+
+ When placing a call to a TCP/TLS SIP endpoint whose certificate is not
+ signed by a configured CA certificate, Asterisk would issue a warning
+ and continue to process the call as if there was not an issue with the
+ certificate. Asterisk now properly fails the call if the certificate
+ fails verification or if the certificate does not exist when
+ certificate checking is enabled (the default behavior).
+
+ (closes issue ASTERISK-20559)
+ Review: https://reviewboard.asterisk.org/r/2163/
+
+ * [r375051] Remove a log message that was left in accidentally from
+ call-id logging development.
+
+2012-10-08 Asterisk Development Team <asteriskteam@digium.com>
+
+ * Asterisk 11.0.0-rc1 Released.
+
+2012-10-08 20:38 +0000 [r374632-374676] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_rtp_asterisk.c, configs/rtp.conf.sample: Disable ICE
+ support by default Since there are a number of legacy devices out
+ there that fail to handle ICE candidates properly (which is a
+ nice way of saying something much uglier), disable it by default.
+ Support for ICE candidates can be enabled in rtp.conf using the
+ icesupport setting.
+
+ * apps/confbridge/conf_state.c (added),
+ apps/confbridge/conf_state_single.c (added),
+ apps/confbridge/conf_state_inactive.c (added),
+ apps/confbridge/conf_state_single_marked.c (added), /,
+ apps/confbridge/include/confbridge.h,
+ apps/confbridge/include/conf_state.h (added),
+ apps/confbridge/conf_state_multi.c (added),
+ apps/app_confbridge.c, apps/confbridge/conf_state_multi_marked.c
+ (added), apps/confbridge/conf_state_empty.c (added): Resolve
+ issues in ConfBridge regarding marked, waitmarked, and unmarked
+ users Thank's to Neil Tallim (flan)'s tireless testing, issue
+ reporting, and patches it became clear that app_confbridge had
+ some complex logic in how it handled interactions between marked,
+ waitmarked, and unmarked users. In particular, there were some
+ areas in which the interactions between the users resulted in
+ inconsistent behavior, and app_confbridge was missing logic in
+ how to handle some corner cases. Some areas included: * Poor
+ handling of mixing unmarked and waitmarked users *
+ Inconsistencies in how MOH and muting was applied to various
+ users * Handling of various announcements for different user
+ profile options flan's patches seem to fix the various issues,
+ but highlighted how hard the code could be to maintain. In an
+ attempt to make things easier to maintain and to more fully
+ enumerate the various cases that exist, this patch breaks up the
+ logic into a state machine-like setup. Please note that the
+ various state transitioned are documented on the Asterisk wiki:
+ https://wiki.asterisk.org/wiki/display/AST/Confbridge+state+changes
+ Review: //https://reviewboard.asterisk.org/r/2072/ Note that for
+ the following issues, mjordan uploaded the patch, although it was
+ written by twilson. Any contributor license discrepency is due to
+ that. (closes issue ASTERISK-19562) Reported by: flan Tested by:
+ flan, mjordan, jrose patches:
+ bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by
+ twilson (license 6283) (closes issue ASTERISK-19726) Reported by:
+ flan Tested by: flan patches:
+ bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by
+ twilson (license 6283) (closes issue ASTERISK-20181) Reported by:
+ Jonathan White Tested by: Jonathan White patches:
+ bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by
+ twilson (license 6283) ........ Merged revisions 374652 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * res/pjproject/pjlib/include/pj/sock.h,
+ res/pjproject/pjlib/src/pj/sock_symbian.cpp,
+ res/pjproject/pjlib/src/pj/sock_bsd.c,
+ res/pjproject/pjlib/src/pj/sock_linux_kernel.c: pjproject: Fix
+ for Solaris builds. Do not undef s_addr. pjproject, in order to
+ solve build problems on Windows [1], undefines s_addr in one of
+ it's headers that is included in res_rtp_asterisk.c. On Solaris
+ s_addr is not a structure member, but defined to map to the real
+ strucuture member, therefore when building on Solaris it's
+ possible to get build errors like: [CC] res_rtp_asterisk.c ->
+ res_rtp_asterisk.o In file included from
+ /export/home/admin/asterisk-11-svn/include/asterisk/stun.h:29,
+ from res_rtp_asterisk.c:51:
+ /export/home/admin/asterisk-11-svn/include/asterisk/network.h: In
+ function `inaddrcmp':
+ /export/home/admin/asterisk-11-svn/include/asterisk/network.h:92:
+ error: structure has no member named `s_addr'
+ /export/home/admin/asterisk-11-svn/include/asterisk/network.h:92:
+ error: structure has no member named `s_addr' res_rtp_asterisk.c:
+ In function `ast_rtp_on_ice_tx_pkt': res_rtp_asterisk.c:706:
+ warning: dereferencing type-punned pointer will break
+ strict-aliasing rules res_rtp_asterisk.c:710: warning:
+ dereferencing type-punned pointer will break strict-aliasing
+ rules res_rtp_asterisk.c: In function
+ `rtp_add_candidates_to_ice': res_rtp_asterisk.c:1085: error:
+ structure has no member named `s_addr' make[2]: ***
+ [res_rtp_asterisk.o] Error 1 make[1]: *** [res] Error 2 make[1]:
+ Leaving directory `/export/home/admin/asterisk-11-svn' gmake: ***
+ [_cleantest_all] Error 2 Unfortunately, in order to make this
+ work, I also had to make sure pjproject only used the typdef
+ pj_in_addr and not the struct pj_in_addr so that when building
+ Asterisk I could "typedef struct in_addr pj_in_addr". It's
+ possible then that the library and users of those interfaces in
+ Asterisk have a different idea about the type of the argument,
+ while on the surface it looks like they are all 32 bit big endian
+ values. [1] http://trac.pjsip.org/repos/changeset/484 (issues
+ ASTERISK-20366) Reported by: Ben Klang Tested by: Ben Klang,
+ mjordan patches:
+ 0001-pjproject-Fix-for-Solaris-builds.-Do-not-undef-s.patch
+ uploaded by Shaun Ruffell (license 5417)
+
+ * main/acl.c: Trivial patch to make 'best_score' defined for all
+ architectures. Fixes trivial build error on Solaris: acl.c: In
+ function `get_local_address': acl.c:196: error: `best_score'
+ undeclared (first use in this function) acl.c:196: error: (Each
+ undeclared identifier is reported only once acl.c:196: error: for
+ each function it appears in.) make[2]: *** [acl.o] Error 1 (issue
+ ASTERISK-20366) Reported by: Ben Klang Tested by: Ben Klang
+ patches:
+ 0002-main-acl.c-Trivial.-best_score-should-be-defined-for.patch
+ by Shaun Ruffell (license 5417)
+
+2012-10-06 03:20 +0000 [r374611-374622] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_xmpp.c: Handle capability stanzas that fail to provide
+ node or version information While XEP-0115 states that the node
+ and ver attributes are both required, some devices fail to
+ provide either field. Prior to this patch, failure to provide the
+ node or ver attribute would cause a crash in res_xmpp. While
+ failing to provide the node or ver attribute is technically
+ invalid, since this information is not utilized by Asterisk
+ except for reporting purposes, for interoperability reasons, we
+ continue to process the capability stanza anyways. (closes issue
+ ASTERISK-20495) Reported by: Martin W Tested by: Martin W
+ patches: 20495.patch uploaded by Martin W (license #6434)
+
+ * res/res_xmpp.c, main/message.c: Update documentation for
+ MessageSend application/command's From field for XMPP When using
+ the channel technology agnostic application/AMI command
+ MessageSend, the "From" field is technically optional for the SIP
+ channel driver. However, if being sent by the XMPP resource
+ module (either res_xmpp or res_jabber), the "From" field is
+ necessary, and must correspond to a defined account. This patch
+ updates the documentation for this application/AMI command to
+ reflect this. (closes issue ASTERISK-20405) Reported by: Leif
+ Madsen
+
+2012-10-05 20:32 +0000 [r374587] dlee <dlee@localhost>:
+
+ * main/manager.c, /: Multiple revisions 374570,374581 ........
+ r374570 | dlee | 2012-10-05 15:14:41 -0500 (Fri, 05 Oct 2012) |
+ 22 lines 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/ ........ r374581
+ | dlee | 2012-10-05 15:20:28 -0500 (Fri, 05 Oct 2012) | 1 line
+ I've committed too much. Reverting part of r374570. ........
+ Merged revisions 374570,374581 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374586 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-05 18:34 +0000 [r374538] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
+ channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.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. ................
+ ........ Merged revisions 374536 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374537 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-04 20:18 +0000 [r374477-374485] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/dsp.c, /, configs/dsp.conf.sample, CHANGES: 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/ ........ Merged
+ revisions 374479 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374481 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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/ ........ Merged
+ revisions 374475 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374476 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-04 15:42 +0000 [r374428] dlee <dlee@localhost>:
+
+ * main/db.c, /, res/res_agi.c: Fix DBDelTree error codes for AMI,
+ CLI and AGI The AMI DBDelTree command will return Success/Key
+ tree deleted successfully even if the given key does not exist.
+ The CLI command 'database deltree' had a similar problem, but was
+ saved because it actually responded with '0 database entries
+ removed'. AGI had a slightly different error, where it would
+ return success if the database was unavailable. This came from
+ confusion about the ast_db_deltree retval, which is -1 in the
+ event of a database error, or number of entries deleted
+ (including 0 for deleting nothing). * 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/ ........ Merged
+ revisions 374426 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374427 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-04 04:43 +0000 [r374379-374386] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/dsp.c, /, configs/dsp.conf.sample, CHANGES: 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/ ........ Merged
+ revisions 374384 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374385 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /: _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 ........ Merged
+ revisions 374365 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374370 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-04 02:15 +0000 [r374196-374337] Matthew Jordan <mjordan@digium.com>
+
+ * /, res/res_jabber.c: Check for presence of buddy in info/dinfo
+ handlers The res_jabber resource module uses the ASTOBJ library
+ for managing its ref counted objects. After calling
+ ASTOBJ_CONTAINER_FIND to locate a buddy object, the pointer to
+ the object has to be checked to see if the buddy existed. Prior
+ to this patch, the buddy object was not checked for NULL; with
+ this patch in both aji_client_info_handler and aji_dinfo_handler
+ the pointer is checked before used and, if no buddy object was
+ found, the handlers return an error code. This patch does not
+ take the approach that our JID can be used to log in from another
+ resource. If that approach is desired, an improvement could be
+ made to this patch to create the buddy on the fly. This patch
+ seeks only to prevent Asterisk from crashing. FYI: In Asterisk
+ 11+, you really should be using res_xmpp. It does not have this
+ problem, as it moved to the astobj2 library. 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 ........ Merged revisions 374335 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374336 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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. ........
+ Merged revisions 374300 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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. ........
+ Merged revisions 374230 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374231 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, main/message.c: Fix findings from check-in on r374177 Richard
+ pointed out two problems with the check-in from r374177: * The
+ ast_msg_shutdown function declaration doesn't match the prototype
+ in main/message.c. * The ref/alloc function usage in astobj2 (in
+ trunk) can use the ao2_t_* variants of the functions to allow the
+ REF_DEBUG flag to enable/disable their debug counterparts.
+ ........ Merged revisions 374210 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/db.c, main/asterisk.c, main/xmldoc.c, main/format.c,
+ main/udptl.c, main/pbx.c, /, main/ccss.c,
+ include/asterisk/astobj2.h, channels/chan_agent.c,
+ main/taskprocessor.c, res/res_musiconhold.c, res/res_xmpp.c,
+ main/cel.c, main/named_acl.c, main/indications.c,
+ main/format_pref.c, main/astobj2.c, main/channel.c, main/data.c,
+ main/manager.c, main/features.c, main/config_options.c,
+ main/event.c, main/message.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 ........ Merged revisions
+ 374177 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 374178 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-01 20:26 +0000 [r374133-374150] Sean Bright <sean@malleable.com>
+
+ * main/db.c, include/asterisk/astdb.h, /, tests/test_db.c,
+ apps/app_queue.c: 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/ ........ Merged
+ revisions 374108 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 374135 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/db.c, /: Use ast_copy_string instead of strncpy to guarantee
+ a NUL terminated string. ........ Merged revisions 374132 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-01 16:12 +0000 [r374106] Mark Michelson <mmichelson@digium.com>
+
+ * apps/confbridge/conf_config_parser.c: Don't destroy confbridge
+ config when error is encountered during a reload. Not panicking
+ means that the old config is kept. (closes issue ASTERISK-20458)
+ Reported by: Leif Madsen Patches: ASTERISK-20458.patch uploaded
+ by Mark Michelson(license #5049) Tested by Leif Madsen
+
+2012-09-29 03:54 +0000 [r374085] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_sip.c: Fix ref leak when adding ICE candidates to
+ an SDP There was a missing decrement to the reference count for
+ the current ICE candidate when local candidates are being added
+ to an outbound SDP. This patch corrects that.
+
+2012-09-28 19:29 +0000 [r374059] Jonathan Rose <jrose@digium.com>
+
+ * /, res/res_jabber.c: res_jabber: Remove CLI command 'jabber test'
+ The opinion of development was that it is both improper to have
+ Matt's personal email address used in the source and that the
+ command wouldn't be useful without it. (closes issue AST-467)
+ Reported by: Malcolm Davenport ........ Merged revisions 374032
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 374045 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-28 13:02 +0000 [r374019] beagles <beagles@localhost>:
+
+ * res/res_xmpp.c, main/message.c: Reset hangup flags on channels
+ created through messages and cleanup globals in res_xmpp on
+ unload. This patch fixes an issue where hangup flags were not
+ being reset on a channel, affecting subsequent use of that
+ channel. The patch also adds some additional cleanup to res_xmpp
+ to fix an issue with reloading the module. (closes
+ ASTERISK-20360) Reported by: Noah Engelberth Tested by: beagles
+ Review: https://reviewboard.asterisk.org/r/2134/
+
+2012-09-28 12:16 +0000 [r373991] Joshua Colp <jcolp@digium.com>
+
+ * /, res/res_agi.c: Update documentation to make it explicit that
+ "stream file" will not restart musiconhold. (issue
+ ASTERISK-17367) Reported by: oej ........ Merged revisions 373989
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 373990 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-27 22:19 +0000 [r373954] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_senddtmf.c: Fix SendDTMF crash and channel reference
+ leak using channel name parameter. The SendDTMF channel name
+ parameter has two issues. 1) Crashes if the channel name does not
+ exist. 2) Leaks a channel reference if the channel is the current
+ channel. Problem introduced by ASTERISK-15956. * Updated SendDTMF
+ documentation. * Renamed app to senddtmf_name and tweaked the
+ type. ........ Merged revisions 373945 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373946 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-27 17:05 +0000 [r373880-373914] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c, include/asterisk/http_websocket.h,
+ res/res_http_websocket.c: Make res_http_websocket an optional
+ dependency on supported platforms for chan_sip. (closes issue
+ ASTERISK-20439) Reported by: sruffell Patches:
+ 0001-chan_sip-websocket-support-is-an-optional-API.patch uploaded
+ by sruffell (license 5417)
+
+ * 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) ........ Merged revisions
+ 373909 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 373910 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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/ ........ Merged
+ revisions 373878 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373879 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-26 21:16 +0000 [r373850] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Move handling of 408 response so there is
+ no misleading warning message. (closes issue ASTERISK-20060)
+ Reported by: Walter Doekes ........ Merged revisions 373848 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373849 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-26 18:18 +0000 [r373818] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_meetme.c: Fixed meetme tab completion and command
+ documentation. * Removed unnecessary case sensitivity in meetme
+ list, lock, unlock, mute, unmute, and kick commands. * Separated
+ meetme lock/unlock, mute/unmute, and kick commands into their own
+ registered commands to simplify tab completion and parameter
+ checking. meetme_lock_cmd(), meetme_mute_cmd(), and
+ meetme_kick_cmd() * Simplified meetme_show_cmd() (closes issue
+ AST-1006) Reported by: John Bigelow Tested by: rmudgett ........
+ Merged revisions 373815 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373816 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-26 08:29 +0000 [r373804] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * apps/app_queue.c: app_queue: 'agent available' hint, cleanup
+ restart, and initial state Fix previously untested senarios; 1).
+ On queue initialisation set queue_avail devstate to INUSE.
+ Previously was unavailable, which indicated an agent was
+ available. 2). When removing members, if there are no other
+ members available, set queue_avail to INUSE. Previously, if a
+ member interface had become 'unavailable', they were never going
+ to be removed, particularly when persistant queues is enabled.
+ 3). When adding a member, check that they are available, if they
+ are set queue_avail to NOT_INUSE. Previously on reloaded, members
+ may have been 'unavailable'. 4). When pausing or unpausing a
+ member, set appropriate queue availability. alecdavis (license
+ 585) Reported by: Alec Davis Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/2129/
+
+2012-09-25 23:09 +0000 [r373738-373775] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/say.c: Fix saying of date in Dutch. The Dutch say the
+ date before the month. (closes issue ASTERISK-20353) Reported by:
+ Teun Ouwehand ........ Merged revisions 373773 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373774 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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
+ ........ Merged revisions 373768 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373769 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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) ........ Merged revisions 373735 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373737 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-25 20:13 +0000 [r373707] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_local.c, /: Fix T.38 support when used with
+ chan_local in between. Users of the T.38 API can indicate
+ AST_T38_REQUEST_PARMS on a channel to request that the channel
+ indicate a T.38 negotiation with the parameters present on the
+ channel. The return value of this indication is expected to be
+ AST_T38_REQUEST_PARMS upon success but with chan_local involved
+ this could never occur. This fix changes chan_local to always
+ return AST_T38_REQUEST_PARMS for this situation. If the
+ underlying channel technology on the other side does not support
+ T.38 this would have been determined ahead of time using
+ ast_channel_get_t38_state and an indication would not occur.
+ (closes issue ASTERISK-20229) Reported by: wdoekes Patches:
+ ASTERISK-20229.patch uploaded by wdoekes (license 5674) Review:
+ https://reviewboard.asterisk.org/r/2070/ ........ Merged
+ revisions 373705 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373706 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-25 19:35 +0000 [r373704] Kinsey Moore <kmoore@digium.com>
+
+ * /: Recorded merge of revisions 373703 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ 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) ........ Merged
+ revisions 373702 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-25 18:52 +0000 [r373690] Terry Wilson <twilson@digium.com>
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c,
+ configs/sip.conf.sample: 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/ ........ Merged
+ revisions 373652 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373665 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-25 18:24 +0000 [r373688] Kinsey Moore <kmoore@digium.com>
+
+ * /, 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 ........ Merged revisions 373666 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373675 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-25 17:21 +0000 [r373635-373650] Richard Mudgett <rmudgett@digium.com>
+
+ * /, codecs/ilbc/iLBC_encode.c, codecs/ilbc/iLBC_decode.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 ........ Merged revisions
+ 373640 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 373645 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * codecs/Makefile, /: Make rebuild GSM, ilbc, or lpc10 codecs if
+ the respective sources change. ........ Merged revisions 373618
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 373633 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-25 16:31 +0000 [r373632] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Set Quality of Service for
+ video rtp instance (closes issue ASTERISK-20201) Reported by:
+ ddkprog Patches: chan_sip.c.diff uploaded by ddkprog (license
+ 6008) ........ Merged revisions 373617 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373631 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-25 14:12 +0000 [r373582] Mark Michelson <mmichelson@digium.com>
+
+ * funcs/func_presencestate.c: "He who go through turnstile sideways
+ is going to Bangkok"
+
+2012-09-25 13:29 +0000 [r373580] Kinsey Moore <kmoore@digium.com>
+
+ * configs/res_odbc.conf.sample, /: Fix documentation for default
+ username in res_odbc This was previously stated to be "root", but
+ is actually the name of the context if unspecified. (closes issue
+ ASTERISK-20258) Reported by: Stefan x ........ Merged revisions
+ 373578 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 373579 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-25 12:07 +0000 [r373552] Joshua Colp <jcolp@digium.com>
+
+ * res/res_rtp_multicast.c, /: Fix an issue where a caller to
+ ast_write on a MulticastRTP channel would determine it failed
+ when in reality it did not. When sending RTP packets via
+ multicast the amount of data sent is stored in a variable and
+ returned from the write function. This is incorrect as any
+ non-zero value returned is considered a failure while a return
+ value of 0 is success. For callers (such as ast_streamfile) that
+ checked the return value they would have considered it a failure
+ when in reality nothing went wrong and it was actually a success.
+ The write function for the multicast RTP engine now returns -1 on
+ failure and 0 on success, as it should. (closes issue
+ ASTERISK-17254) Reported by: wybecom ........ Merged revisions
+ 373550 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 373551 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-24 22:17 +0000 [r373508] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_rtp_asterisk.c, /: Revert change to res_rtp_asterisk
+ committed in r373236 (1.8) The change committed in r373236
+ attempted to account for endpoints that increased their RTP
+ timestamp in DTMF end of event re-transmissions. This change
+ attempted to make Asterisk continue to work with endpoints that
+ failed to follow the RFC while maintaining the fix that allowed
+ for out of order DTMF to be handled. Unfortunately, there is no
+ free lunch, and this patch broke any system that sent DTMF
+ immediately after an RTP session was established or when an SSRC
+ is updated. As such, that patch is being reverted for the
+ previous behavior. Endpoints that erroneously increase the RTP
+ timestamp in DTMF end of event packets will not work properly
+ with Asterisk. (issue ASTERISK-20424) ........ Merged revisions
+ 373504 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 373505 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-24 22:12 +0000 [r373502] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Be consistent, send From: "Anonymous"
+ <sip:anonymous@anonymous.invalid> When setting
+ CALLERID(pres)=unavailable in the dialplan, the From header in
+ the SIP message contains "Anonymous"
+ <sip:Anonymous@anonymous.invalid>. For consistency, Asterisk
+ should use a lowercase a in the userpart of the URI. * Make the
+ From header use a lowercase A in the userpart of the anonymous
+ URI. (closes issue ASTERISK-19838) Reported by: Antti Yrjola
+ Patches: chan_sip_patch_ASTERISK-19838.patch (license #6383)
+ patch uploaded by Antti Yrjola ........ Merged revisions 373500
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 373501 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-24 21:12 +0000 [r373470] Jonathan Rose <jrose@digium.com>
+
+ * funcs/func_audiohookinherit.c, /, apps/app_mixmonitor.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 ........
+ Merged revisions 373467 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373468 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-24 21:08 +0000 [r373469] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Fix potential reentrancy problems in
+ chan_sip. Asterisk v1.8 and later was not as vulnerable to this
+ issue. * Made find_call() lock each private as it processes the
+ found dialogs. (Primary cause of ABE-2876) * Made the other
+ functions that traverse the dialogs container lock each private
+ as it examines them. * Fix race condition in sip_call() if the
+ thread that sent the INVITE is held up long enough for a response
+ to be processed. The p->initid for the INVITE retransmission
+ could be added after it was canceled by the response processing.
+ * Made __sip_destroy() clean up resource pointers after freeing.
+ This is primarily defensive in case someone has a stale private
+ pointer. * Removed redundant memset() in reqprep(). The call to
+ init_req() already does the memset() and is the first reference
+ to req in reqprep(). * Removed useless set of req.method in
+ transmit_invite(). The calls to initreqprep() and reqprep() have
+ to do this because they memset() the req. JIRA ABE-2876
+ .......... Merged -r373423 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 373424 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373466 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-24 19:21 +0000 [r373413-373454] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Fix a deadlock caused by a race condition
+ between removing a hint and reloading the dialplan and
+ subscribing to the removed hint. If conditions were right it was
+ possible for both the PBX core and chan_sip to deadlock by both
+ having a lock that the other wants. In the case of the PBX core
+ it had the contexts lock and wanted a SIP dialog lock, while in
+ the case of chan_sip it had the SIP dialog lock and wanted the
+ contexts lock. This fix unlocks the SIP dialog before getting the
+ extension state so that the other thread will not block on trying
+ to lock it. Once the extension state is retrieved the SIP dialog
+ is locked again and life carries on. As the SIP dialog is
+ reference counted it is not possible for it to go away after
+ unlocking. (closes issue ASTERISK-20437) Reported by: jhutchins
+ ........ Merged revisions 373438 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373440 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/chan_sip.c, res/res_format_attr_h264.c: Fix an issue
+ with H.264 format attribute comparison and fix an issue with
+ improper SDP being produced. The H.264 format attribute module
+ compares two format attribute structures to determine if they are
+ compatible or not. In some instances it was possible for this
+ check to determine that both structures were incompatible when
+ they actually should be considered compatible. This check has now
+ been made even more permissive by assuming that if no attribute
+ information is available the two structures are compatible. If
+ both structures contain attribute information a base level
+ comparison of the H.264 IDC value is done to see if they are
+ compatible or not. The above issue uncovered a secondary issue in
+ chan_sip where the SDP being produced would be incorrect if the
+ formats were considered incompatible. This has now been fixed by
+ checking that all information required to produce the SDP is
+ available instead of assuming it is. (closes issue
+ ASTERISK-20464) Reported by: Leif Madsen
+
+2012-09-24 12:33 +0000 [r373403] beagles <beagles@localhost>:
+
+ * res/res_rtp_asterisk.c, configs/rtp.conf.sample:
+ res_rtp_asterisk: Make TURN and STUN server configurations
+ consistent. This patch removes the turnport configuration
+ property and changes the turnaddr property to be a combined
+ host[:port] configuration string. The patch also modifies the
+ documentation in the example configuration to reflect the
+ property changes and adds some additional text indicating how the
+ STUN port is configured. (closes issue ASTERISK-20344) Reported
+ by: beagles Tested by: beagles Review:
+ https://reviewboard.asterisk.org/r/2111/
+
+2012-09-21 19:29 +0000 [r373318-373368] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/iax2-provision.c: iax2-provision: Fix improper return
+ on failed cache retrieval (closes issue ASTERISK-20337) reported
+ by: John Covert Patches: iax2-provision.c.patch uploaded by John
+ Covert (license 5512) ........ Merged revisions 373342 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373343 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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 ........
+ Merged revisions 373298 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373300 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-20 19:16 +0000 [r373246] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_meetme.c: Fix incorrect MeetME conference bridge
+ reference count decrementing and sometimes premature destruction.
+ When using the 'e' or 'E' option to MeetMe the configured
+ conference bridges are loaded and examined to see if any are
+ empty. If no conference bridges are empty the caller is prompted
+ to enter the number of one. This operation left around a pointer
+ to the last created conference bridge still containing
+ participants. When the caller that was not able to find any empty
+ conference bridge hung up this pointer was disposed of and the
+ reference count of the conference bridge decremented. If there
+ was only a single participant in the conference bridge it was
+ ultimately destroyed prematurely. (closes issue AST-994) Reported
+ by: John Bigelow ........ Merged revisions 373242 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373245 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-20 18:59 +0000 [r373235-373240] Matthew Jordan <mjordan@digium.com>
+
+ * configs/extensions.conf.sample, CHANGES, apps/app_queue.c:
+ app_queue: Support an 'agent available' hint Sets INUSE when no
+ free agents, NOT_INUSE when an agent is free. modifes
+ handle_statechange() scan members loop to scan for a free agent
+ and updates the Queue:queuename_avial devstate. Previously exited
+ early if the member was found in the queue. Now Exits later when
+ both a member was found, and a free agent was found. alecdavis
+ (license 585) Reported by: Alec Davis Tested by: alecdavis
+ Review: https://reviewboard.asterisk.org/r/2121/ ~~~~ Support all
+ ways a member can be available for 'agent available' hints Alec's
+ patch in r373188 added the ability to subscribe to a hint for
+ when Queue members are available. This patch modifies the check
+ that determines when a Queue member is available by refactoring
+ the availability checks in num_available_members into a shared
+ function is_member_available. This should now handle the
+ ringinuse option, as well as device state values other than
+ AST_DEVICE_NOT_INUSE.
+
+ * 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 ........ Merged revisions
+ 373236 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 373237 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * configs/extensions.conf.sample, CHANGES, apps/app_queue.c: Add
+ queue monitoring hints This patch adds support for hints on a
+ queue. Hints can be added using the nomenclature 'Queue:name',
+ where name is the name of the queue being monitored. This nifty
+ feature was done by Alec Davis. Review:
+ https://reviewboard.asterisk.org/r/1619 Reported by: Alec Davis
+ Tested by: alecdavis patches: review1619.diff2 by alecdavis
+ (license 585)
+
+2012-09-20 18:18 +0000 [r373229] Joshua Colp <jcolp@digium.com>
+
+ * channels/sip/include/sip.h, res/res_rtp_asterisk.c,
+ main/rtp_engine.c, channels/chan_sip.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ configs/sip.conf.sample, include/asterisk/rtp_engine.h: Add
+ support for DTLS-SRTP to res_rtp_asterisk and chan_sip. As
+ mentioned on the review for this, WebRTC has moved towards
+ choosing DTLS-SRTP as the mechanism for key exchange for SRTP.
+ This commit adds support for this but makes it available for
+ normal SIP clients as well. Testing has been done to ensure that
+ this introduces no regressions with existing behavior and also
+ that it functions as expected. Review:
+ https://reviewboard.asterisk.org/r/2113/
+
+2012-09-20 17:15 +0000 [r373220] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/features.h, main/channel.c,
+ apps/app_directed_pickup.c, funcs/func_channel.c,
+ main/features.c, include/asterisk/channel.h: Named call pickup
+ groups. Fixes, missing functionality, and improvements. *
+ ASTERISK-20383 Missing named call pickup group features:
+ CHANNEL(callgroup) - Need CHANNEL(namedcallgroup)
+ CHANNEL(pickupgroup) - Need CHANNEL(namedpickupgroup) Pickup() -
+ Needs to also select from named pickup groups. * ASTERISK-20384
+ Using the pickupexten, the pickup channel selection could fail
+ even though there was a call it could have picked up. In a call
+ pickup race when there are multiple calls to pickup and two
+ extensions try to pickup a call, it is conceivable that the loser
+ will not pick up any call even though it could have picked up the
+ next oldest matching call. Regression because of the named call
+ pickup group feature. * See ASTERISK-20386 for the implementation
+ improvements. These are the changes in channel.c and channel.h. *
+ Fixed some locking issues in CHANNEL(). (closes issue
+ ASTERISK-20383) Reported by: rmudgett (closes issue
+ ASTERISK-20384) Reported by: rmudgett (closes issue
+ ASTERISK-20386) Reported by: rmudgett Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/2112/
+
+2012-09-20 13:00 +0000 [r373211] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c: Correct handling of unknown SDP stream types
+ When the patch to handle arbitrary SDP stream arrangements went
+ into Asterisk, it also included an ability to transparently
+ decline unknown stream types. The scanf calls used were not
+ checked properly causing this part of the functionality to be
+ broken. (closes issue ASTERISK-20203)
+
+2012-09-18 20:14 +0000 [r373133] Sean Bright <sean@malleable.com>
+
+ * main/manager.c, /: Don't crash when passing a NULL message to
+ __astman_get_header. Before this commit, __astman_get_header
+ would blindly dereference the passed in 'struct message *' to
+ traverse the header list. There are cases, however, such as
+ '*CLI> sip qualify peer foo' where the message pointer is NULL,
+ so we need to check for that. ........ Merged revisions 373131
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 373132 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-18 15:47 +0000 [r373119] dlee <dlee@localhost>:
+
+ * Makefile, include/asterisk/utils.h, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
+ -fnested-functions compile flag, if needed. In order to use
+ nested functions on some versions of GCC (e.g. GCC on OS X), the
+ -fnested-functions flag must be passed to the compiler. This
+ patch adds detection logic to ./configure to add the flag if
+ necessary. It also adds a comment to utils.h as to why the nested
+ function needs a prototype. (closes issue ASTERISK-20399)
+ Reported by: David M. Lee Review:
+ https://reviewboard.asterisk.org/r/2102/
+
+2012-09-15 00:27 +0000 [r373107] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_ss7.c, /: Made companding law for SS7 calls only
+ determined by SS7 signaling type. For SS7, the companding law for
+ a call was chosen inconsistently depending upon ss7type (ITU vs
+ ANSI) and the DAHDI companding default (T1 vs E1). For incoming
+ calls, the companding law was determined by ss7type. For outgoing
+ calls, the companding law was determined by the DAHDI default.
+ With the wrong combination you would get A-law/u-law conflicts.
+ An A-law/u-law conflict sounds like bad static on the line. SS7
+ ITU signaling with E1 line: ok SS7 ITU signaling with T1 line:
+ noise SS7 ANSI signaling with E1 line: noise SS7 ANSI signaling
+ with T1 line: ok * Fix the companding law used to be determined
+ by the SS7 signaling type only. ........ Merged revisions 373090
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 373101 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-14 19:50 +0000 [r373079] Matthew Jordan <mjordan@digium.com>
+
+ * main/tcptls.c, /, channels/chan_sip.c, main/libasteriskssl.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 ........ Merged revisions
+ 373061 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 373062 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-13 20:04 +0000 [r373029-373047] dlee <dlee@localhost>:
+
+ * main/Makefile: Fixed make clean when configured
+ --disable-asteriskssl
+
+ * main/channel.c, /, include/asterisk/channel.h: 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/ ........ Merged
+ revisions 373024 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 373025 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-12 20:53 +0000 [r372995] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_motif.c: Skip any non-content information when
+ looking for and handling content. This fixes a bug with Jitsi and
+ conference calling. Jitsi implements XEP-0298 which places some
+ conference-info information in the session-initiate request which
+ chan_motif did not expect to occur.
+
+2012-09-12 18:23 +0000 [r372984] Jonathan Rose <jrose@digium.com>
+
+ * res/res_xmpp.c: res_xmpp: Fix a segfault caused by bodyless
+ messages (closes issue ASTERISK-20361) Reported by: Noah
+ Engelberth Review: https://reviewboard.asterisk.org/r/2108/
+
+2012-09-12 15:19 +0000 [r372937] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Add channel name to a warning to make
+ debugging easier. The "autodestruct with owner in place" message
+ is typically indicative of a channel reference leak. Printing out
+ the name of the channel in the message may be helpful when trying
+ to debug the issue. ........ Merged revisions 372932 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372933 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-12 14:18 +0000 [r372930] dlee <dlee@localhost>:
+
+ * main/Makefile: Fixed r372696 when configured
+ --disable-asteriskssl; properly install libasteriskssl.dylib on
+ OS X. I didn't realize that libasteriskssl.c was still compiled,
+ even when you disable asteriskssl; it simple gets statically
+ linked into asterisk.
+
+2012-09-11 22:32 +0000 [r372917] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_local.c, /: chan_local: Switch from using a random
+ 4 digit hex identifier to unique id Changes chan_local channels
+ to use an 8 digit hex identifier generated atomically and
+ sequentially in order to eliminate the chance of having multiple
+ channels with the same name during high call volume situations.
+ (issue ASTERISK-20318) Reported by: Dan Cropp Review:
+ https://reviewboard.asterisk.org/r/2104/ ........ Merged
+ revisions 372902 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372916 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-11 21:15 +0000 [r372886-372888] Mark Michelson <mmichelson@digium.com>
+
+ * main/asterisk.c, /, include/asterisk/_private.h, main/message.c:
+ Fix inability to shutdown gracefully due to an unending channel
+ reference. message.c makes use of a special message queue channel
+ that exists in thread storage. This channel never goes away due
+ to the fact that the taskprocessor used by message.c does not get
+ shut down, meaning that it never ends the thread that stores the
+ channel. This patch fixes the problem by shutting down the
+ taskprocessor when Asterisk is shut down. In addition, the thread
+ storage has a destructor that will release the channel reference
+ when the taskprocessor is destroyed. (closes issue AST-937)
+ Reported by Jason Parker Patches: AST-937.patch uploaded by Mark
+ Michelson (License #5049) Tested by Jason Parker ........ Merged
+ revisions 372885 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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 ........ Merged revisions 372840 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372841 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-11 17:16 +0000 [r372864] dlee <dlee@localhost>:
+
+ * Makefile, /: Corrects the astsbindir setting when installing the
+ sample asterisk.conf. (closes issue ASTERISK-20406) ........
+ Merged revisions 372863 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-10 20:59 +0000 [r372795-372806] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_iax2.c: Ensure iax2 debug output is displayed
+ when expected When IAX2 debug was changed from iax_showframe to
+ iax_outputframe, some instances were missed (or added afterward).
+ This was causing debug output to not be displayed when expected.
+ (closes issue ASTERISK-20338) Reported-by: John Covert Patch-by:
+ John Covert ........ Merged revisions 372804 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372805 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/chan_jingle.c, include/asterisk/doxygen/architecture.h,
+ main/devicestate.c, channels/chan_gtalk.c, res/res_jabber.c:
+ Deprecate chan_gtalk, chan_jingle, and res_jabber chan_gtalk,
+ chan_jingle, and res_jabber are now deprecated in favor of using
+ chan_motif and res_xmpp. They are a feature-equivalent
+ replacement and are written to be more easily maintainable.
+ (closes issue ASTERISK-20298) Review:
+ https://reviewboard.asterisk.org/r/2082/ Reported-by: Leif Madsen
+
+2012-09-10 19:19 +0000 [r372777] dlee <dlee@localhost>:
+
+ * res/res_rtp_asterisk.c: res_rtp_asterisk: Eliminate "type-punned
+ pointer" build warning. Removes "res_rtp_asterisk.c:706: warning:
+ dereferencing type-punned pointer will break strict-aliasing
+ rules" warning from the build on 32-bit platforms. The problem is
+ that 'size' was referenced aliased to both (pj_size_t *) and
+ (pj_ssize_t *). Now just make a copy of size that is the right
+ type so there isn't any pointer aliasing happening. It also adds
+ comments and asserts regarding what looks like an inappropriate
+ use of pj_sock_sendto, but is actually totally fine. (closes
+ issue ASTERISK-20368) Reported by: Shaun Ruffel Tested by:
+ Michael L. Young Patches:
+ 0001-res_rtp_asterisk-Eliminate-type-punned-pointer-build.patch
+ uploaded by Shaun Ruffel (license 5417) slightly modified by
+ David M. Lee.
+
+2012-09-10 18:50 +0000 [r372768] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_meetme.c: app_meetme: Document that 'p' option will
+ continue in dialplan. (closes issue AST-991) Reported by John
+ Bigelow ........ Merged revisions 372765 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372767 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-10 18:37 +0000 [r372766] Kinsey Moore <kmoore@digium.com>
+
+ * /: Recorded merge of revisions 372764 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ 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 ........ Merged revisions
+ 372763 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-10 17:33 +0000 [r372754] Jonathan Rose <jrose@digium.com>
+
+ * main/channel.c, /: Masquerade: Retain parkinglot settings made by
+ CHANNEL function. Prior to this patch, the user would have a
+ parkinglot set on a channel that was parked and when the channel
+ was retrieved, any attempt by that channel to park would simply
+ use the default. This patch makes parkinglot values set in this
+ way be retained through the masquerade. (closes issue AST-990)
+ Reported by: Nick Huskinson Patches:
+ masquerade_parkinglot_patch.diff Uploaded by Jonathan Rose
+ (license 6182) ........ Merged revisions 372736 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372737 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-09 01:25 +0000 [r372711] Matthew Jordan <mjordan@digium.com>
+
+ * channels/sip/sdp_crypto.c, /: Only re-create an SRTP session when
+ needed 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. (issue ASTERISK-20194) Reported
+ by: Nicolo Mazzon Tested by: Nicolo Mazzon Review:
+ https://reviewboard.asterisk.org/r/2099 ........ Merged revisions
+ 372709 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 372710 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-08 05:51 +0000 [r372696] dlee <dlee@localhost>:
+
+ * /, main/Makefile: Recorded merge of revisions 372695 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ 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) ........ Merged revisions 372682 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-07 23:07 +0000 [r372622-372657] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/astmm.c: Fix MALLOC_DEBUG version of ast_strndup().
+ (closes issue ASTERISK-20349) Reported by: Brent Eagles ........
+ Merged revisions 372655 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372656 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, funcs/func_math.c: Remove annoying unconditional debug message
+ from INC/DEC functions. (closes issue AST-1001) Reported by:
+ Guenther Kelleter ........ Merged revisions 372628 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372629 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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 ........ Merged revisions 372624
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 372625 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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 ........ Merged revisions 372620 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372621 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-07 21:04 +0000 [r372609-372611] dlee <dlee@localhost>:
+
+ * res/pjproject/pjlib-util/lib, res/pjproject/pjmedia/bin,
+ res/pjproject/third_party/bin, res/pjproject/third_party/gsm/lib,
+ res/pjproject/lib, res/pjproject/pjlib/lib,
+ res/pjproject/third_party/gsm/bin, res/pjproject/pjnath/lib,
+ res/pjproject/pjsip/lib, res/pjproject/pjsip-apps/lib,
+ res/pjproject/pjsip/bin, res/pjproject/pjsip-apps/bin,
+ res/pjproject/pjmedia/lib, res/pjproject/third_party/lib,
+ codecs/ilbc: svn:ignore cleanup. * pjproject bin and lib
+ directories should pretty much ignore everything * Ignore *.o in
+ codecs/ilbc
+
+ * res/Makefile: Fix parallel make for res_asterisk_rtp. Fixes a
+ build regression introduced in r369517 "Add support for
+ ICE/STUN/TURN in res_rtp_asterisk and chan_sip." [1]. [1]
+ http://svnview.digium.com/svn/asterisk?view=revision&revision=369517
+ When compiling asterisk in parallel like: $ make -j 10 It's
+ possible to get errors like the following:
+ .pjlib-util-test-x86_64-unknown-linux-gnu.depend:120: *** missing
+ separator. Stop. make[4]: *** [depend] Error 2 make[3]: *** [dep]
+ Error 1 make[2]: ***
+ [/home/sruffell/asterisk-working/res/pjproject/pjnath/lib/libpjnath-x86_64-unknown-linux-gnu.a]
+ Error 2 make[3]: warning: jobserver unavailable: using -j1. Add
+ `+' to parent make rule. This is because the build system is
+ trying to build each of the libraries in pjproject in parallel.
+ Now the build will build pjproject in a single job and link the
+ results into res_asterisk_rtp. Parallel builds, on one test
+ system, saves ~1.5 minutes from a default Asterisk build: Single
+ job: $ git clean -fdx >/dev/null && time ( ./configure >/dev/null
+ 2>&1 && make >/dev/null 2>&1 ) real 2m34.529s user 1m41.810s sys
+ 0m15.970s Parallel make: $ git clean -fdx >/dev/null && time (
+ ./configure >/dev/null 2>&1 && make -j10 >/dev/null 2>&1 ) real
+ 1m2.353s user 2m39.120s sys 0m18.850s (closes issue
+ ASTERISK-20362) Reported by: Shaun Ruffel Patches:
+ 0001-res_asterisk_rtp-Fix-build-error-when-using-parallel.patch
+ uploaded by Shaun Ruffel (License #5417)
+
+2012-09-07 02:26 +0000 [r372531-372583] Matthew Jordan <mjordan@digium.com>
+
+ * /, apps/app_minivm.c: Free ast_str objects when temp file fails
+ to be created in MiniVM The previous commit (r372554) was from a
+ patch that was written before r366880, which ensured that ast_str
+ objects allocated in the sendmail routine were free'd in off
+ nominal paths. This commit frees the string objects in the off
+ nominal path introduced in r372554. (issue ASTERISK-17133)
+ Reported by: Tzafrir Cohen ........ Merged revisions 372581 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372582 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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) ........ Merged revisions 372554 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372555 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * apps/app_queue.c: Update QueueMemberStatus event documentation to
+ include member status values The Status: header in a
+ QueueMemberStatus event (and other QueueMember* events) is the
+ numeric value of the device state corresponding to that Queue
+ Member. As those values are not exactly obvious, listing them in
+ the documentation is useful. Matt Riddell reported this
+ indirectly through the wiki page. (closes issue ASTERISK-20243)
+ Reported by: Matt Riddell
+
+2012-09-06 22:12 +0000 [r372523] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Fix loss of MOH on an ISDN channel when
+ parking a call for the second time. Using the AMI redirect action
+ to take an ISDN call out of a parking lot causes the MOH state to
+ get confused. The redirect action does not take the call off of
+ hold. When the call is subsequently parked again, the call no
+ longer hears MOH. * Make chan_dahdi/sig_pri restart MOH on
+ repeated AST_CONTROL_HOLD frames if it is already in a state
+ where it is supposed to be sending MOH. The MOH may have been
+ stopped by other means. (Such as killing the generator.) This
+ simple fix is done rather than making the AMI redirect action
+ post an AST_CONTROL_UNHOLD unconditionally when it redirects a
+ channel and thus potentially breaking something with an
+ unexpected AST_CONTROL_UNHOLD. (closes issue ABE-2873) Patches:
+ jira_abe_2873_c.3_bier.patch (license #5621) patch uploaded by
+ rmudgett ........ Merged revisions 372521 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 372522 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-06 21:42 +0000 [r372519] Kinsey Moore <kmoore@digium.com>
+
+ * /, apps/app_queue.c: Ensure listed queues are not offered for
+ completion When using tab-completion for the list of queues on
+ "queue reset stats" or "queue reload
+ {all|members|parameters|rules}", the tab-completion listing for
+ further queues erroneously listed queues that had already been
+ added to the list. The tab-completion listing now only displays
+ queues that are not already in the list. (closes issue AST-963)
+ Reported-by: John Bigelow ........ Merged revisions 372517 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372518 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-06 18:55 +0000 [r372500] dsessions <dsessions@localhost>:
+
+ * channels/chan_sip.c, configs/res_ldap.conf.sample: 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 15:56 +0000 [r372473] Jonathan Rose <jrose@digium.com>
+
+ * /, UPGRADE-1.8.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)
+ ........ Merged revisions 372471 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372472 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-06 14:30 +0000 [r372446] Kinsey Moore <kmoore@digium.com>
+
+ * /, apps/app_queue.c: Ensure "rules" is tab-completable for "queue
+ show" Previously, tabbing at the end of "queue show" produced a
+ list of available queues about which information could be shown,
+ but did not include an alternative command, "rules", to access
+ information about queue rules. The "rules" item should now be
+ shown in the list of tab-completable items. (closes issue
+ AST-958) Reported-by: John Bigelow ........ Merged revisions
+ 372444 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 372445 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-06 02:50 +0000 [r372392-372419] Matthew Jordan <mjordan@digium.com>
+
+ * /, pbx/pbx_dundi.c: Fix DUNDi message routing bug when
+ neighboring peer is unreachable Consider a scenario where DUNDi
+ peer PBX1 has two peers that are its neighbors, PBX2 and PBX3,
+ and where PBX2 and PBX3 are also neighbors. If the connection is
+ temporarily broken between PBX1 and PBX3, PBX1 should not include
+ PBX3 in the list of peers it sends to PBX2 in a DPDISCOVER
+ message, as it cannot send messages to PBX3. If it does, PBX2
+ will assume that PBX3 already received the message and fail to
+ forward the message on to PBX3 itself. This patch fixes this by
+ only including peers in a DPDISCOVER message that are reachable
+ by the sending node. This includes all peers with an empty
+ address (00:00:00:00:00:00) and that are have been reached by a
+ qualify message. This patch also prevents attempting to qualify a
+ dynamic peer with an empty address until that peer registers.
+ (closes issue ASTERISK-19309) Reported by: Peter Racz patches:
+ dundi_routing.patch uploaded by Peter Racz (license 6290) The
+ patch uploaded by Peter was modified slightly for this commit.
+ ........ Merged revisions 372417 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372418 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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.
+ ........ Merged revisions 372390 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372391 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-05 19:43 +0000 [r372373] Richard Mudgett <rmudgett@digium.com>
+
+ * main/dsp.c, /: Fix compile error. ........ Merged revisions
+ 372372 from http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-05 19:24 +0000 [r372365] Kinsey Moore <kmoore@digium.com>
+
+ * main/manager.c, /: Correct documentation for ModuleLoad AMI
+ action The documentation incorrectly listed 'rtp' as a reloadable
+ subsystem and left out many other reloadable subsystems. It is
+ now also documented that subsystems may only be reloaded, not
+ loaded or unloaded. (closes issue AST-977) Reported-by: John
+ Bigelow ........ Merged revisions 372354 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372358 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-05 18:46 +0000 [r372342] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/dsp.c, /: dsp.c: in ast_mf_detect_init incorrectly sets
+ goertzel samples to 160, should be MF_GSIZE Related
+ https://reviewboard.asterisk.org/r/2097/ ........ Merged
+ revisions 372339 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372341 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-05 18:36 +0000 [r372340] Kinsey Moore <kmoore@digium.com>
+
+ * main/pbx.c, /: Ensure counts generated in
+ manager_show_dialplan_helper are correct When
+ manager_show_dialplan_helper was written, the counter increment
+ for the total number of contexts was placed with the extensions
+ increment instead of in the enclosing loop. This function should
+ now generate correct context counts. (closes issue AST-970)
+ Reported-by: John Bigelow ........ Merged revisions 372337 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372338 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-05 17:35 +0000 [r372327-372328] Richard Mudgett <rmudgett@digium.com>
+
+ * res/res_rtp_asterisk.c: Fix coding guidelines issue with a recent
+ commit.
+
+ * res/res_rtp_asterisk.c: Fix RTP/RTCP read error message
+ confusion. The RTP/RTCP read error message can report "fail:
+ success" when the read failure is because of an ICE failure. *
+ Changed __rtp_recvfrom() to generate a PJ ICE message when ICE
+ fails. * Changed RTP/RTCP read error message to indicate an
+ unspecified error when errno is zero. (closes issue
+ ASTERISK-20288) Reported by: Joern Krebs Patches:
+ jira_asterisk_20288_err_msg.patch (license #5621) patch uploaded
+ by rmudgett (modified)
+
+2012-09-05 16:04 +0000 [r372311] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_rtp_asterisk.c, main/rtp_engine.c,
+ include/asterisk/rtp_engine.h: Re-fix sending unnegotiated
+ payloads during a P2P RTP bridge. The previous fix still would
+ look in the static_RTP_PT table, which is inappropriate since we
+ specifically want to find a codec that has been negotiated.
+ (closes issue ASTERISK-20296) reported by NITESH BANSAL Patches:
+ codec_negotiation.patch Uploaded by NITESH BANSAL (License #6418)
+
+2012-09-05 13:47 +0000 [r372289] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_voicemail.c, /: Fix memory leaks in app_voicemail when
+ using IMAP storage or realtime config This patch fixes two memory
+ leaks: 1. When find_user is called with NULL as its first
+ parameter, the voicemail user returned is allocated on the heap.
+ The inboxcount2 function uses find_user in such a fashion when
+ counting new messages, and fails to free the resulting voicemail
+ user object. 2. When populate_defaults is called on a voicemail
+ user, it wipes whatever flags have been set on the object by
+ copying over the global flags object. If the VM_ALLOCED flag was
+ ste on the voicemail user prior to doing so, that flag is
+ removed. This leaks the voicemail user when free_user is later
+ called. (closes issue ASTERISK-19155) Reported by: Filip Jenicek
+ patches: asterisk.patch2 uploaded by Filip Jenicek (license 6277)
+ Patch slightly modified for this commit. Review:
+ https://reviewboard.asterisk.org/r/2096 ........ Merged revisions
+ 372268 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 372288 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-05 12:17 +0000 [r372266] Michael L. Young <elgueromexicano@gmail.com>
+
+ * res/res_rtp_asterisk.c: Fix breakage caused by last merge.
+ Missing a variable for 11 and trunk.
+
+2012-09-05 07:41 +0000 [r372214-372241] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/dsp.c, /: dsp.c: Fix multiple issues when no-interdigit
+ delay is present, and fast DTMF 50ms/50ms Revert DTMF hit/miss
+ detector to original -r349249 method with some changes, remove
+ unnecessary; 1. reseting of hits=0, when no signal, only need to
+ set it once. 2. incrementing of hits, when the hit is the same as
+ the current hit. 3. setting of lasthit, when it's the same as
+ before. Change HITS_TO_BEGIN to 2, MISSES_TO_END to 3 & 3
+ spelling mistakes (closes issue ASTERISK-19610) alecdavis
+ (license 585) Reported by: Jean-Philippe Lord Tested by:
+ alecdavis Review: https://reviewboard.asterisk.org/r/2085/
+ ........ Merged revisions 372239 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372240 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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/ ........ Merged
+ revisions 372212 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372213 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-05 04:52 +0000 [r372199] Michael L. Young <elgueromexicano@gmail.com>
+
+ * res/res_rtp_asterisk.c, /: Fix Incrementing Sequence Number For
+ Retransmitted DTMF End Packets In Asterisk 1.4+, a fix was put in
+ place to increment the sequence number for retransmitted DTMF end
+ packets. With the introduction of the RTP engine API in 1.8, the
+ sequence number was no longer being incremented. This patch fixes
+ this regression as well as cleans up a few lines that were not
+ doing anything. (closes issue ASTERISK-20295) Reported by: Nitesh
+ Bansal Tested by: Michael L. Young Patches:
+ 01_rtp_event_seq_num.patch uploaded by Nitesh Bansal (license
+ 6418) asterisk-20295-dtmf-fix-cleanup.diff uploaded by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2083/ ........ Merged
+ revisions 372185 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372198 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-05 02:25 +0000 [r372175] Matthew Jordan <mjordan@digium.com>
+
+ * cel/cel_pgsql.c, /: Fix memory leak when CEL is successfully
+ written to PostgreSQL database PQClear is not called when the
+ result object of a call to PQExec has a status of
+ PGRES_COMMAND_OK. Interestingly enough, the off nominal case was
+ handled properly, so this memory leak only occurred when CEL
+ records were successfully written. This patch properly clears the
+ result in the nominal code path. (closes issue ASTERISK-19991)
+ Reported by: Etienne Lessard Tested by: Etienne Lessard patches:
+ mem_leak_cel_pgsql.patch uploaded by Etienne Lessard (license
+ #6394) ........ Merged revisions 372158 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372165 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-09-04 15:48 +0000 [r372135-372137] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix issue where SIP devices were not
+ notified when custom devices changed to "ringing". The problem
+ had to do with logic used when checking for what the oldest
+ ringing channel was. The problem was that if no channel was
+ found, then no notification would be sent. For custom device
+ states, there is no associated channel, so no notification would
+ get sent. This fixes the issue by still sending the notification
+ even if no associated channel can be found for a ringing device
+ state change. (closes issue ASTERISK-20297) Reported by Noah
+ Engelberth
+
+ * main/config_options.c, apps/app_confbridge.c: Prevent crash from
+ using app_page with no confbridge.conf file provided. Also
+ prevents other potential crashes when using aco API with
+ uninitialized aco_info structs. (closes issue ASTERISK-20305)
+ reported by Noah Engelberth Tested by Noah Engelberth Review:
+ https://reviewboard.asterisk.org/r/2086
+
+2012-08-31 21:14 +0000 [r372118] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_rtp_asterisk.c: Prevent local RTP bridges from sending
+ inappropriate formats to participants. A change for Asterisk 11
+ caused a check for failure to incorrectly check the return value.
+ This resulted in the possibility of transmitting media that a
+ party had not negotiated. If this media happened to be G.729,
+ then this could potentially result in one-way audio if no G.729
+ translators are installed. (closes issue ASTERISK-20296) reported
+ by NITESH BANSAL
+
+2012-08-30 20:54 +0000 [r372050-372091] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Prevent crash on shutdown due to refcount
+ error on queues container. When app_queue is unloaded, the queues
+ container has its refcount decremented, potentially to 0. Then
+ the taskprocessor responsible for handling device state changes
+ is unreferenced. If the taskprocessor happens to be just about to
+ run its task, then it will create and destroy an iterator on the
+ queues container. This can cause the refcount on the queues
+ container to increase to 1 and then back to 0. Going back to 0 a
+ second time results in double frees. This failure was seen
+ periodically in the testsuite when Asterisk would shut down.
+ ........ Merged revisions 372089 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 372090 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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) ........ Merged revisions
+ 372048 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 372049 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-30 16:24 +0000 [r371963-372028] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_iax2.c: AST-2012-013: Resolve ACL rules being
+ ignored during calls by some IAX2 peers When an IAX2 call is made
+ using the credentials of a peer defined in a dynamic Asterisk
+ Realtime Architecture (ARA) backend, the ACL rules for that peer
+ are not applied to the call attempt. This allows for a remote
+ attacker who is aware of a peer's credentials to bypass the ACL
+ rules set for that peer. This patch ensures that the ACLs are
+ applied for all peers, regardless of their storage mechanism.
+ (closes issue ASTERISK-20186) Reported by: Alan Frisch Tested by:
+ mjordan, Alan Frisch
+
+ * /: Block r372020
+
+ * 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 ........ Merged revisions 371998 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371999 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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) ........ Merged revisions 371961
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 371962 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-29 22:38 +0000 [r371950] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_meetme.c: Fix compile errors.
+
+2012-08-29 21:07 +0000 [r371921] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_meetme.c: app_meetme: Adding test events for
+ following activity in MeetMe. ........ Merged revisions 371919
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 371920 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-29 19:56 +0000 [r371862-371893] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Fix theoretical compile error with HAVE_EPOLL.
+ Really shows how much epoll is used since it had not been
+ reported yet.
+
+ * 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) ........ Merged revisions 371888 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371890 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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 ........ Merged
+ revisions 371860 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371861 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-29 17:25 +0000 [r371845] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Send 408 on retransmit timeout
+ instead of 603 (closes issue ASTERISK-20124) Reported by: Walter
+ Doekes ........ Merged revisions 371824 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371825 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-27 21:50 +0000 [r371784-371790] Mark Michelson <mmichelson@digium.com>
+
+ * configs/agents.conf.sample, /: Fix misleading documentation in
+ agents.conf.sample regarding ackcall usage. The documentation
+ made it sound as if the DTMF acknowledgment was needed at the
+ time the agent logs in, rather than when the agent is called.
+ This is likely a relic from the days when there were multiple
+ ways of logging in agents. (closes issue AST-962) reported by
+ Steve Pitts ........ Merged revisions 371787 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371789 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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. ........ Merged revisions 371782 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371783 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-27 18:14 +0000 [r371753] dlee <dlee@localhost>:
+
+ * res/pjproject/pjlib-util/bin, res/pjproject/pjnath/build/output,
+ res/pjproject/pjlib/bin, res/pjproject/pjlib-util/build/output,
+ res/pjproject/pjnath/bin, res/pjproject/pjlib/build/output:
+ svn:ignore pjproject bin & output for all platforms.
+
+2012-08-27 17:51 +0000 [r371749-371750] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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 ........ Merged revisions
+ 371747 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 371748 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /: Re-add merge and block properties.
+
+2012-08-27 16:55 +0000 [r371720] dlee <dlee@localhost>:
+
+ * main/lock.c, /: Fixes ast_rwlock_timed[rd|wr]lock for BSD and
+ variants. The original implementations simply wrap pthread
+ functions, which take absolute time as an argument. The spinlock
+ version for systems without those functions treated the argument
+ as a delta. This patch fixes the spinlock version to be
+ consistent with the pthread version. (closes issue
+ ASTERISK-20240) Reported by: Egor Gorlin Patches: lock.c.patch
+ uploaded by Egor Gorlin (license 6416) ........ Merged revisions
+ 371718 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-08-27 14:07 +0000 [r371692] Kinsey Moore <kmoore@digium.com>
+
+ * /, main/utils.c: Implement workaround for BETTER_BACKTRACES crash
+ When compiling with BETTER_BACKTRACES enabled, Asterisk will
+ sometimes crash when "core show locks" is run. This happens
+ regularly in the testsuite since several tests run "core show
+ locks" to help with debugging. This seems to be a fault with
+ libraries on certain operating systems (notably CentOS 6.2/6.3)
+ running on virtual machines and utilizing gcc 4.4.6. (closes
+ issue ASTERISK-20090) ........ Merged revisions 371690 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371691 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-26 23:07 +0000 [r371664] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/dsp.c, /: mf_detect: incorrectly used DTMF_GSIZE instead of
+ MF_GSIZE ........ Merged revisions 371662 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371663 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-22 15:54 +0000 [r371619] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_motif.c: Add support for call-id logging to
+ chan_motif. Review: https://reviewboard.asterisk.org/r/2077/
+
+2012-08-21 20:54 +0000 [r371592] Mark Michelson <mmichelson@digium.com>
+
+ * 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, 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: 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
+ ........ Merged revisions 371590 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371591 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-20 20:09 +0000 [r371571] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_rtp_asterisk.c: Use thread-local storage to store
+ pj_thread_descs. pj_thread_register() takes a parameter of type
+ pj_thread_desc. It was assumed that pj_thread_register either
+ used this item temporarily or made a copy of it. Unfortunately,
+ all it does is keep a pointer to the structure in thread-local
+ storage. This means that if our pj_thread_desc goes out of scope,
+ then pjlib will be referencing bogus data quite often, most
+ commonly on operations involving a pj_mutex_t. In our case, our
+ pj_thread_desc was on the stack and went out of scope very
+ shortly after registering our thread with pjlib. With this
+ change, the pj_thread_desc is stored in thread-local storage so
+ the pointer that pjlib keeps in thread-local storage will
+ reference legitimate memory. (closes issue ASTERISK-20237)
+ reported by Jeremy Pepper Patches: ASTERISK-20237.patch uploaded
+ by Mark Michelson (license #5049) Tested by Jeremy Pepper
+
+2012-08-20 15:34 +0000 [r371546] Kinsey Moore <kmoore@digium.com>
+
+ * main/udptl.c, /: Ignore recovered zero-length secondary UDPTL
+ packets In some cases, recovering lost packets using the
+ secondary packet recovery mechanism with UDPTL/T.38 can result in
+ the recovery of zero-length packets. These must be ignored or the
+ frame generated from them can cause segfaults and allocation
+ failures. (closes issue ASTERISK-19762) (closes issue
+ ASTERISK-19373) Reported-by: Benjamin (bulkorok) Reported-by: Rob
+ Gagnon (rgagnon) ........ Merged revisions 371544 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371545 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-18 02:35 +0000 [r371492-371530] Matthew Jordan <mjordan@digium.com>
+
+ * /: Recorded merge of revisions 371529 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Remove
+ old debug code from http configuration loading (closes issue
+ ASTERISK-20254) Reported by: Andrew Latham Patches: http.diff
+ uploaded by Andrew Latham (license #5985)
+
+ * main/http.c: Remove old debug code from http configuration
+ loading (closes issue ASTERISK-20254) Reported by: Andrew Latham
+ Patches: http.diff uploaded by Andrew Latham (license #5985)
+
+ * res/res_xmpp.c: Fix typo in JabberSend that looked for '2'
+ instead of '@' in recipient argument The summary says about all
+ there is to say. (closes issue ASTERISK-20239) Reported by:
+ Gregory Porras
+
+ * funcs/func_hangupcause.c: Make the name of the "HangupCauseClear"
+ application consistent The name of the "HangupCauseClear"
+ application is "HangupCauseClear", not "HangupcauseClear". The
+ incorrect case of 'cause' caused the XML documentation to not
+ register properly. As an aside, this commit message felt very
+ awkward, but I'm not sure how else to note that "X", which has to
+ be "X", was referred to as "x". (closes issue ASTERISK-20253)
+ Reported by: Andrew Latham Patches: hangupcause.diff uploaded by
+ Andrew Latham (license #5985)
+
+ * build_tools/cflags.xml, utils/utils.xml, res/res_fax.c,
+ sounds/sounds.xml, res/res_curl.c: Update module support level on
+ a variety of modules and compiler options Some core support
+ modules and compiler options were no longer tagged with a module
+ support level. This patch adds 'core' back to those options. Note
+ that this patch modifies a few of the patches provided by Andrew
+ Latham slightly. res_curl and res_fax are both 'core' supported
+ modules. (closes issue ASTERISK-20215) Reported by: Andrew Latham
+ Tested by: mjordan Patches: astcanary.diff (license #5985)
+ uploaded by Andrew Latham cflagsxml.diff (license #5985) uploaded
+ by Andrew Latham curl_fax.diff (license #5985) uploaded by Andrew
+ Latham soundsxml.diff (license #5985) uploaded by Andrew Latham
+
+ * 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 ........ Merged revisions 371469 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371491 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-17 19:49 +0000 [r371482] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: When a peer registers using WebSocket do not
+ resolve the Contact provided. (closes issue ASTERISK-20238)
+ Reported by: james.mortensen
+
+2012-08-17 15:58 +0000 [r371438] Kinsey Moore <kmoore@digium.com>
+
+ * main/loader.c, /: Add instrumentation to subsystem reloads When
+ Asterisk is built with TEST_FRAMEWORK defined, Asterisk will now
+ generate TestEvent AMI events on subsystem reloads such as cdr,
+ dnsmgr, extconfig, etc. (issue PQ-1126) ........ Merged revisions
+ 371436 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 371437 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-17 12:24 +0000 [r371426] Joshua Colp <jcolp@digium.com>
+
+ * res/res_format_attr_h264.c: Add some additional H.264 attributes,
+ "max-smbps" and "max-fps", for passthrough. (closes issue
+ ASTERISK-20206) Reported by: ddkprog Patches:
+ res_format_attr_h264.c.diff uploaded by ddkprog (license 6008)
+
+2012-08-17 12:23 +0000 [r371425] Russell Bryant <russell@russellbryant.com>
+
+ * res/res_rtp_asterisk.c: rtp: Ensure defaults are set without
+ rtp.conf. While building up a new install to test chan_motif, I
+ ran into a failure due to icesupport being disabled. This was due
+ to me not having an rtp.conf. It was intended in the code for it
+ to be enabled by default, but it was only applied if rtp.conf
+ existed. This patch updates res_rtp_asterisk to be consistent in
+ how it handles defaults. A few options didn't have their default
+ values set globally, including icesupport. They are now set and
+ icesupport is enabled by default, even if you do not have an
+ rtp.conf.
+
+2012-08-16 23:02 +0000 [r371399] Terry Wilson <twilson@digium.com>
+
+ * main/config.c, /: Handle integer over/under-flow in
+ ast_parse_args The strtol family of functions will return
+ *_MIN/*_MAX on overflow. To detect when an overflow has happened,
+ errno must be set to 0 before calling the function, then checked
+ afterward. (closes issue ASTERISK-20120) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/2073/ ........ Merged
+ revisions 371392 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371398 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-16 22:44 +0000 [r371395] Kinsey Moore <kmoore@digium.com>
+
+ * 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) ........
+ Merged revisions 371393 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371394 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-16 19:43 +0000 [r371355-371382] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Use pvt outgoing_call variable
+ to set Remote-Party-ID Header Previously the pvt SIP_OUTGOING
+ flag was used instead, which will frequently flip during
+ reinvites. (closes issue AST-897) Reported by: Thomas Arimont
+ ........ Merged revisions 371357 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371358 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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 ........ Merged revisions 371337 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371338 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-15 23:28 +0000 [r371324] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Fix bug where final queue member would not
+ be removed from memory. If a static queue had realtime members,
+ then there could be a potential for those realtime members not to
+ be properly deleted from memory. If the queue's members were
+ loaded from realtime and then all the members were deleted from
+ the backend, then the queue would still think these members
+ existed. The reason was that there was a short- circuit in code
+ such that if there were no members found in the backend, then the
+ queue would not be updated to reflect this. Note that this only
+ affected static queues with realtime members. Realtime queues
+ with realtime members were unaffected by this issue. (closes
+ issue ASTERISK-19793) reported by Marcus Haas ........ Merged
+ revisions 371306 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371313 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-15 20:40 +0000 [r371295] Michael L. Young <elgueromexicano@gmail.com>
+
+ * channels/chan_sip.c: Fix Segfault When Registering SIP Over
+ WebSockets The helper function, get_address_family_filter, in
+ chan_sip for dns resolution by address family was not recognizing
+ the websockets transport and resulting in a null pointer being
+ sent to functions in netsock2, in an attempt to determine if we
+ are bound to ANY address ([::]) or not. This patch fixes this
+ issue by handling the transport types SIP_TRANSPORT_WS and
+ SIP_TRANSPORT_WSS which results in a sock address being set
+ properly for use in determining the address family. (closes issue
+ ASTERISK-20221) Reported by: Sven Beisiegel Tested by: Sven
+ Beisiegel, James Mortensen Patches:
+ asterisk-20221-ws-family-filter.diff uploaded by Michael L. Young
+ (license 5026)
+
+2012-08-15 20:17 +0000 [r371258-371272] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Avoid unconditional NULLing of mwipvt on
+ relatedpeer on SIP dialog destruction The other instance of this
+ bug was fixed by jcolp/file in r121496. If we are destroying a
+ dialog only set the MWI dialog pointer on the related peer to
+ NULL if it is the dialog currently being destroyed. (closes issue
+ ASTERISK-20119) Patch-by: Misha Vodsedalek ........ Merged
+ revisions 371270 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371271 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/sig_ss7.c, channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/chan_sip.c, channels/chan_iax2.c, channels/sig_pri.c:
+ Add HANGUPCAUSE information to callee channels This adds
+ HANGUPCAUSE information to called channels so that hangup
+ handlers can, in conjunction with predial dialplan execution,
+ access the hangupcause information when the dialed channel hangs
+ up on a one-to-one basis instead of a many-to-one basis as with
+ HANGUPCAUSE usage on the caller channel. Review:
+ https://reviewboard.asterisk.org/r/2069/ (closes issue
+ ASTERISK-20198)
+
+2012-08-13 20:28 +0000 [r371227] Kinsey Moore <kmoore@digium.com>
+
+ * main/loader.c, /, apps/app_meetme.c: Add test instrumentation
+ This adds test instrumentation for loading and unloading of
+ modules and for certain actions in MeetMe to be used in the
+ testsuite or any other consumer of AMI events. These will only be
+ generated when Asterisk is built with TEST_FRAMEWORK enabled.
+ (issue PQ-1131) (issue PQ-1133) ........ Merged revisions 371201
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 371203 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-13 19:52 +0000 [r371200] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Fix problem where incorrect pointer was
+ checked for nullity. ........ Merged revisions 371198 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371199 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-10 22:03 +0000 [r371146] Richard Mudgett <rmudgett@digium.com>
+
+ * CHANGES: Update CHANGES for private party ID.
+
+2012-08-10 21:32 +0000 [r371143] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Fix a couple of documentation problems in
+ app_queue.c * The RemoveQueueMember app made mention of options
+ that could be passed in, but no options are supported. I have
+ removed the listing of options from the documentation. * The
+ RQMSTATUS variable did not list "NOTDYNAMIC" as a possible value
+ that could be set. (closes issue AST-949) reported by Steve Pitts
+ (closes issue AST-954) reported by Steve Pitts ........ Merged
+ revisions 371141 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371142 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-10 20:08 +0000 [r371121] Matthew Jordan <mjordan@digium.com>
+
+ * / (added): _ _ _ _ _ _ / \ ___| |_ ___ _ __(_)___| | __ / | / | /
+ _ \ / __| __/ _ \ '__| / __| |/ / | | | | / ___ \__ \| | __/ | |
+ \__ \ < | | | | /_/ \_\___/\__\___|_| |_|___/_|\_\ |_| |_|
+ Because it's one greater than 10.
+
+2012-08-10 19:54 +0000 [r371120] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, channels/chan_misdn.c, channels/chan_sip.c,
+ main/channel_internal_api.c, main/features.c,
+ include/asterisk/channel.h, channels/sig_pri.c,
+ funcs/func_callerid.c, main/cli.c: Add private representation of
+ caller, connected and redirecting party ids. This patch adds the
+ feature "Private representation of caller, connected and
+ redirecting party ids", as previously discussed with us (DATUS)
+ and Digium. 1. Feature motivation Until now it is quite difficult
+ to modify a party number or name which can only be seen by
+ exactly one particular instantiated technology channel
+ subscriber. One example where a modified party number or name on
+ one channel is spread over several channels are supplementary
+ services like call transfer or pickup. To implement these
+ features Asterisk internally copies caller and connected ids from
+ one channel to another. Another example are extension
+ subscriptions. The monitoring entities (watchers) are notified of
+ state changes and - if desired - of party numbers or names which
+ represent the involving call parties. One major feature where a
+ private representation of party names is essentially needed, i.e.
+ where a party name shall be exclusively signaled to only one
+ particular user, is a private user-specific name resolution for
+ party numbers. A lookup in a private destination-dependent
+ telephone book shall provide party names which cannot be seen by
+ any other user at any time. 2. Feature Description This feature
+ comes along with the implementation of additional private party
+ id elements for caller id, connected id and redirecting ids
+ inside Asterisk channels. The private party id elements can be
+ read or set by the user using Asterisk dialplan functions. When a
+ technology channel is initiating a call, receives an internal
+ connected-line update event, or receives an internal redirecting
+ update event, it merges the corresponding public id with the
+ private id to create an effective party id. The effective party
+ id is then used for protocol signaling. The channel technologies
+ which initially support the private id representation with this
+ patch are SIP (chan_sip), mISDN (chan_misdn) and PRI
+ (chan_dahdi). Once a private name or number on a channel is set
+ and (implicitly) made valid, it is generally used for any further
+ protocol signaling until it is rewritten or invalidated. To
+ simplify the invalidation of private ids all internally generated
+ connected/redirecting update events and also all
+ connected/redirecting update events which are generated by
+ technology channels -- receiving regarding protocol information -
+ automatically trigger the invalidation of private ids. If not
+ using the private party id representation feature at all, i.e. if
+ using only the 'regular' caller-id, connected and redirecting
+ related functions, the current characteristic of Asterisk is not
+ affected by the new extended functionality. 3. User interface
+ Description To grant access to the private name and number
+ representation from the Asterisk dialplan, the CALLERID,
+ CONNECTEDLINE and REDIRECTING dialplan functions are extended by
+ the following data types. The formats of these data types are
+ equal to the corresponding regular 'non-private' already existing
+ data types: CALLERID: priv-all priv-name priv-name-valid
+ priv-name-charset priv-name-pres priv-num priv-num-valid
+ priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid
+ priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE:
+ priv-name priv-name-valid priv-name-pres priv-name-charset
+ priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr
+ priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag
+ REDIRECTING: priv-orig-name priv-orig-name-valid
+ priv-orig-name-pres priv-orig-name-charset priv-orig-num
+ priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan
+ priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type
+ priv-orig-subaddr-odd priv-orig-tag priv-from-name
+ priv-from-name-valid priv-from-name-pres priv-from-name-charset
+ priv-from-num priv-from-num-valid priv-from-num-pres
+ priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid
+ priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag
+ priv-to-name priv-to-name-valid priv-to-name-pres
+ priv-to-name-charset priv-to-num priv-to-num-valid
+ priv-to-num-pres priv-to-num-plan priv-to-subaddr
+ priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd
+ priv-to-tag Reported by: Thomas Arimont Review:
+ https://reviewboard.asterisk.org/r/2030/
+
+2012-08-10 17:56 +0000 [r371113] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix a comparison that was causing presence
+ tests to fail. A recent change made it so that device state
+ changes that were not actual "changes" would not get reported to
+ subscribers. The problem was that this inadvertently blocked
+ presence updates as well.
+
+2012-08-10 16:49 +0000 [r371059-371091] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c, /: remove ALREADYGONE flag on ooh323 call
+ data by ooh323_indicate (CONGESTION/BUSY) due to call hasn't gone
+ there really. This indication arrive from asterisk core not h.323
+ stack (closes issue ASTERISK-19308) Reported by: Dmitry Melekhov
+ Patches: ASTERISK-19308.patch ........ Merged revisions 371089
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 371090 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * addons/ooh323c/src/ooGkClient.c, /: Send re-register packets by
+ GRQ (gatekeeper request) interval (close issue ASTERISK-20094)
+ Patches: ASTERISK-20094-2.patch ........ Merged revisions 371060
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 371061 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * addons/ooh323c/src/ooTimer.c: restore calling cb functions by
+ timer expire this was broken in rev 369602
+
+2012-08-10 02:07 +0000 [r371052] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: Fix pickup extension channel reference error.
+ You cannot unref a pointer and then expect to ref it again later.
+ * Fix potential NULL pointer deref if the call pickup search
+ fails.
+
+2012-08-09 21:35 +0000 [r371036-371043] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c: Introdue 'ooh323 show gk' cli command that
+ show status of connection to H.323 Gatekeeper (GkClient state)
+
+ * addons/ooh323c/src/ooGkClient.c, /: Fix to resend GRQ/RRQ if RRJ
+ (registration reject) is received (close issue ASTERISK-20094)
+ Patches: ASTERISK-20094.patch ........ Merged revisions 371011
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 371022 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-09 19:22 +0000 [r371030] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ channels/sig_pri.c, channels/sig_ss7.c: Use better libss7
+ detection test and move libpri compile test. ........ Merged
+ revisions 371012 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 371013 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-09 18:28 +0000 [r371010] Alexandr Anikin <may@telecom-service.ru>
+
+ * /, addons/ooh323c/src/ooh323ep.c: change opening h323 logfile
+ with append mode instead of overwrite ........ Merged revisions
+ 370988 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 370989 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-09 17:40 +0000 [r370987] Kinsey Moore <kmoore@digium.com>
+
+ * /, apps/app_meetme.c: Correct documentation for the MeetMe x flag
+ The documentation for the x flag for MeetMe incorrectly described
+ its function as closing down the conference when the last marked
+ user left. It actually causes the users with that flag to leave
+ the conference when the last marked user exits. The functionality
+ of this flag is not changing. ........ Merged revisions 370985
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 370986 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-09 14:52 +0000 [r370979] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c, channels/chan_sip.c, include/asterisk/pbx.h,
+ channels/sip/include/sip.h: Extend extension state callbacks to
+ have more information. Quote from review board: This patch
+ extends the extension state callbacks so that monitoring channels
+ (as chan_sip) get more information of the devices which are
+ responsible for an extension state change. The additional
+ information is needed by chan_sip to present names/numbers of the
+ caller and callee in an early-state SIP notification. Users of
+ extenstion state callback not interested in the additional
+ information are not affected by the changes. Motivation: to
+ present the involved party's name/number in an early-state
+ nofification (used by the notified device as a pickup offer) one
+ after another so that a user can see which call he will pick up
+ in an undirected pickup. Such a pickup offer to a user shall
+ indicate the same call (number/name-A calls number/name-B) as the
+ call which would be picked up when an undirected pickup is
+ executed. Users interested in additional state info must use the
+ new functions ast_extension_state_add_extended() resp.
+ ast_extension_state_add_destroy_extended() to register an
+ extended state callback. When the callback is registered this
+ way, an extra member device_state_info of struct
+ ast_state_cb_info is passed to the callback in addition to the
+ aggregated extension state. This container holds an object for
+ every device of the monitored extension hint consisting of the
+ device name, the device state and a channel reference to the
+ channel which (presumably) caused the device state. The
+ information is used by chan_sip for early-state notifications.
+ When the state of a device changes and the new state contains
+ AST_EVENT_RINGING, an early-state notification is sent to the
+ subscribed devices with the caller/callee names/numbers of the
+ oldest ringing channel of the monitored extension. The notified
+ user may then invoke a direct pickup, which will pickup exactly
+ this channel. Users of the old non-extended callbacks will only
+ be called when the aggregated state did change (same behavior as
+ before). Users of the extended callback will also be called when
+ the state is unchanged but does contain AST_EVENT_RINGING. That
+ could be the case if two channels are ringing at one device and
+ one of them hangs up, so the aggregated state does not change.
+ This way the monitoring channel can create a new early-state
+ notification with the now ringing party-ids. Review:
+ https://reviewboard.asterisk.org/r/2048 This contribution comes
+ from Guenther Kelleter
+
+2012-08-09 14:36 +0000 [r370978] Jonathan Rose <jrose@digium.com>
+
+ * pbx/pbx_dundi.c, CHANGES: DUNDi: Add CLI commands DUNDi show
+ cache and DUNDi show hints (closes issue ASTERISK-18390) Reported
+ by: Peter Racz Patches: dundi_cli_cache.patch.v2 uploaded by
+ Peter Racz (license #6290)
+ ASTERISK-18390_dundi_cli_cache_jrose_mods_v2.diff uploaded by
+ Jonathan Rose (license #6182)
+
+2012-08-08 22:45 +0000 [r370955] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, apps/app_chanspy.c: Fix Not Unreferencing A Spied Channel When
+ a channel hangs up while being spied upon and the option to exit
+ the ChanSpy application when the spied on channel hangs up is
+ set, ast_autochan_destroy is not being called and therefore a
+ reference to the spied upon channel is not removed. The symptom
+ being reported was that when using func_group in the dialplan and
+ calling "group show channels" at the cli, the spied upon channel
+ was still being shown while "core show channels" showed that the
+ channel was not up. This patch calls ast_autochan_destroy when a
+ spied upon channel hangs up and the option to exit the ChanSpy
+ application is set, removing the reference to the channel
+ allowing the count for the group that the spied channel was part
+ of to be decremented. (closes issue ASTERISK-17515) Reported by:
+ Arkadiusz Malka Tested by: Alexandr Gordeev, Michael L. Young
+ Patches: asterisk-17515-destroy-autochan.diff uploaded by Michael
+ L. Young (license 5026) ........ Merged revisions 370952 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370954 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-08 22:41 +0000 [r370951-370953] Mark Michelson <mmichelson@digium.com>
+
+ * CHANGES: Move a SIP change up to the other SIP changes in the
+ CHANGES file.
+
+ * main/channel.c, main/pbx.c, main/manager.c, pbx/pbx_spool.c,
+ apps/app_originate.c, include/asterisk/channel.h,
+ include/asterisk/pbx.h, CHANGES, res/res_clioriginate.c: Allow
+ support for early media on AMI originates and call files. This is
+ based on the work done by Olle Johansson on review board. The
+ idea is that the channel specified in an AMI originate or call
+ file is typically not connected to the outgoing extension until
+ the channel has been answered. With this change, an EarlyMedia
+ header can be specified for AMI originates and an early_media
+ option can be specified in call files. With this option set, once
+ early media is received on a channel, it will be connected with
+ the outgoing extension. (closes issue ASTERISK-18644) Reported by
+ Olle Johansson Review: https://reviewboard.asterisk.org/r/1472
+
+2012-08-08 21:22 +0000 [r370943] Terry Wilson <twilson@digium.com>
+
+ * main/manager.c, CHANGES: Add AMI_CLIENT dialplan function
+ Implementation of a dialplan function for checking manager
+ accounts. Right now it only returns the number of logged in
+ sessions for a manager account, but other attributes can be added
+ later. Patch by: Olle Johansson Review:
+ https://reviewboard.asterisk.org/r/421/
+
+2012-08-08 20:47 +0000 [r370927] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp_engine.c: Create the payload type if it does not exist
+ when setting information based on the 'm' line. An rtpmap
+ attribute is not required for defined payload numbers.
+
+2012-08-08 20:32 +0000 [r370926] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h: Convert sig_analog to use a global
+ callback table.
+
+2012-08-08 20:30 +0000 [r370925] Kinsey Moore <kmoore@digium.com>
+
+ * main/channel.c, /: Do not define a cause that doesn't actually
+ exist AST_CAUSE_NOTDEFINED is a placeholder for usage when there
+ is no cause information. As such, it should not be defined and
+ translatable as a cause. ........ Merged revisions 370923 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370924 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-08 20:17 +0000 [r370887-370902] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/sig_analog.h: Fix the analog dial *0 flash-hook of
+ bridged peer feature. The flash-hook the bridged peer feature now
+ correctly determines if the bridged peer is another chan_dahdi
+ channel, that it is an analog channel, and that it has the
+ correct signaling for an FXO port. It now also flash-hooks the
+ correct channel. ........ Merged revisions 370900 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370901 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
+ Convert sig_pri to use a global callback table.
+
+ * channels/chan_dahdi.c, channels/sig_ss7.h, channels/sig_ss7.c:
+ Convert sig_ss7 to use a global callback table.
+
+2012-08-07 21:58 +0000 [r370881] Damien Wedhorn <voip@facts.com.au>
+
+ * build_tools/cflags-devmode.xml, channels/chan_skinny.c: Rewrite
+ of skinny debugging. Debugging messages and associated controls
+ only compiled in if configured with --enable-dev-mode. Debug
+ messages provide more detail (including thread id) and are
+ grouped so the user/dev can limit the type of messages displayed.
+ Functionally no real change to chan_skinny. Review:
+ https://reviewboard.asterisk.org/r/2040/
+
+2012-08-07 19:59 +0000 [r370860] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp_engine.c, include/asterisk/rtp_engine.h: Payload and RTP
+ code are must remain separate since in non-Asterisk format cases
+ they differ.
+
+2012-08-07 19:26 +0000 [r370851-370859] Kinsey Moore <kmoore@digium.com>
+
+ * /: Recorded merge of revisions 370858 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Add
+ missing AST_CAUSE_* -> text translations ........ Merged
+ revisions 370856 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/channel.c: Add missing AST_CAUSE_* -> text translations A
+ few of these were missing from the list and are necessary for the
+ Who Hung Up? functionality.
+
+2012-08-07 17:47 +0000 [r370832-370845] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp_engine.c: Fix a bug uncovered by the test suite where
+ the RTP payload number was not getting set.
+
+ * res/res_rtp_asterisk.c, main/rtp_engine.c, channels/chan_sip.c,
+ channels/chan_motif.c, include/asterisk/rtp_engine.h: Reduce
+ memory consumption significantly for users of the RTP engine API
+ by storing only the payloads present and in use instead of every
+ possible one. Review: https://reviewboard.asterisk.org/r/2052/
+
+2012-08-07 12:46 +0000 [r370820-370831] Matthew Jordan <mjordan@digium.com>
+
+ * main/channel.c, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, channels/chan_misdn.c,
+ channels/chan_sip.c, main/channel_internal_api.c,
+ channels/misdn/chan_misdn_config.h, main/features.c,
+ configs/misdn.conf.sample, include/asterisk/channel.h,
+ configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h,
+ channels/misdn_config.c: Add named callgroups/pickupgroups This
+ patch adds named calledgroups/pickupgroups to Asterisk. Named
+ groups are implemented in parallel to the existing numbered
+ callgroup/pickupgroup implementation. However, unlike the
+ existing implementation, which is limited to a maximum of 64
+ defined groups, the number of defined groups allowed for named
+ callgroups/pickupgroups is effectively unlimited. Named groups
+ are configured with the keywords "namedcallgroup" and
+ "namedpickupgroup". This corresponds to the numbered group
+ definitions of "callgroup" and "pickupgroup". Note that as the
+ implementation of named groups coexists with the existing
+ numbered implementation, a defined named group of "4" does not
+ equate to numbered group 4. Support for the named groups has been
+ added to the SIP, DAHDI, and mISDN channel drivers. Review:
+ https://reviewboard.asterisk.org/r/2043 Uploaded by: Guenther
+ Kelleter(license #6372)
+
+ * contrib/realtime/mysql/voicemail_data.sql: Revert r370820 That
+ change is wrong, wrong, wrong.
+
+ * contrib/realtime/mysql/voicemail_data.sql: Update the MySQL
+ voicemail_data contrib script to reflect Asterisk 11 changes All
+ voicemails now have a 'msg_id' included in their metadata. The
+ ODBC message storage backend now requires this column; as such,
+ the MySQL contrib script that creates the voicemail_data table
+ has been updated with the appropriate column information.
+
+2012-08-06 15:18 +0000 [r370801] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Improve debug message for temporary
+ outbound proxies. Thanks to Paul Belanger for pointing this out.
+ ........ Merged revisions 370797 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370798 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-03 21:52 +0000 [r370773] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/config_parser.c,
+ channels/sip/include/sip.h: Multiple revisions 370769-370771
+ ........ r370769 | mmichelson | 2012-08-03 16:35:00 -0500 (Fri,
+ 03 Aug 2012) | 24 lines 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 ........
+ r370770 | mmichelson | 2012-08-03 16:39:35 -0500 (Fri, 03 Aug
+ 2012) | 3 lines Remove unused variable. ........ r370771 |
+ mmichelson | 2012-08-03 16:43:52 -0500 (Fri, 03 Aug 2012) | 5
+ lines Seriously? Another compilation error fixed. Somebody beat
+ me. ........ Merged revisions 370769-370771 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370772 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-08-02 15:51 +0000 [r370740] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c: Fix regression from r370636 When the
+ chan_sip cleanup went in, a typo was included that caused some
+ subscriptions of non-Polycom phones to be limited to the same
+ capabilities as Polycom phones. This resolves the failures in the
+ test suite resulting from this regression.
+
+2012-08-01 19:37 +0000 [r370726] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c: Fix a possible crash due to passing NULL to
+ ast_variables_dup()
+
+2012-08-01 18:52 +0000 [r370720] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/astobj2.h, main/astobj2.c: Make astobj2.h not
+ include linkedlists.h. Using astobj2 does not require
+ linkedlists.h be included even though astob2 uses linked lists
+ internally.
+
+2012-08-01 02:26 +0000 [r370699] Kinsey Moore <kmoore@digium.com>
+
+ * /, utils/extconf.c: Revert alloca changes for utils These changes
+ were a tad overzealous in the utils directory. Unfortunately,
+ these don't compile with a "make". ........ Merged revisions
+ 370697 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 370698 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-31 22:28 +0000 [r370681-370691] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
+ channels/sip/include/sip.h: Add headers from SIPAddHeader to
+ outbound REFER requests. This is a patch from kkm from review
+ board. This is useful for adding headers to REFER requests that
+ emanate from a Transfer() dialplan application call. This also
+ fixes some uses of the Referred-by header, removing an extra set
+ of angle brackets. I've modified the reporter's original patch to
+ not require any additions to the sip_refer header and to just
+ remove the referred_by_name from sip_refer since it is no longer
+ needed or used. (closes Issue ASTERISK-17639) reported by Kirill
+ Katsnelson Patches: 019059-sip-refer-addheaders-trunk-353549.diff
+ uploaded by Kirill Katsnelson (license #5845) Review:
+ https://reviewboard.asterisk.org/r/1159
+
+ * main/manager.c, configs/manager.conf.sample, CHANGES: Add
+ "setvar" option to manager.conf. With this option set, channel
+ variables can be set on every manager originate. The Variable
+ header can still be used to set additional channel variables for
+ individual calls if desired. This work was completed by Olle
+ Johansson on review board. I have applied the review feedback and
+ am committing it in order to get this into trunk before Asterisk
+ 11 is branched. Review: https://reviewboard.asterisk.org/r/1412
+
+2012-07-31 21:20 +0000 [r370677] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Schedule pokes of registered SIP peers
+ within a given timespan after SIP reload With a large number of
+ SIP peers registered, performing a SIP reload causes a flood of
+ SIP OPTIONS request packets. These are immediately sent out, and,
+ as responses come back, can cause peers to be flagged as 'lagged'
+ due to handling of the many response messages. This fix prevents
+ this "packet storm" and schedules the pokes for a random time.
+ That time varies between 1 ms and the peer's qualify time, or, if
+ the qualify time is unknown, the global qualifyfreq setting. The
+ committed patch has some very small modifications to the patch
+ schmidts wrote for the review. (closes issue ASTERISK-19154)
+ Reported by: Nicolo Mazzon patches: issue19154.patch license
+ #6034 uploaded by schmidts Review:
+ https://reviewboard.asterisk.org/r/1652 ........ Merged revisions
+ 370666 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 370672 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-31 20:33 +0000 [r370664] Russell Bryant <russell@russellbryant.com>
+
+ * main/event.c: Move event cache updates into event processing
+ thread. Prior to this patch, updating the device state cache was
+ done by the thread that originated the event. It would update the
+ cache and then queue the event up for another thread to dispatch.
+ This thread moves the cache updating part to be in the same
+ thread as event dispatching. I was working with someone on a
+ heavily loaded Asterisk system and while reviewing backtraces of
+ the system while it was having problems, I noticed that there
+ were a lot of threads contending for the lock on the event cache.
+ By simply moving this into a single thread, this helped
+ performance *a lot* and alleviated some deadlock-like symptoms.
+ Review: https://reviewboard.asterisk.org/r/2066/
+
+2012-07-31 20:21 +0000 [r370655] Kinsey Moore <kmoore@digium.com>
+
+ * /, main/say.c, main/threadstorage.c, funcs/func_strings.c,
+ channels/chan_iax2.c, main/config.c, channels/chan_dahdi.c,
+ pbx/pbx_spool.c, channels/sig_analog.c, main/strcompat.c,
+ main/features.c, pbx/pbx_ael.c, main/http.c, pbx/pbx_realtime.c,
+ channels/chan_alsa.c, channels/sig_ss7.c, main/db.c,
+ include/asterisk/utils.h, main/pbx.c, funcs/func_cut.c,
+ tests/test_linkedlists.c, funcs/func_channel.c, apps/app_macro.c,
+ apps/app_mixmonitor.c, main/asterisk.c, apps/app_voicemail.c,
+ addons/app_mysql.c, apps/app_meetme.c, apps/app_dictate.c,
+ main/utils.c, funcs/func_logic.c, cdr/cdr_pgsql.c,
+ channels/chan_gtalk.c, res/res_jabber.c,
+ res/res_http_websocket.c, res/ael/pval.c, main/channel.c,
+ main/manager.c, apps/app_osplookup.c, res/res_agi.c,
+ apps/app_minivm.c, main/logger.c, main/app.c,
+ addons/chan_mobile.c, apps/app_while.c, res/res_config_pgsql.c,
+ channels/chan_sip.c, apps/app_festival.c, pbx/pbx_lua.c,
+ channels/sig_pri.c, apps/app_getcpeid.c, funcs/func_global.c,
+ channels/chan_jingle.c, main/tcptls.c,
+ apps/app_directed_pickup.c, main/file.c, main/callerid.c,
+ apps/app_sms.c, main/astmm.c, main/event.c, pbx/pbx_dundi.c,
+ include/asterisk/strings.h, utils/extconf.c, main/dsp.c,
+ addons/res_config_mysql.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) ........ Merged revisions 370642 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370643 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-31 19:57 +0000 [r370644] Mark Michelson <mmichelson@digium.com>
+
+ * CHANGES, pbx/pbx_config.c: Add "dialplan remove context" and
+ modify "dialplan add include" From corruptor's review board
+ posting: "I've noticed that we can remove particular extension
+ from context with dialplan remove extension command but in order
+ to remove all extensions in the context we should delete them on
+ by one. I've created dialplan remove context command which uses
+ ast_context_destroy to destroy the whole context with all
+ extensions. I've created to functions for in pbx_config.c:
+ handle_cli_dialplan_remove_context which actually removes context
+ and complete_dialplan_remove_context which completes input. They
+ are based on other similar functions and pretty trivial but I can
+ be mistaken somewhere. "I've also modified dialplan add include
+ <context2> into <context1>. I've made it similar dialplan add
+ extension ... command. It creates <context1> if it doesn't exist
+ and I've also modified complete_dialplan_add_include and removed
+ check for existance of <context2> because we can include
+ non-existent context into another one. (I usually include empty
+ (non-existent) contexts in advance). Should we raise warning in
+ this case as it's raised while reading extensions.conf? "I use
+ those functions with AMI. I think manager commands should be
+ created in addition to those CLI commands." I've addressed the
+ latest comments on review board and have made some other coding
+ guidelines-related cleanup. I also have modified the CHANGES file
+ to mention these new commands. (closes issue ASTERISK-19292)
+ reported by Andrey Solovyev Patches: dialplan_add_include.patch
+ uploaded by Andrey Solovyev (license #5214)
+ dialplan_remove_context.patch uploaded by Andrey Solovyev
+ (license #5214) Review: https://reviewboard.asterisk.org/r/2042
+
+2012-07-31 19:10 +0000 [r370636] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c, channels/sip/security_events.c,
+ channels/sip/include/sip.h: Clean up chan_sip This clean up was
+ broken out from https://reviewboard.asterisk.org/r/1976/ and
+ addresses the following: - struct sip_refer converted to use the
+ stringfields API. - sip_{refer|notify}_allocate ->
+ sip_{notify|refer}_alloc to match other *alloc functions. -
+ Replace get_msg_text, get_msg_text2 and get_pidf_body -> No, not
+ get_pidf_msg_text_body3 but get_content, to match add_content. -
+ get_body doesn't get the request body, renamed to
+ get_content_line. - get_body_by_line doesn't get the body line,
+ and is just a simple if test. Moved code inline and removed
+ function. - Remove camelCase in struct sip_peer peer state
+ variables, onHold -> onhold, inUse -> inuse, inRinging ->
+ ringing. - Remove camelCase in struct sip_request rlPart1 ->
+ rlpart1, rlPart2 -> rlpart2. - Rename instances of pvt->randdata
+ to pvt->nonce because that is what it is, no need to update
+ struct sip_pvt because _it already has a nonce field_. - Removed
+ struct sip_pvt randdata stringfield. - Remove useless (and
+ inconsistent) 'header' suffix on variables in
+ handle_request_subscribe. - Use ast_strdupa on Event header in
+ handle_request_subscribe to avoid overly complicated strncmp
+ calls to find the event package. - Move get_destination check in
+ handle_request_subscribe to avoid duplicate checking for packages
+ that don't need it. - Move extension state callback management in
+ handle_request_subscribe to avoid duplicate checking for packages
+ that don't need it. - Remove duplicate append_date prototype. -
+ Rename append_date -> add_date to match other add_xxx functions.
+ - Added add_expires helper function, removed code that manually
+ added expires header. - Remove _header suffix on
+ add_diversion_header (no other header adding functions have
+ this). - Don't pass req->debug to request handle_request_XXXXX
+ handlers if req is also being passed. - Don't pass req->ignore to
+ check_auth as req is already being passed. - Don't create a
+ subscription in handle_request_subscribe if p->expiry == 0. -
+ Don't walk of the back of referred_by_name when splitting string
+ in get_refer_info - Remove duplicate check for no dialog in
+ handle_incoming when sipmethod == SIP_REFER, handle_request_refer
+ checks for that. Review: https://reviewboard.asterisk.org/r/1993/
+ Patch-by: gareth
+
+2012-07-30 23:26 +0000 [r370565-370598] Richard Mudgett <rmudgett@digium.com>
+
+ * main/test.c: Tweak unit test warning message.
+
+ * funcs/func_presencestate.c, main/test.c: Fix some presence-state
+ unit test typos.
+
+ * apps/app_confbridge.c: DECLINE to load confbridge if the config
+ fails to load.
+
+ * channels/chan_misdn.c, /: Release B channel allocation on error
+ path in chan_misdn. ........ Merged revisions 370563 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370564 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-30 14:52 +0000 [r370548] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_meetme.c: app_meetme: Change app_meetme support level
+ to extended from deprecated (closes issue ASTERISK-20134)
+ Reported by: Leif Madsen ........ Merged revisions 370547 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-30 13:45 +0000 [r370534-370541] Russell Bryant <russell@russellbryant.com>
+
+ * tests/test_event.c: Fix ast_event_new unit test. One of my recent
+ commits broke this test. The error was:
+ [test_event.c:event_new_test:214]: Events expected to be
+ identical have different size: 69 != 59 The difference in size
+ occurred because the first event had the EID IE added to the
+ event twice. ast_event_new() now always adds it automatically.
+ Previously it only added it if there were no IEs specified, which
+ was kind of weird.
+
+ * include/asterisk/event_defs.h, res/res_corosync.c, main/event.c:
+ Add a "corosync ping" CLI command. This patch adds a new CLI
+ command to the res_corosync module. It is primarily used as a
+ debugging tool. It lets you fire off an event which will cause
+ res_corosync on other nodes in the cluster to place messages into
+ the logger if everything is working ok. It verifies that the
+ corosync communication is working as expected. I didn't put
+ anything in the CHANGES file for this, because this module is new
+ in Asterisk 11. There is already a generic "res_corosync new
+ module" entry in there so I figure that covers it just fine.
+
+ * addons/app_mysql.c, CHANGES: Allow specifying a port number for
+ the MySQL server. This patch allows you to specify a port number
+ for the MySQL server. It's useful if a MySQL server is running on
+ a non-standard port. Even though this module is deprecated in
+ favor of func_odbc, someone asked for this feature and it seems
+ pretty harmless to add. It has been tested using a number of
+ combinations of with/without a port number specified in the
+ dialplan and changing the port number for mysqld.
+
+2012-07-26 15:31 +0000 [r370510-370518] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c, CHANGES: chan_sip: Add SIPpeerstatus command
+ to AMI This patch was submitted by mnicholson a while back. It
+ adds a new AMI action which allows users to request SIP peer
+ status on demand similar to existing PeerStatus events and to the
+ output you would see from CLI with sip show peer Review:
+ https://reviewboard.asterisk.org/r/1098/
+
+ * /, 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 ........ Merged revisions 370494 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370495 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-25 14:27 +0000 [r370481-370488] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/Makefile: Repair editline builds using in-tree editline
+ sources. The previous change to the build system for using a
+ system-provided editline library was missing a crucial include
+ directory for building against the copy of the library in the
+ Asterisk source tree.
+
+ * main/Makefile: Use an absolute path when referring to the
+ embedded editline directory. This patch changes the build system
+ to refer to the embedded editline directory using an absolute
+ path, which will resolve a problem seen on the CentOS automated
+ build agents.
+
+ * build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, main/Makefile,
+ main/editline/configure, configure.ac, main/editline/readline
+ (removed), main/editline/readline.c, main/editline/configure.in,
+ CHANGES, makeopts.in, main/editline/readline.h (added),
+ main/asterisk.c, contrib/scripts/install_prereq, main/cli.c:
+ Enable usage of system-provided NetBSD editline library if
+ available. This patch changes the Asterisk configure script and
+ build system to detect the presence of the NetBSD editline
+ library (libedit) on the system. If it is found, it will be used
+ in preference to the version included in the Asterisk source
+ tree. (closes issue ASTERISK-18725) Reported by: Jeffrey C. Ollie
+ Review: https://reviewboard.asterisk.org/r/1528/ Patches:
+ 0001-Allow-linking-building-against-an-external-editline.patch
+ uploaded by jcollie (license #5373) (heavily modified by
+ kpfleming)
+
+2012-07-25 03:51 +0000 [r370474] Terry Wilson <twilson@digium.com>
+
+ * main/pbx.c, /: Revert a change that broke compilation 1) There is
+ no such function as ast_ref() 2) The patch was originally
+ credited as the one uploaded by Guenther Kelleter (license 6372)
+ via issue AST-921, but the patch committed was not the patch
+ referenced on the issue. 3) Guenther Kelleter's patch was
+ actually correct. It moved the ast_free above the
+ presencechange_cleanup label. I am not committing his change as
+ it is not technically necesary--calling ast_free(NULL) is
+ perfectly safe and I worry that moving the ast_free outside of
+ the label could lead to future bugs if someone ever adds another
+ failure conditional and expects 'goto presencechange_cleanup;' to
+ clean up after everything.
+
+2012-07-24 21:30 +0000 [r370466] Jonathan Rose <jrose@digium.com>
+
+ * main/pbx.c, /: Don't attempt free of NULL ptr in pbx.c
+ handle_presencechange (closes issue AST-921) Reported by:
+ Guenther Kelleter Patches: nullptr.patch uploaded by Guenther
+ Kelleter (license 6372)
+
+2012-07-24 19:12 +0000 [r370453] Kevin P. Fleming <kpfleming@digium.com>
+
+ * tests/test_acl.c: Silence a warning message from older versions
+ of GCC. Revision 370426 introduced the use of a nested function
+ in tests/test_acl.c, but the lack of the 'auto' scope specifier
+ on the function and a forward declaration resulted in compilation
+ errors on the automated test systems.
+
+2012-07-24 17:16 +0000 [r370433] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * /, channels/chan_oss.c: chan_oss: fix "sample rate" error message
+ Merged revisions 370428 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 Merged
+ revisions 370432 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-24 16:54 +0000 [r370426-370431] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/frame.c, /: Rewrite a comment that didn't adequately explain
+ the code it was documenting. ........ Merged revisions 370429
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 370430 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * CHANGES: Update CHANGES for list/negation ACL feature.
+
+ * tests/test_acl.c, main/acl.c: Allow permit/deny ACL lines to
+ contain multiple items and negated entries. Rules in ACLs
+ (specified using 'permit' and 'deny') can now contain multiple
+ items (separated by commas), and items in the rule can be negated
+ by prefixing them with '!'. This simplifies Asterisk Realtime
+ configurations, since it is no longer necessray to control the
+ order that the 'permit' and 'deny' columns are returned from
+ queries. Review: https://reviewboard.asterisk.org/r/1592/ Initial
+ patch contributed by Tilghman Lesher Unit tests written by Kevin
+ P. Fleming
+
+2012-07-24 16:15 +0000 [r370419-370420] Joshua Colp <jcolp@digium.com>
+
+ * res/res_rtp_asterisk.c: Build is underway so logging can go away.
+
+ * res/res_rtp_asterisk.c: Temporarily enable pj logging to console
+ for debugging pjnath issue exposed by build slave.
+
+2012-07-24 08:53 +0000 [r370413] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * channels/chan_unistim.c: Remove code, that operate with cdr in
+ attempt_transfer(). That was removed somewhere between 1.2 and
+ 1.4 and acidentaly put back in chan_unistim. (closes issue
+ ASTERISK-19628) Reported by: Igor Olhovskiy
+
+2012-07-23 21:27 +0000 [r370407] Kevin P. Fleming <kpfleming@digium.com>
+
+ * codecs/Makefile, build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ codecs/codec_ilbc.c, CHANGES, makeopts.in: Enable usage of
+ system-provided iLBC library. The WebRTC version of the iLBC
+ codec is now package as a library and is available on some
+ platforms. This patch allows codec_ilbc to be built against that
+ library if it is present. Review:
+ https://reviewboard.asterisk.org/r/1964/
+
+2012-07-23 21:15 +0000 [r370387] Matthew Jordan <mjordan@digium.com>
+
+ * tests/test_abstract_jb.c (added), main/abstract_jb.c,
+ funcs/func_jitterbuffer.c, include/asterisk/abstract_jb.h: Unit
+ tests for the Jitter Buffer API; remove unnecessary resync This
+ patch includes the following: * Unit tests for the abstract
+ Jitter Buffer API. This includes both fixed and adaptive flavors,
+ testing nominal creation, frame input, frame retrieval,
+ resyncing; off nominal frame input overflow, out of order, and
+ others. * Tweaks to the abstract_jb API to remove the unnecessary
+ resync_threshold parameter from the create function
+ (resync_threshold is already in the struct passed into the create
+ function) * Ensure the fixed jitter buffer is empty before
+ destroying it, to avoid an ASSERT * Don't "resync" the adaptive
+ jitter buffer. The mechanism that was being used actually causes
+ the jitter buffer to think its being overflowed by going around
+ the jitterbuf API and attempting to 'resynch' it improperly. If a
+ resync is needed, the jitter buffer will do it properly by
+ itself. Note that this is only an optimization needed for trunk,
+ as the worst that happens is the loss of three voice packets
+ before the adaptive jitter buffer will resync anyway. Review:
+ https://reviewboard.asterisk.org/r/2035
+
+2012-07-23 21:10 +0000 [r370386] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add
+ separate configuration options for subscription and registration
+ minexpiry and maxexpiry. This offers more fine-grained control
+ over how long subscriptions last without negatively affecting the
+ expiration range for registrations. Uploaded by: Guenther
+ Kelleter(license #6372) Review:
+ https://reviewboard.asterisk.org/r/2051
+
+2012-07-23 21:10 +0000 [r370385] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, funcs/func_shell.c: Improve documentation for the SHELL()
+ dialplan function. ........ Merged revisions 370383 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370384 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-23 21:02 +0000 [r370382] Mark Michelson <mmichelson@digium.com>
+
+ * UPGRADE.txt: Add notes to UPGRADE.txt about addition of msg_id to
+ VoiceMails.
+
+2012-07-23 00:15 +0000 [r370354] Joshua Colp <jcolp@digium.com>
+
+ * UPGRADE.txt: Update UPGRADE.txt with notes about ICE support and
+ res_xmpp.
+
+2012-07-22 23:37 +0000 [r370353] Matthew Jordan <mjordan@digium.com>
+
+ * CHANGES: Update CHANGES for Asterisk 11 This updates the CHANGES
+ file with things that were committed for Asterisk 11, but were
+ not noted in that file.
+
+2012-07-22 17:03 +0000 [r370347] Joshua Colp <jcolp@digium.com>
+
+ * res/res_rtp_asterisk.c, channels/chan_sip.c,
+ configs/sip.conf.sample, channels/sip/include/sip.h: Prevent
+ multiple local candidates from being added with the same
+ information and add support for disabling ICE on a per-peer
+ basis. (closes issue ASTERISK-20088) Reported by: wimpy Review:
+ https://reviewboard.asterisk.org/r/2044/
+
+2012-07-21 13:25 +0000 [r370341] Terry Wilson <twilson@digium.com>
+
+ * main/config_options.c, apps/app_confbridge.c,
+ apps/confbridge/conf_config_parser.c: Fix segfault introduced by
+ conversion to ACO API The value "none" is specified in the config
+ file as a valid value for the "video_mode" option. The code prior
+ to the ACO conversion did not check for "none", but just ignored
+ it and relied on the default zero value. The parsing with ACO is
+ more strict, so without handling "none" specifically, parsing
+ would fail. When parsing failed, but the module loaded anyway,
+ the config info would never be stored, and one place in the code
+ did not check for this case and would segfault. It was also
+ possible that the aco_info struct's internals would be destroyed
+ and used as well. This patch keeps the module from loading after
+ parse failures, adds the "none" option to "video_mode", registers
+ CLI functions only after parsing has completed, checks the config
+ data for NULL before accessing it, and returns -1 on some
+ allocation failures when initializing. (closes issue
+ ASTERISK-20159) Reported by: Birger "WIMPy" Harzenetter Tested
+ by: Birger "WIMPy" Harzenetter Patches: confbridge_fix3.txt
+ uploaded by Terry Wilson
+
+2012-07-20 19:36 +0000 [r370335] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_iax2.c: chan_iax2: Fix a segfault introduced by
+ call ID logging Didn't previously check that a non NULL IAX
+ channel was stored in the array at the requested position before
+ attempting iax_pvt_callid_get (closes issue ASTERISK-20145)
+ Reported by: Birger "WIMPy" Harzenetter
+
+2012-07-20 19:08 +0000 [r370329] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_dial.c: Clean up ManagerEvent Dial documentation The
+ paragraph describing the SubEvent belongs with the SubEvent
+ parameter itself, and not with its enum values. The order of
+ parsing was placing the description after the last enum, which
+ isn't correct.
+
+2012-07-20 18:37 +0000 [r370328] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_misdn.c: Fix build error in chan_misdn from commit
+ 370316 chan_misdn was not updated properly to account for a
+ change in parameters for HANGUPCAUSE functionality. It now builds
+ properly.
+
+2012-07-20 16:25 +0000 [r370322] Joshua Colp <jcolp@digium.com>
+
+ * res/res_http_websocket.exports.in: Export the
+ ast_websocket_set_nonblock function for use by other modules.
+
+2012-07-20 15:48 +0000 [r370316] Kinsey Moore <kmoore@digium.com>
+
+ * funcs/func_hangupcause.c (added), main/channel.c,
+ channels/chan_dahdi.c, channels/sig_analog.c, main/rtp_engine.c,
+ channels/chan_sip.c, main/channel_internal_api.c, UPGRADE.txt,
+ include/asterisk/channel.h, channels/chan_iax2.c,
+ channels/sig_pri.c, include/asterisk/frame.h, channels/sig_ss7.c:
+ Add hangupcause translation support The HANGUPCAUSE hash (trunk
+ only) meant to replace SIP_CAUSE has now been replaced with the
+ HANGUPCAUSE and HANGUPCAUSE_KEYS dialplan functions to better
+ facilitate access to the AST_CAUSE translations for
+ technology-specific cause codes. The HangupCauseClear application
+ has also been added to remove this data from the channel. (closes
+ issue SWP-4738) Review: https://reviewboard.asterisk.org/r/2025/
+
+2012-07-20 15:40 +0000 [r370309-370315] Richard Mudgett <rmudgett@digium.com>
+
+ * CHANGES: Update CHANGES about adding the AccountCode header to
+ the AMI Hangup event. (issue ASTERISK-19963)
+
+ * main/channel.c: Add the AccountCode header to the AMI Hangup
+ event. It's harder to correlate the Newchannel and Hangup AMI
+ events without specifying "AccountCode" in both. (closes issue
+ ASTERISK-19963) Reported by: Oleg A. Arkhangelsky Patches:
+ hangup_acctcode.diff (license #6397) patch uploaded by Oleg A.
+ Arkhangelsky
+
+2012-07-19 23:21 +0000 [r370303] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/config_options.h,
+ apps/confbridge/include/confbridge.h, main/config_options.c,
+ apps/confbridge/conf_config_parser.c: Convert app_confbridge to
+ use the config options framework Review:
+ https://reviewboard.asterisk.org/r/2024/
+
+2012-07-19 22:25 +0000 [r370298] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/cel.c: Fix compiler warnings. gcc (GCC) 4.2.4 has
+ problems casting away constness. ........ Merged revisions 370275
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 370277 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-19 22:17 +0000 [r370272-370278] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_sip.c, res/res_xmpp.c, doc/appdocsxml.dtd,
+ main/message.c, main/xmldoc.c: Add the ability to specify
+ technology specific documentation A number of applications/AMI
+ commands in Asterisk have specific behavioral differences
+ depending on the resource or channel technology those
+ applications are executed on. For example, the MessageSend
+ application/ command is technology agnostic, but how the channel
+ drivers that support that functionality behave is dependant on
+ the protocols and channel driver implementation. Prior to this
+ patch, those details were either documented in the
+ application/command documentation itself, or were left
+ undocumented. This patch adds a new element to the documentation
+ schema, <info/>. An info node is essentially a piece of
+ technology specific reference information that can be included by
+ any top level XML documentation node. For example, the
+ MessageSend application can now include XMPP/SIP specific
+ information, where that technology specific information can be
+ defined in chan_motif/res_xmpp/ chan_sip. Likewise, that
+ information can also be included in the MessageSend AMI command.
+ Review: https://reviewboard.asterisk.org/r/2049
+
+ * /, 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 ........ Merged revisions 370273 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370274 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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/ (closes issue
+ ASTERISK-18404) Reported by: Stephane Chazelas Tested by: Matt
+ Jordan ........ Merged revisions 370252 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370271 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-19 20:37 +0000 [r370246-370265] Jonathan Rose <jrose@digium.com>
+
+ * main/named_acl.c, configs/acl.conf.sample: named_acl: Remove
+ systemname option from acl.conf, use asterisk.conf value Review:
+ https://reviewboard.asterisk.org/r/2057/
+
+ * main/channel_internal_api.c: CallID Logging: Remove new
+ line/carriage return from callID change test event
+
+2012-07-19 12:14 +0000 [r370234-370240] Joshua Colp <jcolp@digium.com>
+
+ * res/Makefile, res/pjproject/build/os-auto.mak.in: Use the
+ bruteforce method to get debugging enabled for pjproject.
+
+ * res/Makefile: Turn on debugging for pjproject so we can get a
+ better idea of what is causing the generic CCSS test crash.
+
+2012-07-18 19:48 +0000 [r370225] Jonathan Rose <jrose@digium.com>
+
+ * main/channel_internal_api.c: callid logging: Issue test events
+ when the callid is changed for a channel Review:
+ https://reviewboard.asterisk.org/r/2054/
+
+2012-07-18 19:18 +0000 [r370187-370211] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, main/cel.c: Resolve severe memory leak in CEL logging modules.
+ A customer reported a significant memory leak using Asterisk 1.8.
+ They have tracked it down to
+ ast_cel_fabricate_channel_from_event() in main/cel.c, which is
+ called by both in-tree CEL logging modules (cel_custom.c and
+ cel_sqlite3_custom.c) for each and every CEL event that they log.
+ The cause was an incorrect assumption about how data attached to
+ an ast_channel would be handled when the channel is destroyed;
+ the data is now stored in a datastore attached to the channel,
+ which is destroyed along with the channel at the proper time.
+ (closes issue AST-916) Reported by: Thomas Arimont Review:
+ https://reviewboard.asterisk.org/r/2053/ ........ Merged
+ revisions 370205 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370206 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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, apps/app_macro.c, channels/chan_iax2.c,
+ apps/app_mixmonitor.c, apps/app_stack.c, funcs/func_global.c,
+ res/res_odbc.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. ........ Merged revisions 370183 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370184 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-18 15:15 +0000 [r370171-370177] Joshua Colp <jcolp@digium.com>
+
+ * res/res_rtp_asterisk.c: Fix a crash in pjnath when starting an
+ ICE connectivity check and immediately destroying the ICE
+ session. The initial ICE connectivity check is scheduled as a
+ timer item that is to be executed immediately. It is possible for
+ this timer item to start executing while the ICE session it is
+ working on is destroyed. To reduce the chance of this any timer
+ items that need to be immediately executed will be executed
+ within the thread that has started the initial ICE connectivity
+ check.
+
+ * channels/chan_sip.c, include/asterisk/rtp_engine.h: Fix a crash
+ occurring as a result of excess stack usage. This fix involves
+ moving the allocation of some temporary codec structures to the
+ heap and also reduces the number of maximum payloads to something
+ more sane for both regular and low memory builds. (closes issue
+ ASTERISK-20140) Reported by: jonnt
+
+2012-07-18 07:17 +0000 [r370165] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * channels/chan_unistim.c, configs/unistim.conf.sample, CHANGES:
+ Added option 'interdigit_timer' to unistim.conf to make able
+ controll hardcoded dial timeout constant.
+
+2012-07-17 19:05 +0000 [r370152-370157] Joshua Colp <jcolp@digium.com>
+
+ * res/res_xmpp.c: Add pubsub unsubscription support so
+ subscriptions do not linger for MWI and device state progatation.
+ The pubsub code did not attempt to remove subscriptions at all.
+ This has now changed so that if a client is being disconnected it
+ will unsubscribe. It will also unsubscribe at connection time so
+ if it unexpectedly disconnected duplicate subscriptions will not
+ occur. (closes issue ASTERISK-19882) Reported by: mattvryan
+
+ * include/asterisk/xmpp.h, res/res_xmpp.c: Fix a crash as a result
+ of propagating MWI or device state over XMPP when the client is
+ disconnected. The MWI and device state propagation code wrongly
+ assumes that an XMPP client connection will remain established at
+ all times. This fix corrects that by making the lifetime of the
+ subscription the same as the lifetime of the connection itself.
+ As the connection is established and disconnected the
+ subscription itself is created and destroyed. (closes issue
+ ASTERISK-18078) Reported by: elguero
+
+2012-07-16 19:58 +0000 [r370133] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, channels/chan_sip.c: Code cleanup and bugfix in chan_sip
+ outboundproxy parsing. The bug was clearing the global
+ outboundproxy when a peer-specific outboundproxy was bad. The
+ cleanup reduces duplicate code. Review:
+ https://reviewboard.asterisk.org/r/2034/ Reviewed by: Mark
+ Michelson ........ Merged revisions 370131 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370132 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-16 19:14 +0000 [r370111-370126] Joshua Colp <jcolp@digium.com>
+
+ * res/res_xmpp.c: Fix an issue where a service discovery request
+ could crash Asterisk. A server sending a service discovery
+ request to us may or may not put a from attribute in the message.
+ If the from attribute is present use it in the to attribute for
+ the result. If the from attribute is not present do not add a to
+ attribute. (issue ASTERISK-16203) Reported by: wubbla
+
+ * res/res_xmpp.c: Fix a bug where some XMPP servers would reject
+ authentication. We need to use the user portion of the JID and
+ not the full configured username.
+
+ * res/res_xmpp.c: Add missing namespace for old non-SASL based
+ authentication.
+
+ * channels/chan_sip.c: Fix a bug exposed by the testsuite where
+ text streams would no longer be parsed correctly.
+
+2012-07-16 14:02 +0000 [r370083] Kinsey Moore <kmoore@digium.com>
+
+ * /, UPGRADE-10.txt, CHANGES, UPGRADE-1.8.txt: Add comments about
+ the BUILD_NATIVE change This is a significant change and mention
+ of it should have gone into UPGRADE.txt and CHANGES. ........
+ Merged revisions 370081 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370082 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-16 12:58 +0000 [r370072-370073] Joshua Colp <jcolp@digium.com>
+
+ * res/res_xmpp.c: Fix an issue where specifying the resource in the
+ username would cause authentication to fail.
+
+ * channels/sip/sdp_crypto.c, channels/chan_sip.c,
+ channels/sip/security_events.c,
+ include/asterisk/http_websocket.h, configs/sip.conf.sample,
+ CHANGES, res/res_http_websocket.c, channels/sip/include/sip.h:
+ Add support for SIP over WebSocket. This allows SIP traffic to be
+ exchanged over a WebSocket connection which is useful for rtcweb.
+ Review: https://reviewboard.asterisk.org/r/2008
+
+2012-07-16 07:38 +0000 [r370066-370067] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * channels/chan_unistim.c: Deactivate timer for dialing entered
+ number on hook switch hang up. (closes issue ASTERISK-19554)
+ Reported by: Stefano Villani
+
+ * channels/chan_unistim.c, contrib/unistimLang/fr.po (added),
+ CHANGES: Add French translation for chan_unistim phones on-screen
+ menus.
+
+2012-07-13 18:41 +0000 [r370055-370060] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/format.h, res/res_format_attr_h263.c (added),
+ res/res_format_attr_h264.c (added): Reduce memory consumption and
+ add the H.264 and H.263 modules I shamefully neglected to add.
+
+ * main/format.c, channels/chan_sip.c, main/translate.c,
+ include/asterisk/format.h, res/res_format_attr_silk.c,
+ res/res_format_attr_celt.c: Add support for parsing SDP
+ attributes, generating SDP attributes, and passing it through.
+ This support includes codecs such as H.263, H.264, SILK, and
+ CELT. You are able to set up a call and have attribute
+ information pass. This should help considerably with video calls.
+ Review: https://reviewboard.asterisk.org/r/2005/
+
+2012-07-13 00:05 +0000 [r370048] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * contrib/scripts/live_ast: live_ast: don't set working directory
+ contrib/scripts/live_ast currently assumes that it is being run
+ from the top-level directory of the source tree. It creates a
+ script that will also set the working directory. This fix avoids
+ the need to set the working directory if the caller sets
+ LIVE_AST_BASE_DIR instead. It relies on realpath for that. If
+ realpath is not available, it will fall back to the original
+ behaviour. Review: https://reviewboard.asterisk.org/r/2027/
+
+2012-07-12 21:43 +0000 [r370043] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/config_options.h,
+ configs/config_test.conf.sample, main/config_options.c,
+ tests/test_config.c: Handle deprecated (aliased) option names
+ with the config options api Add a simple way to register
+ "deprecated" option names that alias to a different "current"
+ name. Review: https://reviewboard.asterisk.org/r/2026/
+
+2012-07-12 20:28 +0000 [r370037] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: Add missing
+ ast_hangup() calls on some analog exception paths. Make starting
+ analog_ss_thread() or __analog_ss_thread() failure paths hangup
+ the channel. ........ Merged revisions 370017 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370025 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-12 20:06 +0000 [r369995-370016] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Include Expires header for SIP PUBLISH
+ requests RFC3903 requres SIP PUBLISH requests to have Expires
+ headers, so add them. Review:
+ https://reviewboard.asterisk.org/r/2003/ Patch-by: gareth
+ ........ Merged revisions 370014 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 370015 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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 ........ Merged revisions 369993 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369994 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-12 14:38 +0000 [r369972-369974] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, funcs/func_math.c: Correct Documentation For DEC Function The
+ documentation for DEC in func_math.c was incorrect. Looks like a
+ copy and paste error. (Closes issue ASTERISK-20095) Reported by:
+ Billy Chia Tested by: Michael L. Young Patches: func_math.patch
+ uploaded by Billy Chia (license 6381) ........ Merged revisions
+ 369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 369971 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * funcs/func_math.c: Reverting last merge since it wasn't completed
+ properly.
+
+ * 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) ........ Merged revisions
+ 369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-07-11 18:33 +0000 [r369959] Jonathan Rose <jrose@digium.com>
+
+ * include/asterisk/acl.h, channels/chan_sip.c,
+ include/asterisk/config.h, main/acl.c,
+ include/asterisk/channel.h, configs/manager.conf.sample,
+ channels/chan_iax2.c, CHANGES, main/named_acl.c (added),
+ main/config.c, main/loader.c, configs/iax.conf.sample,
+ main/manager.c, include/asterisk/event_defs.h,
+ configs/extconfig.conf.sample, configs/sip.conf.sample,
+ channels/sip/include/sip.h, main/asterisk.c,
+ configs/acl.conf.sample (added): Named ACLs: Introduces a system
+ for creating and sharing ACLs This patch adds Named ACL
+ functionality to Asterisk. This allows system administrators to
+ define an ACL and refer to it by a unique name. Configurable
+ items can then refer to that name when specifying access control
+ lists. It also includes updates to all core supported consumers
+ of ACLs. That includes manager, chan_sip, and chan_iax2. This
+ feature is based on the deluxepine-trunk by Olle E. Johansson and
+ provides a subset of the Named ACL functionality implemented in
+ that branch. For more information on this feature, see acl.conf
+ and/or the Asterisk wiki. Review:
+ https://reviewboard.asterisk.org/r/1978/
+
+2012-07-11 17:16 +0000 [r369940] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, main/ast_expr2.h, main/ast_expr2f.c, res/ael/ael_lex.c,
+ funcs/func_realtime.c, main/ast_expr2.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/ ........ Merged
+ revisions 369937 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369938 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-11 17:14 +0000 [r369939] Matthew Jordan <mjordan@digium.com>
+
+ * main/features.c: Don't perform an XInclude to a document node
+ that may not always be present Because some of the manager events
+ are defined in the top of the source, due to the macro calls not
+ containing all necessary information to have the documentation
+ colocated with the call itself, several include statements were
+ failing when built with 'make'. While this did not cause any
+ problems in compilation or validation, it did result in a number
+ of warnings being dumped to stderr. This patch changes those
+ references such that they always resolve, regardless of the
+ documentation build options.
+
+2012-07-11 16:42 +0000 [r369936] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_motif.c: Do not consider failure to read the
+ configuration file in chan_motif to be a show stopper for loading
+ Asterisk by returning decline instead of failure. (closes issue
+ ASTERISK-20103) Reported by: Terry Wilson
+
+2012-07-11 02:06 +0000 [r369905-369910] Matthew Jordan <mjordan@digium.com>
+
+ * main/cdr.c, main/channel.c, channels/sig_analog.c, main/logger.c,
+ channels/sig_pri.c, main/asterisk.c, main/loader.c: Fix
+ validation errors when producing documentation using default
+ build script The awk script parses out the first instance of the
+ DOCUMENTATION tag that it finds within a file. If a file did not
+ previously have a DOCUMENTATION tag but received one due to it
+ having an AMI event, then the XML fragment associated with the
+ AMI event was erroneously placed in the resulting XML file.
+ Without the python scripts, these XML fragments will not
+ validate. This patch adds DOCUMENTATION tags at the top of those
+ files that did not previously have them to prevent the awk script
+ from pulling AMI event documentation.
+
+ * main/cdr.c, main/channel.c, channels/chan_dahdi.c, main/pbx.c,
+ channels/chan_local.c, channels/sig_analog.c, main/manager.c,
+ channels/chan_agent.c, main/features.c, main/logger.c,
+ channels/sig_pri.c, doc/appdocsxml.dtd, main/asterisk.c,
+ main/loader.c: Add some additional documentation for core AMI
+ events This patch adds some basic documentation for a number of
+ modules. This includes core source files in Asterisk (those in
+ main), as well as chan_agent, chan_dahdi, chan_local, sig_analog,
+ and sig_pri. The DTD has also been updated to allow referencing
+ of AMI commands.
+
+2012-07-10 15:36 +0000 [r369900] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c: Fix failing SDP_offer_answer test Asterisk
+ now generates image stream declinations with the same transport
+ case that it used to before the stream declination improvements.
+ (udptl vs UDPTL) (closes issue SWP-4736)
+
+2012-07-10 15:25 +0000 [r369873-369898] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_motif.c: Add additional description stanza names
+ from the old Google Talk protocol which is used with Google
+ Voice. (closes issue ASTERISK-20114) Reported by: Malcolm
+ Davenport
+
+ * channels/chan_motif.c: Respect codec preference order when adding
+ codecs to a media description. This change allows an endpoint in
+ motif.conf to be configured with a preference of G.722 and
+ fallback of ulaw. With Google this allows communication with
+ Google Talk clients to use G.722 while when using Google Voice
+ ulaw will be used. (closes issue ASTERISK-20114) Reported by:
+ Malcolm Davenport
+
+2012-07-10 13:40 +0000 [r369872] Kinsey Moore <kmoore@digium.com>
+
+ * main/pbx.c, /, apps/app_stack.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 ........ Merged
+ revisions 369869 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369871 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-10 13:34 +0000 [r369870] Matthew Jordan <mjordan@digium.com>
+
+ * main/libasteriskssl.c: Fix initial loading problem with res_curl
+ When the OpenSSL duplicate initialization issues were resolved in
+ r351447, res_curl could fail to load if it checked
+ SSL_library_init after SSL initialization completed. This is due
+ to the SSL_library_init stub returning a value of 0 for success,
+ as opposed to a value of 1. OpenSSL uses a value of 1 to indicate
+ success - in fact, SSL_library_init is documented to always
+ return 1. Interestingly, the CURL libraries actually checked the
+ return value - the fact that nothing else that depends on OpenSSL
+ was having problems loading probably means they don't check the
+ return value. (closes issue AST-924) Reported by: Guenther
+ Kelleter patches: (AST-924.patch license #6372 uploaded by
+ Guenther Kelleter)
+
+2012-07-10 11:49 +0000 [r369837-369864] Joshua Colp <jcolp@digium.com>
+
+ * res/res_rtp_asterisk.c, channels/chan_motif.c: Add required items
+ for Google video support. This adds legacy STUN support for RTCP
+ sockets, adds RTCP candidates to the Google transport
+ information, and adds required codec parameters. (closes issue
+ ASTERISK-20106) Reported by: Malcolm Davenport
+
+ * main/stun.c: When receiving a STUN binding request send one out
+ as the Google Talk client uses this as a method to determine if
+ the remote party is still reachable or not. Failure to do this
+ results in the Google Talk client ignoring RTP packets after a
+ specific period of time. This is also done as a result of
+ receiving a STUN binding request so that the username information
+ can be used from the inbound request, thus not requiring it to be
+ stored on a per candidate basis. (closes issue ASTERISK-20107)
+ Reported by: Malcolm Davenport
+
+ * channels/chan_sip.c: Add support for exposing the received
+ contact URI and also for setting the request URI in messages.
+ (closes issue AST-911)
+
+ * channels/chan_motif.c: Force the clock rate of G.722 to be 16000
+ when using the Google transports as it is 8000 elsewhere. (closes
+ issue ASTERISK-20105) Reported by: Malcolm Davenport
+
+ * configs/motif.conf.sample: Document that multiple endpoints using
+ the same connection is not supported. (closes issue
+ ASTERISK-20104) Reported by: Malcolm Davenport
+
+2012-07-09 17:07 +0000 [r369820] Jason Parker <jparker@digium.com>
+
+ * configs/sip_notify.conf.sample, /: Add Digium phones context to
+ sip_notify sample config. This makes it so that they can be
+ reconfigured remotely. (closes issue ASTERISK-19910) ........
+ Merged revisions 369818 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369819 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-09 16:44 +0000 [r369811-369817] Joshua Colp <jcolp@digium.com>
+
+ * res/res_rtp_asterisk.c: Fix an issue where media would not flow
+ for situations where the legacy STUN code is in use. The STUN
+ packets should *not* be blocked by strict RTP. (closes issue
+ ASTERISK-20102) Reported by: Malcolm Davenport
+
+ * res/res_xmpp.c: Add additional namespaces for Google Talk which
+ are used for the gmail client. (closes issue ASTERISK-20101)
+ Reported by: Malcolm Davenport
+
+ * channels/chan_motif.c: Fix dependency to be on res_xmpp. Long ago
+ in a galaxy far far away it used to use res_jabber.
+
+2012-07-09 14:54 +0000 [r369794] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Fix small behavioral change
+ accidentally introduced in r369750 When removing the warning for
+ AST_CONTROL_FLASH from sip_indicate, I also inadvertently changed
+ the return value, which would likely make the indication not be
+ sent in audio. This fixes that while still removing the warning
+ message. ........ Merged revisions 369792 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369793 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-07 17:06 +0000 [r369769] Joshua Colp <jcolp@digium.com>
+
+ * res/res_xmpp.exports.in (added), include/asterisk/xmpp.h,
+ channels/chan_motif.c (added), UPGRADE.txt,
+ channels/chan_gtalk.c, res/res_xmpp.c, CHANGES, res/res_jabber.c,
+ configs/motif.conf.sample (added): Add a new unified Jingle,
+ Google Jingle, and Google Talk channel driver written from
+ scratch called chan_motif. This channel driver is a replacement
+ for both chan_gtalk and chan_jingle but adds additional features
+ not found in either. These features include full configuration
+ reload, video, full codec support, bidirectional cause code
+ mapping, hold, unhold, and ringing indication. It is also
+ compliant with the current published Jingle and Google Jingle
+ specifications. The original Google Talk protocol is also
+ supported for Google Voice interoperability. You may ask yourself
+ though where the name motif comes from... and I would say to
+ you... music! motif: a perceivable or salient recurring fragment
+ or succession of notes Sorta like a jingle! Review:
+ https://reviewboard.asterisk.org/r/1917/
+
+2012-07-06 22:03 +0000 [r369765] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/chan_iax2.c, channels/sig_pri.c, channels/sig_ss7.c:
+ Remove unnecessary generation of informational cause frames It is
+ not necessary to generate information cause code frames on every
+ protocol event that occurs. This removes all the instances where
+ the frame was not conveying a cause code and was instead just
+ conveying a protocol-specific message. This also corrects the
+ generation of the message associated with disconnects for MFC/R2
+ to use the MFC/R2 specific text for the disconnect cause.
+
+2012-07-06 21:28 +0000 [r369764] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Add case for FLASH control
+ frames so that we don't display a warning. chan_sip channels can
+ receive flash control frames when connected to analog phones and
+ possibly for other reasons. There really isn't a reason to warn
+ when these frames are received, we can safely ignore them.
+ Patches: dahdi_sip_flash.diff uploaded by Jonathan Rose (license
+ 6182) ........ Merged revisions 369750 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369751 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-06 18:49 +0000 [r369710-369733] Mark Michelson <mmichelson@digium.com>
+
+ * main/tcptls.c, /: Remove a superfluous and dangerous freeing of
+ an SSL_CTX. The problem here is that multiple server sessions
+ share a SSL_CTX. When one session ended, the SSL_CTX would be
+ freed and set NULL, leaving the other sessions unable to
+ function. The code being removed is superfluous because the
+ SSL_CTX structures for servers will be properly freed when
+ ast_ssl_teardown is called. (closes issue ASTERISK-20074)
+ Reported by Trevor Helmsley Patches: ASTERISK-20074.diff uploaded
+ by Mark Michelson (license #5049) Testers: Trevor Helmsley
+ ........ Merged revisions 369731 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369732 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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
+ ........ Merged revisions 369708 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369709 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-06 14:32 +0000 [r369703] Joshua Colp <jcolp@digium.com>
+
+ * res/pjproject/pjnath/include/pjnath/ice_session.h,
+ res/pjproject/pjnath/src/pjnath/ice_session.c: Import revision
+ 4196 from pjproject trunk. Fix a crash issue when starting ICE
+ connectivity checks and immediately destroying the ICE session.
+ This was exposed by the SIP CCSS test. Full fix for this issue
+ will be worked on as a medium to long term roadmap item. pjroject
+ issue viewable at https://trac.pjsip.org/repos/ticket/1548
+
+2012-07-05 21:36 +0000 [r369681] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_stun_monitor.c, CHANGES: Add 'stun show status' command
+ This patch adds a new CLI command, 'stun show status'. This
+ command will show a table describing all known STUN servers and
+ statuses. (closes issue ASTERISK-18046) Reported by: Jeremy
+ Kister Tested by: Jeremy Kister patches:
+ (stun-show-status-v4-trunk.patch license #6232 uploaded by Jeremy
+ Kister) Review: https://reviewboard.asterisk.org/r/2001
+
+2012-07-05 19:36 +0000 [r369677] Richard Mudgett <rmudgett@digium.com>
+
+ * res/pjproject/pjmedia/include/pjmedia,
+ res/pjproject/pjsip/include/pjsip,
+ res/pjproject/pjlib/include/pj/compat,
+ res/pjproject/pjmedia/include/pjmedia-codec: Make res/pjproject
+ ignore more files.
+
+2012-07-05 19:36 +0000 [r369676] Kinsey Moore <kmoore@digium.com>
+
+ * /, apps/app_voicemail.c: AST-2012-011: Resolve heap corruption
+ issue with voicemail The heard and deleted arrays in the
+ voicemail state structure were not handled properly following the
+ memory leak fix in r354890 and a fix for an invalid free in
+ r356797. This could result in accessing and writing into freed
+ memory. The allocation for these arrays has been reworked to
+ avoid the possibility of invalid frees, access of freed memory,
+ and crashes that were occurring as a result of this. Locking
+ around accesses and modifications of the voicemail state
+ structure members dh_arraysize, heard, and deleted has been added
+ to prevent simultaneous modification and access when IMAP storage
+ is in use. If IMAP storage is not in use, this locking is not
+ compiled in. Review: https://reviewboard.asterisk.org/r/1994/
+ (closes issue ASTERISK-19923) Reported by: Dan Delaney Tested by:
+ Dan Delaney, Julian Yap Patches: vm_alloc_fix.diff uploaded by
+ kmoore (license 6273) ........ Merged revisions 369652 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369653 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-05 19:32 +0000 [r369666-369673] Richard Mudgett <rmudgett@digium.com>
+
+ * res/pjproject/pjsip/src/pjsip-ua,
+ res/pjproject/pjsip-apps/src/ipjsystest/ipjsystest.xcodeproj,
+ res/pjproject/pjnath/src/pjnath-test,
+ res/pjproject/third_party/build/speex,
+ res/pjproject/third_party/build/gsm/output,
+ res/pjproject/pjmedia/include/pjmedia-codec,
+ res/pjproject/third_party/build/baseclasses,
+ res/pjproject/third_party/build/srtp,
+ res/pjproject/pjsip-apps/src/samples,
+ res/pjproject/pjlib-util/lib, res/pjproject/pjmedia/bin,
+ res/pjproject/pjlib/include/pj++,
+ res/pjproject/tests/pjsua/scripts-call,
+ res/pjproject/third_party/srtp/doc,
+ res/pjproject/pjsip-apps/src/pocketpj/output,
+ res/pjproject/pjnath/bin,
+ res/pjproject/third_party/srtp/crypto/replay,
+ res/pjproject/pjsip/include/pjsip,
+ res/pjproject/third_party/build/speex/speex,
+ res/pjproject/build.symbian, res/pjproject/third_party/bin,
+ res/pjproject/pjsip/src/pjsua-lib,
+ res/pjproject/third_party/srtp/include,
+ res/pjproject/third_party/portaudio/doc, res/pjproject/lib,
+ res/pjproject/pjmedia/include/pjmedia-videodev,
+ res/pjproject/pjlib/bin,
+ res/pjproject/third_party/srtp/crypto/cipher,
+ res/pjproject/third_party/build/speex/output,
+ res/pjproject/pjlib-util/src/pjlib-util,
+ res/pjproject/third_party/portaudio/test,
+ res/pjproject/third_party/build/gsm,
+ res/pjproject/third_party/portaudio/include,
+ res/pjproject/pjsip-apps/src/pjsua_wince,
+ res/pjproject/pjsip/include/pjsip-simple,
+ res/pjproject/pjmedia/src/pjmedia-codec,
+ res/pjproject/tests/pjsua,
+ res/pjproject/pjsip-apps/src/pocketpj/res,
+ res/pjproject/pjsip-apps/src/3rdparty_media_sample,
+ res/pjproject/third_party/gsm/inc,
+ res/pjproject/pjsip-apps/build/wince-evc4,
+ res/pjproject/pjsip-apps/src/ipjsua/Resources-iPad,
+ res/pjproject/third_party/portaudio/src/hostapi,
+ res/pjproject/third_party/portaudio/build, res/pjproject/build,
+ res/pjproject/third_party/build/resample,
+ res/pjproject/third_party/speex/include,
+ res/pjproject/pjsip/src/pjsip,
+ res/pjproject/pjlib/build/wince-evc4,
+ res/pjproject/pjsip-apps/src/symbian_ua_gui/group,
+ res/pjproject/pjsip-apps/src/symbian_ua,
+ res/pjproject/tests/pjsua/wavs,
+ res/pjproject/third_party/portaudio/src/os/win,
+ res/pjproject/pjsip-apps/src/ipjsua/Classes,
+ res/pjproject/pjmedia/include/pjmedia,
+ res/pjproject/tests/pjsua/scripts-sendto,
+ res/pjproject/third_party/gsm/src,
+ res/pjproject/third_party/portaudio/build/msvc,
+ res/pjproject/pjsip-apps/src/confbot,
+ res/pjproject/pjnath/src/pjturn-client,
+ res/pjproject/pjlib-util/build/output,
+ res/pjproject/third_party/BaseClasses,
+ res/pjproject/third_party/portaudio/src/hostapi/wasapi,
+ res/pjproject/third_party/portaudio/src/hostapi/wdmks,
+ res/pjproject/pjlib/src/pj/compat,
+ res/pjproject/third_party/srtp/crypto/include,
+ res/pjproject/third_party/speex/include/speex,
+ res/pjproject/third_party/gsm/add-test,
+ res/pjproject/pjsip/build,
+ res/pjproject/pjsip-apps/src/pjsua_wince/output,
+ res/pjproject/third_party/gsm/lib, res/pjproject/pjsip,
+ res/pjproject/pjsip-apps/src/pjsystest,
+ res/pjproject/third_party/portaudio/src,
+ res/pjproject/third_party/speex/libspeex,
+ res/pjproject/pjsip/build/wince-evc4/output,
+ res/pjproject/pjlib-util/src/pjlib-util-test,
+ res/pjproject/pjsip-apps/src/symsndtest,
+ res/pjproject/third_party/srtp/tables,
+ res/pjproject/third_party/g7221, res/pjproject/pjmedia/include,
+ res/pjproject/pjlib/include/pj,
+ res/pjproject/third_party/build/portaudio/output,
+ res/pjproject/pjsip-apps/bin,
+ res/pjproject/pjsip-apps/src/ipjsua/ipjsua.xcodeproj,
+ res/pjproject/pjsip-apps/src/pjsua,
+ res/pjproject/third_party/srtp/test,
+ res/pjproject/pjsip/include/pjsip-ua,
+ res/pjproject/third_party/resample,
+ res/pjproject/third_party/build/ilbc,
+ res/pjproject/pjmedia/src/pjmedia-audiodev,
+ res/pjproject/pjsip-apps/src/ipjsua,
+ res/pjproject/third_party/srtp/srtp,
+ res/pjproject/third_party/build/milenage,
+ res/pjproject/pjmedia/src/pjmedia, res/pjproject/pjlib-util,
+ res/pjproject/third_party/portaudio/src/common,
+ res/pjproject/third_party/portaudio/bindings/cpp,
+ res/pjproject/pjlib-util/build/wince-evc4/output,
+ res/pjproject/third_party/srtp/crypto/kernel,
+ res/pjproject/tests/pjsua/scripts-pres, res/pjproject/pjnath,
+ res/pjproject/pjsip/build/output,
+ res/pjproject/pjsip-apps/build/output,
+ res/pjproject/pjsip-apps/build, res/pjproject/tests/automated,
+ res/pjproject/pjnath/build/wince-evc4/output,
+ res/pjproject/third_party/portaudio/src/hostapi/asio,
+ res/pjproject/pjnath/include/pjnath,
+ res/pjproject/pjsip/src/test,
+ res/pjproject/pjsip-apps/src/symbian_ua_gui/gfx,
+ res/pjproject/pjsip/bin,
+ res/pjproject/third_party/build/portaudio,
+ res/pjproject/pjlib/build/output, res/pjproject/pjmedia/src,
+ res/pjproject/pjlib/src/pj, res/pjproject/pjlib,
+ res/pjproject/pjlib/build/wince-evc4/output,
+ res/pjproject/pjmedia/src/test/vectors,
+ res/pjproject/third_party/portaudio/src/hostapi/jack,
+ res/pjproject/pjmedia/src/pjmedia-codec/g722,
+ res/pjproject/third_party/portaudio/src/hostapi/coreaudio,
+ res/pjproject/pjmedia/build/output,
+ res/pjproject/pjlib-util/include/pjlib-util,
+ res/pjproject/third_party/portaudio/src/hostapi/asihpi,
+ res/pjproject/third_party/milenage, res/pjproject/pjnath/src,
+ res/pjproject/tests/pjsua/scripts-run,
+ res/pjproject/pjlib-util/build/wince-evc4,
+ res/pjproject/pjmedia/lib, res/pjproject/pjmedia/src/test,
+ res/pjproject/third_party/speex/symbian,
+ res/pjproject/third_party/speex/win32,
+ res/pjproject/third_party/srtp/crypto/test,
+ res/pjproject/pjlib-util/bin,
+ res/pjproject/third_party/portaudio/build/scons,
+ res/pjproject/tests/cdash,
+ res/pjproject/tests/pjsua/scripts-media-playrec,
+ res/pjproject/third_party/build/portaudio/src,
+ res/pjproject/pjlib/src, res/pjproject/third_party/mp3,
+ res/pjproject/pjnath/lib, res/pjproject/third_party/build/g7221,
+ res/pjproject/third_party/gsm/man,
+ res/pjproject/third_party/portaudio/src/os/unix,
+ res/pjproject/third_party/portaudio/bindings,
+ res/pjproject/pjsip-apps/src/python,
+ res/pjproject/pjnath/src/pjnath, res/pjproject/third_party/lib,
+ res/pjproject/third_party/portaudio/src/os/mac_osx,
+ res/pjproject/third_party/srtp/crypto/ae_xfm,
+ res/pjproject/pjsip-apps/bin/samples,
+ res/pjproject/pjnath/src/pjturn-srv,
+ res/pjproject/third_party/portaudio/pablio,
+ res/pjproject/pjlib/lib, res/pjproject/third_party/g7221/decode,
+ res/pjproject/pjlib/include/pj/compat,
+ res/pjproject/third_party/gsm,
+ res/pjproject/third_party/build/baseclasses/output,
+ res/pjproject/third_party/build/srtp/output,
+ res/pjproject/third_party/srtp, res/pjproject/pjnath/build,
+ res/pjproject/tests/pjsua/scripts-sipp, res/pjproject/pjsip-apps,
+ res/pjproject/pjnath/build/wince-evc4,
+ res/pjproject/third_party/srtp/crypto/rng,
+ res/pjproject/pjsip/build/wince-evc4,
+ res/pjproject/pjsip-apps/build/wince-evc4/output,
+ res/pjproject/third_party/gsm/tst,
+ res/pjproject/third_party/portaudio/src/hostapi/dsound,
+ res/pjproject/third_party/portaudio/testcvs,
+ res/pjproject/pjsip-apps/src/ipjsystest/Classes,
+ res/pjproject/pjlib/build, res/pjproject/third_party/portaudio,
+ res/pjproject/third_party/portaudio/src/hostapi/wmme,
+ res/pjproject/pjlib-util/docs,
+ res/pjproject/pjmedia/include/pjmedia-audiodev,
+ res/pjproject/pjsip-apps/src/vidgui,
+ res/pjproject/pjlib/src/pjlib-test,
+ res/pjproject/pjsip-apps/src/py_pjsua,
+ res/pjproject/third_party/portaudio/src/os,
+ res/pjproject/pjsip/include,
+ res/pjproject/pjmedia/build/wince-evc4,
+ res/pjproject/pjmedia/src/pjmedia-videodev,
+ res/pjproject/pjsip-apps/src, res/pjproject/third_party/speex,
+ res/pjproject/third_party/gsm/tls,
+ res/pjproject/third_party/g7221/common,
+ res/pjproject/tests/pjsua/tools,
+ res/pjproject/third_party/resample/include,
+ res/pjproject/third_party/build/samplerate/output,
+ res/pjproject/third_party/build/samplerate,
+ res/pjproject/third_party/gsm/bin,
+ res/pjproject/pjsip/src/pjsip-simple,
+ res/pjproject/third_party/g7221/encode,
+ res/pjproject/pjlib/src/pjlib-samples,
+ res/pjproject/pjsip-apps/lib,
+ res/pjproject/pjsip-apps/src/ipjsystest,
+ res/pjproject/pjlib-util/include,
+ res/pjproject/third_party/build/resample/output,
+ res/pjproject/third_party/build/ilbc/output,
+ res/pjproject/third_party/srtp/crypto,
+ res/pjproject/pjsip-apps/src/python/samples, res/pjproject/tests,
+ res/pjproject/pjsip-apps/src/symbian_ua_gui/sis,
+ res/pjproject/pjnath/include,
+ res/pjproject/pjsip-apps/src/symbian_ua_gui,
+ res/pjproject/pjmedia/build, res/pjproject/pjmedia,
+ res/pjproject/third_party/build/milenage/output,
+ res/pjproject/pjlib-util/build, res/pjproject/pjsip/src,
+ res/pjproject/pjmedia/build/wince-evc4/output,
+ res/pjproject/third_party/portaudio/src/hostapi/alsa,
+ res/pjproject/pjsip-apps/docs,
+ res/pjproject/pjsip-apps/src/symbian_ua_gui/inc,
+ res/pjproject/pjsip-apps/src/symbian_ua_gui/data,
+ res/pjproject/tests/pjsua/scripts-pesq,
+ res/pjproject/third_party/srtp/pjlib,
+ res/pjproject/pjlib/include, res/pjproject/pjnath/build/output,
+ res/pjproject/third_party/srtp/crypto/hash,
+ res/pjproject/build/vs, res/pjproject/pjlib/docs,
+ res/pjproject/third_party/build,
+ res/pjproject/third_party/resample/src,
+ res/pjproject/third_party, res/pjproject/pjlib/src/pjlib++-test,
+ res/pjproject/third_party/build/g7221/output,
+ res/pjproject/third_party/srtp/crypto/math,
+ res/pjproject/pjsip/lib, res/pjproject/pjsip-apps/src/pocketpj,
+ res/pjproject/tests/pjsua/scripts-recvfrom,
+ res/pjproject/third_party/portaudio/build/dev-cpp,
+ res/pjproject/pjsip/include/pjsua-lib,
+ res/pjproject/pjsip-apps/src/symbian_ua_gui/src, res/pjproject,
+ res/pjproject/third_party/portaudio/src/hostapi/oss,
+ res/pjproject/pjlib-util/src, res/pjproject/third_party/ilbc:
+ Make res/pjproject ignore some generated files.
+
+ * include/asterisk/utils.h: Tweak some comments and whitespace in
+ utils.h
+
+2012-07-05 18:11 +0000 [r369644] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_mixmonitor.c: app_mixmonitor: Fix a reference leak in
+ manager_mixmonitor function Manager_mixmonitor included an early
+ return on failed executions of mixmonitor that would result in a
+ leaked channel reference. (closes issue ASTERISK-19943) Reported
+ by: Mark Murawski Patches: mixmonitor-trunk-368394.patch uploaded
+ by Mark Murawski (license 5791)
+
+2012-07-05 17:03 +0000 [r369628] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Do not send a BYE when a provisional
+ response arrives during a re-INVITE Commits r369557 and r369579
+ were done to improve handling of re-INVITEs when the UA that was
+ supposed to receive the re-INVITE fails to respond. A limitation
+ of those patches occurred when a UA sent a provisional response
+ to the re-INVITE. This triggered a sending of a BYE in
+ check_pending. This patch tweaks the handling of the re-INVITE
+ such that a BYE is not sent in response to those messages. (issue
+ ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies
+ patches: (reinvite_tweak.diff license #5012 by Steve Davies)
+ ........ Merged revisions 369626 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369627 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-05 11:42 +0000 [r369602-369620] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooCmdChannel.c,
+ addons/ooh323c/src/ooStackCmds.c, addons/ooh323c/src/ooq931.c:
+ Fix dev mode ooh323 warnings
+
+ * addons/chan_ooh323.c, addons/ooh323c/src/ooq931.h,
+ addons/ooh323c/src/ooCalls.h, configs/chan_ooh323.conf.sample
+ (removed), addons/ooh323c/src/ooh323ep.c, CHANGES,
+ configs/ooh323.conf.sample (added),
+ addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooStackCmds.c,
+ addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooLogChan.h,
+ addons/ooh323c/src/ooStackCmds.h, addons/ooh323c/src/ooh245.c,
+ addons/ooh323cDriver.c, addons/ooh323c/src/ooh245.h,
+ addons/ooh323c/src/ooCmdChannel.c, addons/ooh323c/src/ooq931.c:
+ Added direct media support to ooh323 channel driver options are
+ documented in config sample sample config rename to proper name -
+ ooh323.conf To change media address ooh323 send empty TCS if
+ there was completed TCS exchange or send facility
+ forwardedelements with new fast start proposal if not. Then close
+ transmit logical channels and renew TCS exchange. If new fast
+ start proposal is received then ooh323 stack call back channel
+ driver routine to change rtp address in the rtp instance. If
+ empty TCS is received then close transmit logical channels and
+ renew TCS exchange Review:
+ https://reviewboard.asterisk.org/r/1607/
+
+ * addons/ooh323cDriver.c: fix small mistake in the previous
+
+ * addons/ooh323c/src/ooTimer.c, addons/ooh323c/src/ooCapability.c,
+ addons/ooh323c/src/decode.c, addons/ooh323c/src/perutil.c,
+ addons/ooh323cDriver.c, addons/ooh323c/src/ooSocket.c,
+ addons/ooh323c/src/ooq931.c: Fix modern gcc warning Review:
+ https://reviewboard.asterisk.org/r/1767
+
+2012-07-03 17:07 +0000 [r369559-369581] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: More improvements to re-INVITEs timing
+ out after a provisional response There is no need to call
+ check_pendings() on a final response to an INVITE when destroying
+ the scheduler entry as it will be done later during normal
+ processing. (issue ASTERISK-19992) ........ Merged revisions
+ 369579 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 369580 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: 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/ ........ Merged
+ revisions 369557 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369558 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-07-02 14:06 +0000 [r369517-369527] Joshua Colp <jcolp@digium.com>
+
+ * configs/xmpp.conf.sample (added), include/asterisk/xmpp.h
+ (added), configs/cli_aliases.conf.sample, res/res_xmpp.c (added):
+ Add a cleaned up drop-in replacement for res_jabber called
+ res_xmpp. This provides the same externally facing functionality
+ but is implemented differently internally. This is currently not
+ built by default but this will be changed once chan_jingle2
+ (insert actual name in your head when reading this after it has
+ been merged) is in the tree. Review:
+ https://reviewboard.asterisk.org/r/1983/
+
+ * res/res_rtp_asterisk.c: Ensure the timer heap is protected by a
+ lock.
+
+ * res/pjproject/pjlib/include/pj/config_site.h: Enable IPv6 support
+ in pjproject.
+
+ * res/res_rtp_asterisk.c: Don't try to send connectivity checks on
+ RTCP if RTCP is no longer present and don't do multiple ICE
+ connectivity checks at once.
+
+ * res/pjproject/pjlib/src/pj/sock_qos_common.c (added),
+ res/pjproject/pjlib-util/src/pjlib-util/crc32.c (added),
+ res/pjproject/pjsip/src/pjsip-simple/xpidf.c (added),
+ res/pjproject/third_party/gsm/src/gsm_implode.c (added),
+ res/pjproject/tests/pjsua/scripts-sipp/uas-cancel-no-final.xml
+ (added), res/pjproject/build.symbian/pjmedia.mmp (added),
+ res/pjproject/third_party/build/portaudio/src/pa_hostapi.h
+ (added), res/pjproject/pjlib/src/pjlib-test/fifobuf.c (added),
+ res/pjproject/pjlib/src/pj/file_access_unistd.c (added),
+ res/pjproject/third_party/gsm/src/toast_ulaw.c (added),
+ res/pjproject/pjsip/include/pjsip/sip_transport_tls.h (added),
+ res/pjproject/pjsip/include/pjsip/sip_multipart.h (added),
+ res/pjproject/pjmedia/src/pjmedia/errno.c (added),
+ res/pjproject/pjsip-apps/src/pjsua_wince/pjsua_wince.vcp (added),
+ res/pjproject/third_party/speex/COPYING (added),
+ res/pjproject/pjlib/src/pj/os_core_darwin.m (added),
+ res/pjproject/third_party/ilbc/packing.c (added),
+ res/pjproject/third_party/build/portaudio/src/pa_mac_core_internal.h
+ (added),
+ res/pjproject/tests/pjsua/scripts-sendto/300_srtp_receive_crypto_tag_zero.py
+ (added), res/pjproject/third_party/ilbc/packing.h (added),
+ res/pjproject/pjlib/src/pj/pool_caching.c (added),
+ res/pjproject/pjnath/include/pjnath/errno.h (added),
+ res/pjproject/pjmedia/include/pjmedia-codec/h264_packetizer.h
+ (added), res/pjproject/pjmedia/include/pjmedia/sdp_neg.h (added),
+ res/pjproject/third_party/speex/libspeex/lsp_bfin.h (added),
+ res/pjproject/third_party/portaudio/aclocal.m4 (added),
+ res/pjproject/third_party/mp3/mp3_port.h (added),
+ res/pjproject/third_party/BaseClasses/ctlutil.cpp (added),
+ res/pjproject/pjsip-apps/src/pocketpj/PocketPJDlg.cpp (added),
+ res/pjproject/tests/pjsua/scripts-recvfrom/240_publish_scenarios.py
+ (added), res/pjproject/README-RTEMS (added),
+ res/pjproject/third_party/build/portaudio/output (added),
+ res/pjproject/pjsip-apps/build/Makefile (added),
+ res/pjproject/tests/pjsua/scripts-sipp/prack_fork.xml (added),
+ res/pjproject/pjlib-util/src/pjlib-util-test/stun.c (added),
+ res/pjproject/pjlib-util/src/pjlib-util/dns_dump.c (added),
+ res/pjproject/pjmedia/include/pjmedia/circbuf.h (added),
+ res/pjproject/pjlib/build/os-darwinos.mak (added),
+ res/pjproject/third_party/srtp/test/rtpw.c (added),
+ res/pjproject/tests/pjsua/scripts-sipp/uas-answer-180-multiple-fmts.xml
+ (added),
+ res/pjproject/third_party/srtp/crypto/include/cryptoalg.h
+ (added), res/pjproject/third_party/portaudio/bindings/cpp
+ (added),
+ res/pjproject/tests/pjsua/scripts-sipp/uas-answer-200-reinvite-without-sdp.xml
+ (added), res/pjproject/third_party/portaudio/configure.in
+ (added), res/pjproject/pjmedia/include/pjmedia-codec/g722.h
+ (added), res/pjproject/pjsip-apps/src/vidgui/pj-pkgconfig.mak
+ (added), res/pjproject/pjmedia/include/pjmedia-codec/speex.h
+ (added), res/pjproject/config.guess (added),
+ res/pjproject/tests/cdash/cfg_site_sample.py (added),
+ res/pjproject/third_party/portaudio/src/common/pa_skeleton.c
+ (added),
+ res/pjproject/pjsip-apps/src/symbian_ua_gui/inc/symbian_ua_guiSettingItemList.hrh
+ (added), res/pjproject/third_party/srtp/test/getopt_s.c (added),
+ res/pjproject/pjmedia/src/pjmedia-codec/g722 (added),
+ res/pjproject/tests/pjsua/scripts-pesq/201_codec_g722.py (added),
+ res/pjproject/pjnath/src/pjturn-client/client_main.c (added),
+ res/pjproject/third_party/gsm/src/short_term.c (added),
+ res/pjproject/build.symbian/libg7221codec.mmp (added),
+ res/pjproject/pjmedia/src/pjmedia/wsola.c (added),
+ res/pjproject/pjlib-util/include/pjlib-util/hmac_sha1.h (added),
+ res/pjproject/pjlib/include/pj++/list.hpp (added),
+ res/pjproject/third_party/ilbc/anaFilter.c (added),
+ res/pjproject/third_party/mp3 (added),
+ res/pjproject/pjmedia/src/pjmedia/tonegen.c (added),
+ res/pjproject/pjsip-apps/src/samples/stateful_proxy.c (added),
+ res/pjproject/third_party/ilbc/anaFilter.h (added),
+ res/pjproject/pjsip-apps/src/symsndtest/app_main.cpp (added),
+ res/pjproject/pjsip-apps/src/pocketpj/SettingsDlg.cpp (added),
+ res/pjproject/tests/pjsua/scripts-sipp/uas-invite.xml (added),
+ res/pjproject/third_party/g7221/encode/sam2coef.c (added),
+ res/pjproject/pjlib/src/pj/compat/string.c (added),
+ res/pjproject/pjlib/include/pj/compat/cc_gcce.h (added),
+ res/pjproject/pjlib/include/pj/config_site_sample.h (added),
+ res/pjproject/third_party/build/srtp/output (added),
+ res/pjproject/tests/pjsua/scripts-pesq/200_codec_speex_8000.py
+ (added), res/pjproject/tests/pjsua/scripts-sipp/uac-options.xml
+ (added), res/pjproject/third_party/ilbc/iCBConstruct.c (added),
+ res/pjproject/tests/pjsua/scripts-sendto/153_err_sdp_unsupported_codec.py
+ (added), res/pjproject/pjsip/build/wince-evc4 (added),
+ res/pjproject/third_party/ilbc/iCBConstruct.h (added),
+ res/pjproject/pjsip-apps/src/py_pjsua/py_pjsua.def (added),
+ res/pjproject/pjnath/build/pjstun_srv_test.vcproj (added),
+ res/pjproject/pjlib/src/pjlib-test/util.c (added),
+ res/pjproject/pjmedia/include/pjmedia-audiodev (added),
+ res/pjproject/pjlib/src/pj/ctype.c (added),
+ res/pjproject/third_party/ilbc/enhancer.c (added),
+ res/pjproject/pjsip-apps/src/py_pjsua (added),
+ res/pjproject/third_party/speex/libspeex/modes_wb.c (added),
+ res/pjproject/third_party/gsm/tst/gsm2cod.c (added),
+ res/pjproject/third_party/ilbc/enhancer.h (added),
+ res/pjproject/pjsip-apps/src (added),
+ res/pjproject/build/m-arm.mak (added),
+ res/pjproject/third_party/gsm/src/add.c (added),
+ res/pjproject/pjsip/src/pjsip/sip_parser_wrap.cpp (added),
+ res/pjproject/pjlib/src/pj/timer_symbian.cpp (added),
+ res/pjproject/pjsip-apps/src/vidgui/vidwin.cpp (added),
+ res/pjproject/pjlib/include/pj/pool_buf.h (added),
+ res/pjproject/third_party/g7221/encode (added),
+ res/pjproject/pjmedia/src/pjmedia-audiodev/wmme_dev.c (added),
+ res/pjproject/tests/pjsua/scripts-call/300_ice_1_0.py (added),
+ res/pjproject/tests/pjsua/config_site.py (added),
+ res/pjproject/pjsip-apps/src/pjsua/main.c (added),
+ res/pjproject/pjlib/src/pj/os_timestamp_posix.c (added),
+ res/pjproject/pjmedia/include/pjmedia-videodev/videodev_imp.h
+ (added),
+ res/pjproject/tests/pjsua/scripts-recvfrom/230_reg_bad_fail_stale_true.py
+ (added), res/pjproject/third_party/srtp/config.h_win32vc7
+ (added), res/pjproject/tests/pjsua/scripts-pesq (added),
+ res/pjproject/tests/pjsua/scripts-sipp/uas-reinv-glare.xml
+ (added), res/pjproject/pjmedia/src/pjmedia/dummy.c (added),
+ res/pjproject/tests/pjsua/scripts-recvfrom/209c_reg_handle_423_bad_min_expires2.py
+ (added), res/pjproject/pjlib/include/pj++/hash.hpp (added),
+ res/pjproject/pjmedia/include/pjmedia-audiodev/audiodev_imp.h
+ (added),
+ res/pjproject/tests/pjsua/scripts-sendto/401_fmtp_g7221_with_bitrate_24000.py
+ (added), res/pjproject/pjsip-apps/src/pjsua/pjsua_app.c (added),
+ res/pjproject/pjsip-apps/src/samples/stereotest.c (added),
+ res/pjproject/build.symbian/pjstun_client.mmp (added),
+ res/pjproject/pjsip-apps/src/pjsua_wince/pjsua_wince.cpp (added),
+ res/pjproject/pjsip-apps/src/ipjsua/Classes/FirstViewController.h
+ (added), res/pjproject/pjlib-util/lib (added),
+ res/pjproject/pjsip-apps/src/samples (added),
+ res/pjproject/pjsip-apps/src/ipjsua/Classes/FirstViewController.m
+ (added), res/pjproject/tests/pjsua/scripts-call/150_srtp_1_1.py
+ (added), res/pjproject/pjmedia/include/pjmedia/vid_stream.h
+ (added), res/pjproject/pjsip/src/pjsip/sip_dialog.c (added),
+ res/pjproject/pjlib/include/pj/compat/cc_armcc.h (added),
+ res/pjproject/third_party/build/speex/speex (added),
+ res/pjproject/third_party/bin (added),
+ res/pjproject/pjsip/build/Makefile (added),
+ res/pjproject/pjlib-util/include/pjlib-util/stun_simple.h
+ (added), res/pjproject/pjsip/src/pjsip/sip_util_proxy_wrap.cpp
+ (added), res/pjproject/pjlib/include/pj/compat/m_m68k.h (added),
+ res/pjproject/third_party/srtp/srtp.def (added),
+ res/pjproject/pjlib/src/pjlib-test/rand.c (added),
+ res/pjproject/third_party/build/gsm/config.h (added),
+ res/pjproject/pjmedia/include/pjmedia/avi.h (added),
+ res/pjproject/tests/pjsua/scripts-sipp/uac-bad-ack.xml (added),
+ res/pjproject/tests/pjsua/scripts-pesq/200_codec_gsm.py (added),
+ res/pjproject/pjsip/src/pjsip-ua/sip_reg.c (added),
+ res/pjproject/pjsip/build/wince-evc4/pjsip_ua_wince.vcp (added),
+ res/pjproject/pjsip/include/pjsip-ua/sip_regc.h (added),
+ res/pjproject/tests/pjsua/mod_pesq.py (added),
+ res/pjproject/pjnath/src/pjnath/ice_session.c (added),
+ res/pjproject/pjlib-util/src/pjlib-util/scanner.c (added),
+ res/pjproject/pjmedia/src/pjmedia-audiodev/audiodev.c (added),
+ res/pjproject/pjsip-apps/src/confbot/confbot.py (added),
+ res/pjproject/tests/pjsua/scripts-call/150_srtp_0_3.py (added),
+ res/pjproject/pjsip-apps/src/3rdparty_media_sample/alt_pjsua_vid.c
+ (added), res/pjproject/tests/pjsua/tools/cmp_wav.c (added),
+ res/pjproject/tests/pjsua/scripts-sendto/320_srtp_with_unknown_media_2.py
+ (added), res/pjproject/pjsip-apps/src/symbian_ua (added),
+ res/pjproject/pjmedia/src/pjmedia-audiodev/alsa_dev.c (added),
+ res/pjproject/third_party/portaudio/build/msvc (added),
+ res/pjproject/pjmedia/src/pjmedia/sound_legacy.c (added),
+ res/pjproject/third_party/ilbc/lsf.c (added),
+ res/pjproject/pjsip/src/test/inv_offer_answer_test.c (added),
+ res/pjproject/pjsip-apps/src/confbot (added),
+ res/pjproject/third_party/portaudio/src/hostapi/coreaudio/pa_mac_core_utilities.c
+ (added), res/pjproject/third_party/speex/libspeex/ltp_bfin.h
+ (added),
+ res/pjproject/pjsip-apps/src/symbian_ua_gui/group/ABLD.BAT
+ (added), res/pjproject/pjlib/src/pj/ioqueue_winnt.c (added),
+ res/pjproject/third_party/ilbc/lsf.h (added),
+ res/pjproject/third_party/speex/libspeex/lsp_tables_nb.c (added),
+ res/pjproject/third_party/portaudio/src/hostapi/coreaudio/pa_mac_core_utilities.h
+ (added),
+ res/pjproject/third_party/portaudio/build/scons/SConscript_common
+ (added), res/pjproject/pjmedia/include/pjmedia/frame.h (added),
+ res/pjproject/pjmedia/src/pjmedia-codec/audio_codecs.c (added),
+ res/pjproject/pjlib-util/src/pjlib-util/xml_wrap.cpp (added),
+ res/pjproject/pjsip-apps/src/pocketpj/res/PocketPJ.rc2 (added),
+ res/pjproject/third_party/build/portaudio/src/pa_mac_core_utilities.c
+ (added),
+ res/pjproject/pjmedia/include/pjmedia-audiodev/audiotest.h
+ (added), res/pjproject/pjlib/src/pj/guid_win32.c (added),
+ res/pjproject/pjlib/build/os-sunos.mak (added),
+ res/pjproject/third_party/build/srtp/Makefile (added),
+ res/pjproject/third_party/speex/libspeex/gain_table.c (added),
+ res/pjproject/third_party/build/portaudio/src/pa_mac_core_utilities.h
+ (added), res/pjproject/third_party/BaseClasses/wxlist.h (added),
+ res/pjproject/tests/pjsua/scripts-sendto/122_sdp_with_unknown_dynamic_1.py
+ (added),
+ res/pjproject/tests/pjsua/scripts-sendto/001_torture_4475_3_1_1_5.py
+ (added), res/pjproject/pjsip-apps/src/pjsua/gui.h (added),
+ res/pjproject/third_party/srtp/crypto/test/auth_driver.c (added),
+ res/pjproject/pjlib/include/pj/activesock.h (added),
+ res/pjproject/pjlib/src/pjlib-test/exception.c (added),
+ res/pjproject/pjlib/src/pjlib-test/main_rtems.c (added),
+ res/pjproject/third_party/build/portaudio/src/pa_linux_alsa.c
+ (added), res/pjproject/pjlib-util/src/pjlib-util/symbols.c
+ (added), res/pjproject/pjlib/include/pj/types.h (added),
+ res/pjproject/pjnath/src/pjnath/turn_sock.c (added),
+ res/pjproject/pjlib-util/src/pjlib-util/resolver_wrap.cpp
+ (added),
+ res/pjproject/third_party/build/portaudio/src/pa_linux_alsa.h
+ (added), res/pjproject/pjlib/include/pj/compat/errno.h (added),
+ res/pjproject/tests/pjsua/scripts-recvfrom/100_simple.py (added),
+ res/pjproject/pjsip-apps/src/ipjsystest/RootViewController.xib
+ (added), res/pjproject/pjlib/build/wince-evc4/output (added),
+ res/pjproject/pjlib/src/pjlib-test/echo_clt.c (added),
+ res/pjproject/third_party/portaudio/src/os/unix/pa_unix_util.c
+ (added), res/pjproject/pjsip/build/wince-evc4/pjsua_lib_wince.vcp
+ (added), res/pjproject/svn_add (added),
+ res/pjproject/third_party/portaudio/src/hostapi/coreaudio/pa_mac_core_old.c
+ (added), res/pjproject/pjlib-util/build/wince-evc4 (added),
+ res/pjproject/pjmedia/src/test (added),
+ res/pjproject/third_party/srtp/crypto/test (added),
+ res/pjproject/third_party/portaudio/src/os/unix/pa_unix_util.h
+ (added), res/pjproject/tests/pjsua/scripts-media-playrec (added),
+ res/pjproject/pjsip-apps/src/samples/vid_streamutil.c (added),
+ res/pjproject/pkgconfig.py (added),
+ res/pjproject/third_party/srtp/crypto/hash/sha1.c (added),
+ res/pjproject/pjlib/src/pj/addr_resolv_sock.c (added),
+ res/pjproject/pjnath/src/pjturn-srv (added),
+ res/pjproject/pjmedia/include/pjmedia/wav_playlist.h (added),
+ res/pjproject/pjsip/include/pjsip/sip_resolve.h (added),
+ res/pjproject/pjmedia/src/pjmedia-codec/ilbc.c (added),
+ res/pjproject/pjmedia/src/pjmedia/format.c (added),
+ res/pjproject/pjsip/src/pjsip/sip_dialog_wrap.cpp (added),
+ res/pjproject/third_party/speex/include/speex/speex_buffer.h
+ (added),
+ res/pjproject/pjmedia/src/pjmedia/transport_adapter_sample.c
+ (added), res/pjproject/pjsip-apps/src/vidgui/vidwin.h (added),
+ res/pjproject/pjlib/src/pjlib-test/main_symbian.cpp (added),
+ res/pjproject/pjlib/docs/doxygen.css (added),
+ res/pjproject/third_party/gsm/src/gsm_explode.c (added),
+ res/pjproject/pjsip-apps/src/symbian_ua_gui/inc/symbian_ua_guiContainer.hrh
+ (added), res/pjproject/third_party/speex/libspeex/kiss_fftr.c
+ (added), res/pjproject/pjlib/src/pj/addr_resolv_linux_kernel.c
+ (added), res/pjproject/third_party/gsm/tst/lin2cod.c (added),
+ res/pjproject/pjmedia/src/pjmedia-codec/l16.c (added),
+ res/pjproject/third_party/speex/libspeex/kiss_fftr.h (added),
+ res/pjproject/pjsip-apps/src/pocketpj/SettingsDlg.h (added),
+ res/pjproject/third_party/resample/src/stddefs.h (added),
+ res/pjproject/pjmedia/src/pjmedia/rtcp_xr.c (added),
+ res/pjproject/pjsip-apps/src/vidgui/vidgui.cpp (added),
+ res/pjproject/pjsip/src/pjsip/sip_resolve.c (added),
+ res/pjproject/pjsip/src/test/transport_tcp_test.c (added),
+ res/pjproject/tests/pjsua/scripts-sendto/121_sdp_with_video_static_2.py
+ (added), res/pjproject/build.symbian/libpassthroughcodec.mmp
+ (added), res/pjproject/third_party/srtp/crypto/rng/ctr_prng.c
+ (added),
+ res/pjproject/tests/pjsua/scripts-sipp/uas-subscribe-notify-terminate.xml
+ (added), res/pjproject/third_party/portaudio/fixfile.bat (added),
+ res/pjproject/pjsip/src/test/multipart_test.c (added),
+ res/pjproject/pjsip-apps/lib (added),
+ res/pjproject/third_party/portaudio/pablio/pablio.c (added),
+ res/pjproject/pjmedia/src/pjmedia/rtp.c (added),
+ res/pjproject/pjmedia/src/pjmedia/stereo_port.c (added),
+ res/pjproject/pjsip/src/test/tsx_uas_test.c (added),
+ res/pjproject/third_party/portaudio/pablio/pablio.h (added),
+ res/pjproject/third_party/speex/libspeex/vq_bfin.h (added),
+ res/pjproject/pjmedia/include/pjmedia/bidirectional.h (added),
+ res/pjproject/third_party/BaseClasses/arithutil.cpp (added),
+ res/pjproject/third_party/build/milenage/output (added),
+ res/pjproject/pjlib-util/src/pjlib-util/http_client.c (added),
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+ res/pjproject/pjnath/src/pjturn-srv/auth.h (added),
+ res/pjproject/pjmedia/src/pjmedia/stream_common.c (added),
+ res/pjproject/pjlib/include/pj/compat/m_auto.h.in (added),
+ res/pjproject/tests/pjsua/scripts-pesq/200_codec_l16_8000.py
+ (added), res/pjproject/pjmedia/src/pjmedia/rtcp.c (added),
+ res/pjproject/pjlib/build/os-auto.mak.in (added),
+ res/pjproject/tests/cdash/cfg_msvc.py (added),
+ res/pjproject/third_party/gsm/src/lpc.c (added),
+ res/pjproject/third_party/resample/README.resample (added),
+ res/pjproject/pjsip-apps/src/ipjsua/MainWindow.xib (added),
+ res/pjproject/third_party/portaudio/include/pa_win_wmme.h
+ (added), res/pjproject/third_party/speex/symbian/config.h
+ (added), res/pjproject/pjnath/src/pjnath-test/test.c (added),
+ res/pjproject/pjsip-apps/src/samples/siprtp_report.c (added),
+ res/pjproject/pjnath/src/pjnath-test/test.h (added),
+ res/pjproject/third_party/srtp (added),
+ res/pjproject/third_party/build/g7221/libg7221codec.vcproj
+ (added),
+ res/pjproject/pjlib-util/build/wince-evc4/pjlib_util_test_wince.vcp
+ (added), res/pjproject/build/m-i386.mak (added),
+ res/pjproject/pjlib-util/include/pjlib-util/srv_resolver.h
+ (added), res/pjproject/tests/pjsua/scripts-call/150_srtp_3_2.py
+ (added), res/pjproject/pjsip/include/pjsip_simple.h (added),
+ res/pjproject/pjmedia/src/test/audio_tool.c (added),
+ res/pjproject/pjlib/src/pj/exception_symbian.cpp (added),
+ res/pjproject/pjmedia/build/m-i386.mak (added),
+ res/pjproject/third_party/BaseClasses/wxutil.h (added),
+ res/pjproject/pjsip-apps/src/vidgui (added),
+ res/pjproject/pjsip/src/pjsua-lib/pjsua_media.c (added),
+ res/pjproject/pjlib-util/build/pjlib_util.vcproj (added),
+ res/pjproject/pjnath/include/pjnath/stun_transaction.h (added),
+ res/pjproject/third_party/portaudio/src/hostapi/oss/recplay.c
+ (added), res/pjproject/third_party/resample/include (added),
+ res/pjproject/pjmedia/include/pjmedia/transport.h (added),
+ res/pjproject/pjlib-util/src/pjlib-util/srv_resolver.c (added),
+ res/pjproject/build.symbian/pjsua_libU.def (added),
+ res/pjproject/pjsip/src/pjsip-simple/presence_body.c (added),
+ res/pjproject/pjmedia/include/pjmedia/stereo.h (added),
+ res/pjproject/tests/pjsua/scripts-call/301_ice_public_b.py
+ (added), res/pjproject/tests/automated/configure.py (added),
+ res/pjproject/pjsip-apps/src/symbian_ua/symbian_ua_reg.rss
+ (added), res/pjproject/pjsip-apps/src/pocketpj/PopUpWnd.h
+ (added),
+ res/pjproject/third_party/speex/libspeex/high_lsp_tables.c
+ (added), res/pjproject/pjlib/src/pj/ssl_sock_ossl.c (added),
+ res/pjproject/tests/pjsua/scripts-sendto/313_srtp1_unsupported_crypto.py
+ (added), res/pjproject/pjsip-apps/src/symbian_ua_gui (added),
+ res/pjproject/third_party/build/milenage/libmilenage.vcp (added),
+ res/pjproject/pjmedia/include/pjmedia/transport_loop.h (added),
+ res/pjproject/third_party/build/gsm/libgsmcodec.vcp (added),
+ res/pjproject/third_party/speex/libspeex/window.c (added),
+ res/pjproject/tests/pjsua/scripts-sendto/125_sdp_with_multi_audio_2.py
+ (added), res/pjproject/pjsip-apps/src/symbian_ua_gui/data
+ (added), res/pjproject/pjsip/src/pjsip/sip_transport_wrap.cpp
+ (added), res/pjproject/pjmedia/include/pjmedia-videodev/errno.h
+ (added), res/pjproject/pjlib/src/pj/os_time_common.c (added),
+ res/pjproject/third_party/resample/src (added),
+ res/pjproject/pjlib/docs (added),
+ res/pjproject/tests/pjsua/scripts-sendto/161_err_replaces_dlg_not_found.py
+ (added), res/pjproject/pjsip-apps/src/pocketpj (added),
+ res/pjproject/pjsip-apps/src/samples/simple_pjsua.c (added),
+ res/pjproject/pjsip-apps/src/symbian_ua_gui/src (added),
+ res/pjproject/tests/pjsua/scripts-media-playrec/100_resample_lf_11_48.py
+ (added), res/pjproject/pjmedia/include/pjmedia/rtcp.h (added),
+ res/pjproject/tests/pjsua/scripts-sendto/300_srtp_invalid_crypto_tag_non_numeric.py
+ (added), res/pjproject/tests/pjsua/scripts-pres/100_peertopeer.py
+ (added), res/pjproject/pjmedia/src/pjmedia/vid_codec_util.c
+ (added), res/pjproject/third_party/gsm/MACHINES (added),
+ res/pjproject/tests/pjsua/scripts-sipp/uac-subscribe.xml (added),
+ res/pjproject/third_party/build/baseclasses (added),
+ res/pjproject/third_party/srtp/include/srtp.h (added),
+ res/pjproject/tests/pjsua/scripts-sendto/173_timer_offer_refresher_uac.py
+ (added), res/pjproject/pjmedia/src/pjmedia/stream.c (added),
+ res/pjproject/tests/pjsua/scripts-recvfrom/209a_reg_handle_423_ok.py
+ (added), res/pjproject/pjlib/src/pjlib-samples/log.c (added),
+ res/pjproject/third_party/build/portaudio/src/pa_mac_core_old.c
+ (added), res/pjproject/pjsip/src/pjsip/sip_transport_tcp.c
+ (added),
+ res/pjproject/tests/pjsua/scripts-sendto/150_err_extension.py
+ (added),
+ res/pjproject/pjlib-util/src/pjlib-util-test/encryption.c
+ (added), res/pjproject/lib (added),
+ res/pjproject/pjmedia/include/pjmedia/codec.h (added),
+ res/pjproject/pjmedia/src/pjmedia/converter_libswscale.c (added),
+ res/pjproject/pjlib/src/pj/ip_helper_win32.c (added),
+ res/pjproject/pjmedia/include/pjmedia-videodev/avi_dev.h (added),
+ res/pjproject/pjlib-util/src/pjlib-util/scanner_cis_bitwise.c
+ (added), res/pjproject/third_party/gsm/README (added),
+ res/pjproject/pjlib-util/src/pjlib-util (added),
+ res/pjproject/third_party/build/gsm (added),
+ res/pjproject/pjlib/include/pj/compat/cc_msvc.h (added),
+ res/pjproject/pjsip-apps/src/pjsua_wince (added),
+ res/pjproject/tests/pjsua (added),
+ res/pjproject/pjlib/include/pj++/timer.hpp (added),
+ res/pjproject/build.symbian/pjlib.mmp (added),
+ res/pjproject/pjsip/src/test/test.c (added),
+ res/pjproject/third_party/portaudio/build (added),
+ res/pjproject/pjsip/src/test/test.h (added),
+ res/pjproject/pjsip/include/pjsip_auth.h (added),
+ res/pjproject/pjlib/src/pj/errno.c (added),
+ res/pjproject/third_party/BaseClasses/wxdebug.cpp (added),
+ res/pjproject/pjsip/include/pjsip-simple/rpid.h (added),
+ res/pjproject/pjlib/include/pj/compat/os_sunos.h (added),
+ res/pjproject/third_party/portaudio/install-sh (added),
+ res/pjproject/pjlib/src/pj/os_info.c (added),
+ res/pjproject/tests/pjsua/scripts-sipp/uas-reinv-no-media.xml
+ (added),
+ res/pjproject/tests/pjsua/scripts-sendto/172_timer_supported_but_not_used.py
+ (added),
+ res/pjproject/third_party/build/resample/libresample_dll.vcproj
+ (added), res/pjproject/pjmedia/include (added),
+ res/pjproject/third_party/portaudio/src/hostapi/asio/ASIO-README.txt
+ (added), res/pjproject/pjsip-apps/src/python/samples/presence.py
+ (added),
+ res/pjproject/build/vs/pjproject-vs8-debug-static-defaults.vsprops
+ (added), res/pjproject/pjmedia/src/pjmedia/transport_srtp.c
+ (added),
+ res/pjproject/pjmedia/include/pjmedia-codec/amr_sdp_match.h
+ (added), res/pjproject/pjsip/src/pjsip-simple/rpid.c (added),
+ res/pjproject/pjlib-util/src/pjlib-util/dns_server.c (added),
+ res/pjproject/tests/pjsua/runall.py (added),
+ res/pjproject/pjlib/include/pj/compat/m_armv4.h (added),
+ res/pjproject/pjsip/src/pjsip/sip_util_proxy.c (added),
+ res/pjproject/pjlib-util/include/pjlib-util/crc32.h (added),
+ res/pjproject/pjlib-util/build/os-auto.mak.in (added),
+ res/pjproject/tests/pjsua/scripts-sipp/uas-subscribe-multipart-notify.xml
+ (added), res/pjproject/pjlib/build/wince-evc4/pjlib_wince.vcp
+ (added), res/pjproject/pjmedia/include/pjmedia/sound.h (added),
+ res/pjproject/pjsip/build/output (added), res/pjproject/pjnath
+ (added), res/pjproject/INSTALL.txt (added),
+ res/pjproject/tests/pjsua/mod_call.py (added),
+ res/pjproject/pjlib/build/wince-evc4/pjlib_wince.vcw (added),
+ res/pjproject/pjsip/src/test/dlg_core_test.c (added),
+ res/pjproject/tests/pjsua/scripts-pesq/200_codec_g711u.py
+ (added),
+ res/pjproject/tests/pjsua/scripts-sendto/411_fmtp_amrnb_offer_band_eff.py
+ (added), res/pjproject/third_party/build/resample/config.h
+ (added), res/pjproject/pjsip-apps/src/pjsua_wince/pjsua_wince.rc
+ (added), res/pjproject/pjlib/build/output (added),
+ res/pjproject/pjlib/include/pj/compat/m_powerpc.h (added),
+ res/pjproject/pjsip/src/test/msg_logger.c (added),
+ res/pjproject/pjsip-apps/src/pjsua_wince/resource.h (added),
+ res/pjproject/pjsip/src/pjsip/sip_auth_parser_wrap.cpp (added),
+ res/pjproject/aconfigure.ac (added),
+ res/pjproject/tests/pjsua/scripts-sendto/140_sdp_with_direction_attr_in_session_1.py
+ (added),
+ res/pjproject/pjsip-apps/src/pjsystest/pjsystest_wince.rc2
+ (added), res/pjproject/pjlib/include/pj/compat/os_win32.h
+ (added), res/pjproject/pjmedia/include/pjmedia/doxygen.h (added),
+ res/pjproject/pjsip/src/test/main_rtems.c (added),
+ res/pjproject/pjlib-util/include/pjlib-util/scanner_cis_bitwise.h
+ (added), res/pjproject/pjsip-apps/src/ipjsystest/main.m (added),
+ res/pjproject/build.symbian/pjsip.mmp (added),
+ res/pjproject/third_party/speex/include/speex/speex_jitter.h
+ (added), res/pjproject/tests/pjsua/run.py (added),
+ res/pjproject/third_party/speex/symbian (added),
+ res/pjproject/tests/pjsua/scripts-pesq/200_codec_l16_8000_stereo.py
+ (added), res/pjproject/pjsip-apps/src/samples/auddemo.c (added),
+ res/pjproject/tests/pjsua/scripts-sendto/300_srtp_crypto_case_insensitive.py
+ (added), res/pjproject/third_party/g7221/common/basic_op.c
+ (added),
+ res/pjproject/pjnath/build/wince-evc4/pjnath_test_wince.vcp
+ (added), res/pjproject/third_party/g7221/common/basic_op.h
+ (added), res/pjproject/third_party/portaudio/config.guess
+ (added), res/pjproject/third_party/portaudio/src/os/unix (added),
+ res/pjproject/third_party/speex/libspeex/cb_search_sse.h (added),
+ res/pjproject/tests/pjsua/tools/Makefile (added),
+ res/pjproject/pjlib/src/pj/compat/longjmp_i386.S (added),
+ res/pjproject/third_party/portaudio/pablio (added),
+ res/pjproject/build.symbian/symbian_ua_udeb.pkg (added),
+ res/pjproject/README.txt (added),
+ res/pjproject/third_party/srtp/srtp.vcproj (added),
+ res/pjproject/pjnath/build (added),
+ res/pjproject/third_party/portaudio/src/hostapi/dsound (added),
+ res/pjproject/tests/automated/prepare.xml.template (added),
+ res/pjproject/pjsip/src/pjsua-lib/pjsua_pres.c (added),
+ res/pjproject/tests/pjsua/scripts-sipp/uas-reinv-and-ack(same-branch)-without-sdp.xml
+ (added), res/pjproject/pjlib/build (added),
+ res/pjproject/third_party/build/baseclasses/libbaseclasses.vcproj
+ (added),
+ res/pjproject/third_party/speex/include/speex/speex_preprocess.h
+ (added), res/pjproject/pjlib/src/pjlib-test (added),
+ res/pjproject/pjsip-apps/src/symbian_ua_gui/data/symbian_ua_gui.l01
+ (added), res/pjproject/pjlib/build/privkey.pem (added),
+ res/pjproject/pjmedia/src/pjmedia/alaw_ulaw_table.c (added),
+ res/pjproject/configure-legacy (added),
+ res/pjproject/tests/pjsua/scripts-sendto/200_ice_success_1.py
+ (added), res/pjproject/pjsip/include/pjsip/sip_transport.h
+ (added), res/pjproject/pjnath/src/pjturn-srv/server.c (added),
+ res/pjproject/pjmedia/build/os-linux.mak (added),
+ res/pjproject/pjlib/include/pj/compat/os_win32_wince.h (added),
+ res/pjproject/pjsip/src/pjsip-ua/sip_replaces.c (added),
+ res/pjproject/third_party/portaudio/src/common/pa_util.h (added),
+ res/pjproject/pjsip-apps/src/symbian_ua_gui/inc/symbian_ua_guiDocument.h
+ (added), res/pjproject/pjlib/src/pj/fifobuf.c (added),
+ res/pjproject/third_party/gsm/tls/sour1.dta (added),
+ res/pjproject/pjsip/include/pjsip/sip_types.h (added),
+ res/pjproject/pjlib/include/pj/compat/time.h (added),
+ res/pjproject/pjsip/src/pjsip/sip_auth_msg.c (added),
+ res/pjproject/tests/pjsua/scripts-sendto/001_torture_4475_3_1_1_1.py
+ (added), res/pjproject/pjsip/include/pjsip_ua.h (added),
+ res/pjproject/pjlib/build/Makefile (added),
+ res/pjproject/third_party/srtp/README (added),
+ res/pjproject/tests/pjsua/scripts-sendto/311_srtp1_recv_avp.py
+ (added), res/pjproject/pjsip-apps/src/pjsua/main_rtems.c (added),
+ res/pjproject/pjsip-apps/src/pocketpj/res/invisibl.bmp (added),
+ res/pjproject/pjlib/src/pjlib-test/rtems_network_config.h
+ (added), res/pjproject/third_party/srtp/crypto/math/stat.c
+ (added), res/pjproject/third_party/srtp/test/replay_driver.c
+ (added), res/pjproject/pjmedia/src/pjmedia-audiodev/audiotest.c
+ (added), res/pjproject/pjlib/src/pjlib++-test (added),
+ res/pjproject/pjsip-apps/src/samples/streamutil.c (added),
+ res/pjproject/pjmedia/src/pjmedia/ffmpeg_util.c (added),
+ res/pjproject/tests/pjsua/scripts-sendto/500_pres_subscribe_with_bad_event.py
+ (added), res/pjproject/third_party/srtp/install-sh (added),
+ res/pjproject/tests/pjsua/scripts-pesq/200_codec_speex_16000.py
+ (added),
+ res/pjproject/third_party/srtp/crypto/cipher/null_cipher.c
+ (added), res/pjproject/pjmedia/src/pjmedia/ffmpeg_util.h (added),
+ res/pjproject/pjlib-util/src (added),
+ res/pjproject/pjsip/include/pjsip/sip_config.h (added),
+ res/pjproject/pjlib/docs/doxygen.cfg (added): Add support for
+ ICE/STUN/TURN in res_rtp_asterisk and chan_sip. Review:
+ https://reviewboard.asterisk.org/r/1891/
+
+2012-06-29 20:32 +0000 [r369512] Mark Michelson <mmichelson@digium.com>
+
+ * main/rtp_engine.c, /: Fix apparent copy and paste error where
+ incorrect "glue" is used. ........ Merged revisions 369511 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-29 17:02 +0000 [r369493] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, main/channel.c, main/autoservice.c, main/pbx.c,
+ channels/chan_local.c, funcs/func_channel.c,
+ main/channel_internal_api.c, main/features.c,
+ configs/cdr.conf.sample, include/asterisk/channel.h,
+ include/asterisk/pbx.h, CHANGES, apps/app_followme.c,
+ apps/app_queue.c: Hangup handlers - Dialplan subroutines that run
+ when the channel hangs up. Hangup handlers are an alternative to
+ the h extension. They can be used in addition to the h extension.
+ The idea is to attach a Gosub routine to a channel that will
+ execute when the call hangs up. Whereas which h extension gets
+ executed depends on the location of dialplan execution when the
+ call hangs up, hangup handlers are attached to the call channel.
+ You can attach multiple handlers that will execute in the order
+ of most recently added first. (closes issue ASTERISK-19549)
+ Reported by: Mark Murawski Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/2002/
+
+2012-06-29 16:56 +0000 [r369492] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: With some configurations a transport is
+ not actually specified so assume UDP in these cases. ........
+ Merged revisions 369490 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369491 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-29 16:42 +0000 [r369489] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel_internal_api.c, .cleancount: Remove obsolete struct
+ ast_channel note. The opaquing the ast_channel struct no longer
+ requires .cleancount to be changed when the struct is changed. *
+ Bump .cleancount value one last time because of struct
+ ast_channel for old times sake.
+
+2012-06-29 15:33 +0000 [r369473] Joshua Colp <jcolp@digium.com>
+
+ * /, 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/ ........
+ Merged revisions 369471 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369472 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-28 01:12 +0000 [r369449-369454] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/config_options.h,
+ configs/config_test.conf.sample, main/config_options.c,
+ tests/test_config.c: Add the ability to set flags via the config
+ options api Allows the setting of flags via the config options
+ api. For example, code like this: #define OPT1 1 << 0 #define
+ OPT2 1 << 1 #define OPT3 1 << 2 struct thing { unsigned int
+ flags; }; and a config like this: [blah] opt1=yes opt2=no
+ opt3=yes Review: https://reviewboard.asterisk.org/r/2004/
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: 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 ........ Merged revisions 369436 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369437 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-26 21:45 +0000 [r369414] Jonathan Rose <jrose@digium.com>
+
+ * include/asterisk/logger.h, channels/chan_dahdi.c,
+ main/autoservice.c, main/pbx.c, channels/chan_local.c,
+ channels/sig_analog.c, main/channel_internal_api.c,
+ channels/chan_agent.c, main/features.c, main/logger.c,
+ channels/chan_iax2.c, channels/sig_pri.c, channels/sig_ss7.c,
+ main/bridging.c, main/cli.c: Unique Call ID logging Phases III
+ and IV Adds call ID logging changes to specific channel drivers
+ that weren't handled handled in phase II of Call ID Logging. Also
+ covers logging for threads for threads created by systems that
+ may be involved with many different calls. Extra special thanks
+ to Richard for rigorous review of chan_dahdi and its various
+ signalling modules. review:
+ https://reviewboard.asterisk.org/r/1927/ review:
+ https://reviewboard.asterisk.org/r/1950/
+
+2012-06-26 13:23 +0000 [r369370-369392] Matthew Jordan <mjordan@digium.com>
+
+ * /, main/adsi.c: Fix crash in unloading of res_adsi module When
+ res_adsi is unloaded, it removes the ADSI functions that it
+ previously installed by passing a NULL adsi_funcs pointer to
+ ast_adsi_install_funcs. This function was not checking whether or
+ not the adsi_funcs pointer passed in was NULL before
+ dereferencing it to check whether or not the version of the
+ functions matches what the core was expecting it. This patch
+ makes it so that the version is only checked if a potentially
+ valid adsi_funcs pointer was passed in. Passing in NULL removes
+ the installed functions, bypassing the version check. ........
+ Merged revisions 369390 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369391 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/manager.c: Update "manager show event" to support tab
+ completion Thank you rmudgett for pointing out that I was missing
+ this in the initial check-in for AMI event documentation
+ (r369346)
+
+ * 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/ ........ Merged
+ revisions 369351 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369369 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-25 19:26 +0000 [r369367] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: 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 ........ Merged revisions
+ 369352 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 369353 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-25 17:59 +0000 [r369346] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_dial.c, apps/app_meetme.c, configure.ac,
+ apps/app_userevent.c, CHANGES, apps/app_queue.c, Makefile,
+ build_tools/get_documentation.py (added), main/manager.c,
+ configure, build_tools/post_process_documentation.py (added),
+ include/asterisk/xmldoc.h, apps/app_confbridge.c, makeopts.in,
+ apps/app_stack.c, apps/app_chanspy.c, doc/appdocsxml.dtd,
+ main/xmldoc.c, apps/app_voicemail.c: Add AMI event documentation
+ This patch adds the core changes necessary to support AMI event
+ documentation in the source files of Asterisk, and adds
+ documentation to those AMI events defined in the core application
+ modules. Event documentation is built from the source by two new
+ python scripts, located in build_tools: get_documentation.py and
+ post_process_documentation.py. The get_documentation.py script
+ mirrors the actions of the existing AWK get_documentation
+ scripts, except that it will scan the entirety of a source file
+ for Asterisk documentation. Upon encountering it, if the
+ documentation happens to be an AMI event, it will attempt to
+ extract information about the event directly from the manager
+ event macro calls that raise the event. The
+ post_process_documentation.py script combines manager event
+ instances that are the same event but documented in multiple
+ source files. It generates the final core-[lang].xml file. As
+ this process can take longer to complete than a typical 'make
+ all', it is only performed if a new make target, 'full', is
+ chosen. Review: https://reviewboard.asterisk.org/r/1967/
+
+2012-06-25 16:07 +0000 [r369329] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Fix Bridge application occasionally returning
+ to the wrong location. * Fix do_bridge_masquerade() getting the
+ resume location from the zombie channel. The code must not touch
+ a clone channel after it has masqueraded it. The clone channel
+ has become a zombie and is starting to hangup. (closes issue
+ ASTERISK-19985) Reported by: jamicque Patches:
+ jira_asterisk_19985_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: jamicque ........ Merged revisions 369327
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 369328 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-25 15:55 +0000 [r369304-369326] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/adsi.h, /, main/Makefile, res/res_adsi.c,
+ main/adsi.c (added), res/res_adsi.exports.in (removed): Multiple
+ revisions 369323-369324 ........ r369323 | mmichelson |
+ 2012-06-25 10:35:43 -0500 (Mon, 25 Jun 2012) | 9 lines 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 ........ r369324 |
+ mmichelson | 2012-06-25 10:50:17 -0500 (Mon, 25 Jun 2012) | 2
+ lines Forgot to svn add this file in my last commit. ........
+ Merged revisions 369323-369324 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369325 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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) ........ Merged revisions 369302 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369303 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-23 00:33 +0000 [r369237-369296] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: Fix F and F(x) action logic in Bridge
+ application.
+
+ * /, 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. ........ Merged
+ revisions 369282 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369283 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, apps/app_queue.c: Explicitly check caller hangup in app Queue
+ rather than a polluted res2 value. ........ Merged revisions
+ 369262 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 369263 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * apps/app_queue.c: Fix F and F(x) action logic in Queue
+ application.
+
+ * apps/app_dial.c, /: Check if PBX was started and fix F and F(x)
+ action logic in Dial application. ........ Merged revisions
+ 369258 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 369259 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, main/ccss.c: Check if PBX was started for generic CCSS recall.
+ ........ Merged revisions 369238 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369239 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, channels/chan_sip.c: Change incorrect chan_sip zombie hangup
+ debug message. They are all zombies now. ........ Merged
+ revisions 369235 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369236 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-22 20:05 +0000 [r369217] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Don't crash on a guest directmedia call A
+ sip_pvt may not have relatedpeer set if a call doesn't match up
+ with a peer. If there is no relatedpeer, there is no direct media
+ ACL to apply, so just return that it is allowed. (closes issue
+ ASTERISK-20040) Reported by: Terry Wilson ........ Merged
+ revisions 369214 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369215 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-22 19:54 +0000 [r369184-369216] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_dahdi.c: Fix wrong variable name in the R2
+ disconnect callback
+
+ * /, 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. ........ Merged revisions
+ 369195 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 369206 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/chan_dahdi.c: Add HANGUPCAUSE hash implementation for
+ DAHDI MFC/R2 subtech This adds a minimal implementation of the
+ "Who Hung Up?" Asterisk 11 work to chan_dahdi.c for the MFC/R2
+ DAHDI subtech. Given the way that OpenR2 interfaces with
+ chan_dahdi, it is much harder to expose the type of protocol
+ information that is available in PRI, SS7, or other channel
+ technologies.
+
+ * channels/sig_analog.c, channels/sig_pri.c: Add HANGUPCAUSE hash
+ support for analog and PRI DAHDI subtechs This is part of the
+ DAHDI support for the Asterisk 11 "Who Hung Up?" project and
+ covers the implementation for the technologies implemented in
+ sig_analog.c and sig_pri.c. Tested on a local machine to verify
+ protocol and cause information is available. Review:
+ https://reviewboard.asterisk.org/r/1953/ (issue SWP-4222)
+
+ * channels/sig_ss7.c: Add "Who Hung Up?" implementation for DAHDI
+ SS7 subtechnology Testing was done on a local machine to verify
+ that protocol and cause information was being sent properly.
+ Review: https://reviewboard.asterisk.org/r/1955/ (issue SWP-4222)
+
+2012-06-20 21:33 +0000 [r369166-369167] Richard Mudgett <rmudgett@digium.com>
+
+ * main/logger.c: Don't waste time initializing the whole
+ call_identifer_str[]. The array is either setup with a callid
+ string or only the first element needs to be initialized.
+
+ * channels/chan_misdn.c: Fix chan_misdn compile error.
+
+2012-06-20 17:48 +0000 [r369148] Alexandr Anikin <may@telecom-service.ru>
+
+ * /, addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c: fix
+ locking issue on empty callList (issue ASTERISK-19298) Reported
+ by: Dmitry Melekhov Patches: ASTERISK-18322-2.patch ........
+ Merged revisions 369146 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369147 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-20 11:47 +0000 [r369142] Sean Bright <sean@malleable.com>
+
+ * apps/app_externalivr.c: Remove declaration of eivr_connect_socket
+ because it no longer exists.
+
+2012-06-20 11:20 +0000 [r369141] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c: use right definition for channel name
+
+2012-06-20 03:18 +0000 [r369110-369126] Michael L. Young <elgueromexicano@gmail.com>
+
+ * main/manager.c, CHANGES: Add IPv6 Support To Manager This patch
+ adds IPv6 support to AMI. (Closes issue ASTERISK-19965) Reported
+ by: Michael L. Young Tested by: Michael L. Young Patches:
+ ami_ipv6_v3.diff uploaded by Michael L. Young (license 5026)
+ Review: https://reviewboard.asterisk.org/r/1968/
+
+ * main/netsock2.c, /, include/asterisk/netsock2.h: 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/ ........ Merged
+ revisions 369108 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369109 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-19 23:36 +0000 [r369092] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c, /: check rtptimeouts in ooh323 channels as
+ per config file (rtp voice, video, udptl except rtcp) (closes
+ issue ASTERISK-19179) Reported by: TSAREGORODTSEV Yury Patches:
+ 19179-ooh323-ast10.patch ........ Merged revisions 369091 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-19 21:13 +0000 [r369086] Kinsey Moore <kmoore@digium.com>
+
+ * main/channel.c, channels/chan_dahdi.c, channels/chan_misdn.c,
+ main/rtp_engine.c, include/asterisk/channel.h,
+ channels/chan_iax2.c: Ensure that pvt cause information does not
+ break native bridging Channel drivers that allow native bridging
+ need to handle AST_CONTROL_PVT_CAUSE_CODE frames and previously
+ did not handle them properly, usually breaking out of the native
+ bridge. This change corrects that behavior and exposes the
+ available cause code information to the dialplan while native
+ bridges are in place. This required exposing the HANGUPCAUSE hash
+ setter outside of channel.c, so additional documentation has been
+ added.
+
+2012-06-19 15:44 +0000 [r369068] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Fix request routing issue when
+ outboundproxy is used. Asterisk was incorrectly setting the
+ destination of CANCELs and ACKs for error responses to the URI of
+ the initial INVITE. This resulted in further requests, such as
+ INVITEs with authentication credentials, to be routed
+ incorrectly. Instead, when these CANCEL or ACKs are to be sent,
+ we should simply keep the destination the same as what it
+ previously was. There is no need to alter it any. (closes issue
+ ASTERISK-20008) Reported by Marcus Hunger Patches:
+ ASTERISK-20008.patch uploaded by Mark Michelson (license #5049)
+ ........ Merged revisions 369066 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369067 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-18 22:56 +0000 [r369061] Kinsey Moore <kmoore@digium.com>
+
+ * main/features.c: Fix AST_CONTROL_PVT_CAUSE_CODE handling When the
+ IAX2 Who Hung Up? changes were added, they uncovered a bug in the
+ way AST_CONTROL_PVT_CAUSE_CODE was handled in
+ feature_request_and_dial(). This particular frame subtype was
+ being treated like more terminal control frames causing the
+ function to be exited prematurely.
+
+2012-06-18 18:25 +0000 [r369057] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Fix monitoring calls put in a parking lot. *
+ Fix a regression that was introduced by -r366167 which
+ effectively disabled monitoring parked calls. (closes issue
+ ASTERISK-20012) Reported by: sdolloff Tested by: rmudgett
+ ........ Merged revisions 369043 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 369044 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-15 21:18 +0000 [r369034] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Various small chan_skinny fixes and
+ cleanup Added test to skinny_register to only allow device to
+ register against a device that is not already registered. Addback
+ l->device test for skinny_show_lines. Fixes segfault if a line is
+ configured but not configured to a device. Reverses part of
+ r368680. Removed redundant l->device tests in subsubstate and
+ dumpsub. l->device will always be valid if these routines are
+ called. Reverses 368948 - discussed with mjordan on irc. Some
+ indentation cleanup.
+
+2012-06-15 17:13 +0000 [r369028] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c, channels/sip/include/sip.h: Allow chan_sip
+ to decline unwanted media streams This change replaces the static
+ array of four representable media streams with an AST_LIST so
+ that chan_sip can keep track of offered media streams. This
+ allows chan_sip to deal with offers containing multiple same-type
+ streams and many other situations without rejecting the SDP offer
+ in its entirety, yet still generating a valid response. This also
+ covers cases where Asterisk can not comprehend the offer if it is
+ in the correct format. Previously, chan_sip would reject SDP
+ offers or entirely ignore individual stream offers in an effort
+ to be more compatible which would often result in invalid SDP
+ responses. Review: https://reviewboard.asterisk.org/r/1988/
+
+2012-06-15 16:30 +0000 [r369027] Jason Parker <jparker@digium.com>
+
+ * /, apps/app_voicemail.c: Fix voicemail API tests by using the
+ correct argument order for create/destroy. ........ Merged
+ revisions 369024 from
+ http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
+ ........ Merged revisions 369026 from
+ http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones
+
+2012-06-15 16:20 +0000 [r369013] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/format.c, main/udptl.c, main/netsock2.c, main/autoservice.c,
+ main/rtp_engine.c, main/frame.c, main/security_events.c, /,
+ main/say.c, main/threadstorage.c, channels/console_video.c,
+ main/devicestate.c, main/astfd.c, main/taskprocessor.c,
+ main/format_pref.c, main/astobj2.c, main/indications.c,
+ main/config.c, main/loader.c, main/term.c,
+ apps/confbridge/conf_config_parser.c, main/cli.c,
+ channels/sig_analog.c, main/framehook.c, main/strcompat.c,
+ main/plc.c, main/fskmodem_int.c, main/syslog.c,
+ main/stdtime/localtime.c, main/bridging.c, main/db.c,
+ channels/sig_ss7.c, main/datastore.c, main/sched.c,
+ channels/sip/sdp_crypto.c, main/strings.c, main/pbx.c,
+ channels/vcodecs.c, channels/sip/security_events.c,
+ main/libasteriskssl.c, channels/iax2-provision.c,
+ pbx/dundi-parser.c, main/aoc.c, main/cel.c, utils/astdb2bdb.c,
+ channels/iax2-parser.c, main/chanvars.c, main/netsock.c,
+ build_tools/find_missing_support_level (added), main/data.c,
+ main/srv.c, channels/chan_misdn.c, main/privacy.c,
+ main/fixedjitterbuf.c, channels/sip/dialplan_functions.c,
+ main/test.c, main/audiohook.c, codecs/codec_dahdi.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, main/xml.c,
+ channels/misdn/isdn_msg_parser.c, main/utils.c, main/autochan.c,
+ channels/misdn/isdn_lib.c, main/enum.c, main/presencestate.c,
+ main/fskmodem.c, channels/misdn_config.c, main/io.c,
+ main/channel.c, main/cdr.c, res/ael/pval.c, main/ulaw.c,
+ main/dial.c, main/format_cap.c, main/tdd.c,
+ channels/console_gui.c, main/heap.c, channels/misdn/ie.c,
+ main/logger.c, main/app.c, channels/console_board.c,
+ main/image.c, main/message.c, main/dns.c, main/lock.c,
+ main/stun.c, channels/sip/srtp.c, main/dnsmgr.c,
+ main/slinfactory.c, main/channel_internal_api.c,
+ main/translate.c, main/jitterbuf.c, main/acl.c,
+ utils/astdb2sqlite3.c, channels/sip/utils.c, channels/sig_pri.c,
+ apps/app_system.c, funcs/func_realtime.c, main/tcptls.c,
+ main/hashtab.c, funcs/func_presencestate.c,
+ apps/app_celgenuserevent.c, main/abstract_jb.c, main/callerid.c,
+ main/file.c, main/config_options.c, res/snmp/agent.c,
+ main/astmm.c, main/event.c, channels/misdn/portinfo.c,
+ channels/sip/config_parser.c, channels/vgrabbers.c, main/dsp.c,
+ main/xmldoc.c: Multiple revisions 369001-369002 ........ r369001
+ | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11
+ lines 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. ........ r369002 |
+ kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3
+ lines Add a script to enable finding source files without
+ support-levels defined. ........ Merged revisions 369001-369002
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 369005 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-15 16:17 +0000 [r369007] Kinsey Moore <kmoore@digium.com>
+
+ * main/frame.c, channels/chan_iax2.c, include/asterisk/frame.h: Add
+ HANGUPCAUSE hash support to IAX2 Continuing with the Who Hung Up?
+ project for Asterisk 11, this adds support to IAX2 for the
+ HANGUPCAUSE hash. Additionally, this breaks out some
+ functionality in frame.c for getting information about frame
+ types and subclasses. Review:
+ https://reviewboard.asterisk.org/r/1941/ (issue SWP-4222)
+
+2012-06-15 15:33 +0000 [r369000] Jason Parker <jparker@digium.com>
+
+ * /, apps/app_voicemail.exports.in: Remove some symbol exports that
+ got missed in the removal of global symbols. (issue AST-807)
+ (issue AST-901) (issue AST-908) ........ Merged revisions 368998
+ from
+ http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
+ ........ Merged revisions 368999 from
+ http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones
+
+2012-06-15 00:55 +0000 [r368972-368991] Richard Mudgett <rmudgett@digium.com>
+
+ * /: Remove remaining properties mmichelson left laying around from
+ phones branch merge.
+
+ * apps/app_dial.c, main/channel.c, include/asterisk/app.h,
+ main/ccss.c, main/app.c, apps/app_followme.c, apps/app_queue.c,
+ apps/app_stack.c: Allow non-normal execution routines to be able
+ to run on hungup channels. * Make non-normal dialplan execution
+ routines be able to run on a hung up channel. This is preparation
+ work for hangup handler routines. * Fixed ability to support
+ relative non-normal dialplan execution routines. (i.e., The
+ context and exten are optional for the specified dialplan
+ location.) Predial routines are the only non-normal routines that
+ it makes sense to optionally omit the context and exten. Setting
+ a hangup handler also needs this ability. * Fix Return
+ application being able to restore a dialplan location exactly.
+ Channels without a PBX may not have context or exten set. * Fixes
+ non-normal execution routines like connected line interception
+ and predial leaving the dialplan execution stack unbalanced.
+ Errors like missing Return statements, popping too many stack
+ frames using StackPop, or an application returning non-zero could
+ leave the dialplan stack unbalanced. * Fixed the AGI gosub
+ application so it cleans up the dialplan execution stack and
+ handles the autoloop priority increments correctly. * Eliminated
+ the need for the gosub_virtual_context return location. Review:
+ https://reviewboard.asterisk.org/r/1984/
+
+ * main/pbx.c: Make the Hangup application set a softhangup flag.
+ The Hangup application used to just return -1 to cause normal
+ dialplan execution to hangup a channel. For the non-normal
+ execution routines like predial and connected-line interception
+ routines, the hangup request would exit the routine early but
+ otherwise be ignored. * Made the Hangup application not allow
+ setting a cause code of zero. A zero cause code is not defined.
+
+ * include/asterisk/app.h: Move vm defines to group them better.
+
+2012-06-14 19:40 +0000 [r368966] Jason Parker <jparker@digium.com>
+
+ * include/asterisk/app.h, /, tests/test_voicemail_api.c,
+ main/app.c, include/asterisk/app_voicemail.h (removed),
+ apps/app_voicemail.c: Multiple revisions 368963,368965 ........
+ r368963 | qwell | 2012-06-14 13:47:03 -0500 (Thu, 14 Jun 2012) |
+ 14 lines Remove global symbol requirement from app_voicemail.
+ This uses the existing "function installation" stuff that already
+ existed for other functions, like getting message counts. (closes
+ issue AST-807) (issue AST-901) (issue AST-908) Review:
+ https://reviewboard.asterisk.org/r/1965/ ........ Merged
+ revisions 368962 from
+ http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
+ ........ r368965 | qwell | 2012-06-14 14:04:57 -0500 (Thu, 14 Jun
+ 2012) | 11 lines These functions that were moved need to be
+ static. Also wrap test functions in a #ifdef. (issue AST-807)
+ (issue AST-901) (issue AST-908) ........ Merged revisions 368964
+ from
+ http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
+ ........ Merged revisions 368963,368965 from
+ http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones
+
+2012-06-14 17:34 +0000 [r368948] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_skinny.c: AST-2012-009: Fix crash in chan_skinny
+ due to Key Pad Button Message handling AST-2012-008 (r367844)
+ fixed a denial of service attack exploitable in the Skinny
+ channel driver that occurred when certain messages are sent after
+ a previously registered station sends an Off Hook message.
+ Unresolved in that patch is an issue in the Asterisk 10 releases,
+ wherein, if a Station Key Pad Button Message is processed after
+ an Off Hook message, the channel driver will inappropriately
+ dereference a NULL pointer. This patch fixes those places where
+ the message handling or the channel callback functions would
+ attempt to dereference the line's pointer to the device. (issue
+ ASTERISK-19905) Reported by: Christoph Hebeisen Tested by:
+ mjordan, Christoph Hebeisen Patches: AST-2012-009-10.diff
+ uploaded by mjordan (license 6283) ........ Merged revisions
+ 368947 from http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-14 15:28 +0000 [r368929] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/Makefile: Revert Makefile change to remove embedding
+ res_adsi.so The change has resulted in a linking error for
+ certain versions of GCC. This is much worse than the original
+ issue, so for now, temporarily revert the change. A more thorough
+ change will be sought out. ........ Merged revisions 368927 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368928 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-14 13:41 +0000 [r368920-368921] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/config_options.h, main/config_options.c: Add a
+ post_apply callback to the Config Options API This adds a
+ callback that only fires when changes have been successfully
+ applied via the Config Options API. Review:
+ https://reviewboard.asterisk.org/r/1980/
+
+ * include/asterisk/config_options.h, main/config_options.c: Add
+ filename alias support to the Config Options API This adds the
+ ability to handle a single filename alias for a config file. This
+ is useful if a config filename has changed, but the old filename
+ should be supported for backwards compatibility. Review:
+ https://reviewboard.asterisk.org/r/1981/
+
+2012-06-13 21:17 +0000 [r368900] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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) ........ Merged revisions 368898 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368899 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-13 20:28 +0000 [r368896] Matthew Jordan <mjordan@digium.com>
+
+ * res/res_smdi.c, /, res/res_adsi.c: Mark res_smdi/res_adsi as
+ 'core' supported modules Recently, various issues surrounding
+ weak symbols have caused problems with modules that rely on that
+ feature to be enabled in menuselect. This includes app_voicemail
+ and chan_dahdi, as they both rely upon res_smdi and res_adsi,
+ which, in certain circumstances, may not be enabled by default in
+ menuselect. Because res_smdi/res_adsi are dependencies for
+ chan_dahdi/app_voicemail, this patch marks both as 'core'
+ supported modules. This will allow both app_voicemail and
+ chan_dahdi to be enabled as well, regardless of whether or not
+ that system supports weak symbols. (issue AST-900) Reported by:
+ Thomas Arimont (issue AST-885) Reported by: Denis Alberto
+ Martinez ........ Merged revisions 368894 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368895 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-13 19:51 +0000 [r368886] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/Makefile: Remove forced linking of res_adsi.o In GCC 4.5+
+ the result is that Asterisk has a phantom module loaded at
+ startup, claiming to be res_adsi. (closes issue ASTERISK-19920)
+ reported by Leif Madsen ........ Merged revisions 368873 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368885 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-13 14:55 +0000 [r368832-368855] Matthew Jordan <mjordan@digium.com>
+
+ * Makefile: Replace MODULES_DIR with ASTMODDIR in Makefile's
+ INSTALLDIRS Post Asterisk 10, the MODULES_DIR variable no longer
+ exists, and was replaced with ASTMODDIR.
+
+ * 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. ........
+ Merged revisions 368852 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368853 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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) ........ Merged revisions 368830 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368831 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-12 15:46 +0000 [r368809] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Set the Caller ID "tag" on peers even if
+ remote party information is present. On incoming calls, we were
+ setting the cid_tag on the dialog only if there was no remote
+ party information (Remote-Party-ID or P-Asserted-Identity)
+ present. The Caller ID tag is an invented parameter, though, and
+ should be set no matter the circumstance. (closes issue
+ ASTERISK-19859) Reported by Thomas Arimont (closes issue AST-884)
+ Reported by Trey Blancher ........ Merged revisions 368807 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368808 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-12 14:09 +0000 [r368793-368794] Matthew Jordan <mjordan@digium.com>
+
+ * /: Update merge property information
+
+ * channels/chan_sip.c: Fix deadlock in SIP transfers that involve a
+ REFER request In r367163, "send to voicemail" functionality was
+ added to the SIP channel driver. This required updating the party
+ redirecting information for the channel based on the headers
+ provided in the REFER request. When the redirecting party
+ information is updated on the channel, a call to
+ ast_indicate_data occurs. Because handle_request_refer still had
+ the sip_pvt locked, a deadlock could occur between the pbx_thread
+ and the do_monitor thread servicing the REFER request. This patch
+ preserves the proper locking order between the channel and the
+ sip_pvt by ensuring that the sip_pvt is unlocked prior to
+ updating the party redirecting information on the channel.
+ (closes issue AST-903) Reported by: Matt Jordan patches:
+ jira_ast_903_trunk.patch by rmudgett (license 5621)
+
+2012-06-12 04:03 +0000 [r368784] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c, UPGRADE.txt: Parse ANI2 information from SIP
+ From header parameters ANI2 information is now parsed out of SIP
+ From headers when present in the oli, isup-oli, and ss7-oli
+ parameters and is available via the CALLERID(ani2) dialplan
+ function. (closes issue ASTERISK-19912) Patch-by: Rob Gagnon
+ Review: https://reviewboard.asterisk.org/r/1947/
+
+2012-06-11 17:34 +0000 [r368772] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/chan_sip.c, include/asterisk/channel.h,
+ channels/chan_iax2.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 ........ Merged revisions 368759 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368760 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-11 15:23 +0000 [r368722-368751] Kinsey Moore <kmoore@digium.com>
+
+ * channels/sip/sdp_crypto.c, /, channels/chan_sip.c, main/say.c,
+ res/res_fax.c, channels/sip/reqresp_parser.c, apps/app_queue.c,
+ main/loader.c, channels/chan_dahdi.c, res/res_config_odbc.c,
+ channels/sip/dialplan_functions.c, apps/app_directory.c,
+ pbx/pbx_config.c, res/res_odbc.c, res/res_speech.c,
+ apps/app_voicemail.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)
+ ........ Merged revisions 368738 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368739 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /: Recorded merge of revisions 368721 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ 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. ........ Merged revisions 368719 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-06-08 21:08 +0000 [r368712-368714] Richard Mudgett <rmudgett@digium.com>
+
+ * main/manager.c, main/utils.c, include/asterisk/strings.h: Fix
+ error paths in action_hangup() for AMI Hangup action. * Check
+ allocation function return values for failure. Crashing is bad. *
+ Tweak ast_regex_string_to_regex_pattern() parameters for proper
+ ast_str usage.
+
+ * main/channel.c, include/asterisk/channel.h: Tweak
+ ast_channel_softhangup_withcause_locked() to take a typed
+ parameter.
+
+2012-06-08 08:32 +0000 [r368688] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * channels/chan_unistim.c: Fix MWI update so LED display correct
+ voicemail state after phone usage. Also fixes few warnings.
+ (closes issue #19675) Reported by: dbohling Patches: fixmwi.patch
+ uploaded by dbohling (license 6378)
+
+2012-06-07 21:44 +0000 [r368680-368681] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Skinny cleanup (mwi_event_cb). Original
+ was testing for d->session, setting and testing again (all
+ nested). Removed duplicate testing and restructured function to
+ test/return and then the main code.
+
+ * channels/chan_skinny.c: Skinny cleanup. Removed d->registered
+ which was mirroring d->session. Changed relevant references to
+ use d->session instead. Moved setting and unsetting of l->device
+ from session register to device configuration. As such, l->device
+ will always be valid unless it is has not been configured to a
+ device. Revised various test where checking if a device is
+ registered to use l->device->session.
+
+2012-06-07 20:39 +0000 [r368674-368675] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_queue.c: Fix app_queue debug message use of args.options
+ after the string has been parsed.
+
+ * apps/app_queue.c: Fix inverted test in app_queue for ringinuse.
+ Regression from -r367080 ringinuse commit. (issue ASTERISK-19536)
+
+2012-06-07 20:32 +0000 [r368673] Terry Wilson <twilson@digium.com>
+
+ * main/udptl.c, include/asterisk/config_options.h, apps/app_skel.c,
+ main/config_options.c, tests/test_config.c: Fix reloading an
+ unchanged file with the Config Options API Adding multiple file
+ support broke reloading an unchanged file. This adds an enum for
+ return values for the aco_process_* functions and ensures that
+ the config is not applied if res is not ACO_PROCESS_OK. Review:
+ https://reviewboard.asterisk.org/r/1979/
+
+2012-06-07 20:00 +0000 [r368668] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * formats/format_ogg_vorbis.c: Fix a typo in format_ogg_vorbis.c:
+ suport Review: https://reviewboard.asterisk.org/r/1970/
+
+2012-06-07 15:43 +0000 [r368663] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/config_options.h, main/config_options.c,
+ tests/test_config.c: Add default handler documentation and
+ standardize acl handler Added documentation describing what flags
+ and arguments to pass to aco_option_register for default option
+ types. Also changed the ACL handler to use the flags parameter to
+ differentiate between "permit" and "deny" instead of adding an
+ additional vararg parameter. Review:
+ https://reviewboard.asterisk.org/r/1969/
+
+2012-06-06 21:34 +0000 [r368646] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: Fix POTS flash
+ hook to orignate a second call deadlock. A deadlock can occur
+ when a POTS phone tries to flash hook to originate a second call
+ for 3-way or transfer. If another process is scanning the
+ channels container when the POTS line flash hooks then a deadlock
+ will occur. * Release the channel and private locks when creating
+ a new channel as a result of a flash hook. (closes issue
+ ASTERISK-19842) Reported by: rmudgett Tested by: rmudgett
+ ........ Merged revisions 368644 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368645 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-06 19:25 +0000 [r368637] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Fix a specific scenario where ACKs are
+ not matched. If a dialog-starting INVITE contains a to-tag, then
+ Asterisk will respond with a 481. In this case, the resulting
+ incoming ACK would not be matched, so Asterisk would continue
+ retransmitting the 481 until the transaction times out. There
+ were two issues. Asterisk, upon creating a sip_pvt would generate
+ a local tag. However, when the time came to transmit the 481,
+ since there was a to-tag in the INVITE, Asterisk would place this
+ original to-tag in the 481 response. When the ACK came in,
+ Asterisk would attempt to match the to-tag in the ACK to the
+ generated local tag. Unfortunately, Asterisk never actually
+ transmitted a response with the generated local tag, so the
+ to-tag in the ACK would not match. The other problem was that
+ when the 481 was sent, nothing was set on the sip_pvt to indicate
+ what CSeq is expected in the ACK. To fix the first problem, we
+ zero out the to-tag seen in the incoming INVITE. This way,
+ Asterisk, when time to send a response, will send its generated
+ local tag instead. To fix the second problem, we set the
+ sip_pvt's pendinginvite to the CSeq of the INVITE when we send a
+ 481. (closes issue ASTERISK-19892) Reported by Mark Michelson
+ ........ Merged revisions 368625 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368629 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-06 17:22 +0000 [r368606] Matthew Jordan <mjordan@digium.com>
+
+ * /, build_tools/make_version: Add feature modifier to versions
+ produced from branches Certain branches, such as Certified
+ Asterisk, may have a modifier added to them that specifies the
+ features available in that branch. For branches, this modifier is
+ expected to be reflected in the location of the branch in
+ subversion. For example, a subversion of URL of
+ /certified/branches/1.8.11 would have a feature modifier of
+ 'certified'. This is slightly different then how features are
+ determined for tags, where the feature is part of the actual tag
+ name, e.g., "10.5.0-digiumphones". In keeping with the
+ nomenclature used for tags, the feature specifier for branches is
+ translated and placed after the revision numbers. For the example
+ given previously, this would result in a branch version of
+ "Asterisk SVN-branch-1.8.11-cert-rXXXXXX". ........ Merged
+ revisions 368604 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368605 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-06 16:11 +0000 [r368588] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Ensure overlapping hold flags do not
+ conflict When changing between different modes of hold, the flags
+ were not being cleared out properly causing a failure to change
+ hold states. (closes issue ASTERISK-19919) Patch-by: Morten
+ Tryfoss Reported-by: Morten Tryfoss ........ Merged revisions
+ 368586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 368587 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-06 01:11 +0000 [r368566-368569] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Fix parked call performing a DTMF blind
+ transfer after being retrieved. When a parked call was retrieved
+ from the parking lot, it could not do a blind transfer because it
+ caused the involved calls to be hung up unconditionally. * Made
+ the ParkedCall application return the ast_bridge_call() return
+ value. (closes issue ABE-2862) Reported by: Vlad Povorozniuc
+ ........ Merged revisions 368567 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368568 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/features.c: Make builtin_blindtransfer() fully use
+ ast_async_goto() abilities.
+
+2012-06-05 16:25 +0000 [r368550] Jonathan Rose <jrose@digium.com>
+
+ * CHANGES: Merge 'core' and 'core changes' sections in CHANGES
+ file.
+
+2012-06-05 15:28 +0000 [r368519-368537] Kinsey Moore <kmoore@digium.com>
+
+ * /: Recorded merge of revisions 368536 from
+ http://svn.asterisk.org/svn/asterisk/branches/10 ........ Resolve
+ some build warnings My newly upgraded compiler caught these
+ usages of uninitialized values. They weren't actually used.
+ ........ Merged revisions 368533 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, 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)
+ ........ Merged revisions 368520 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368524 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * apps/app_dial.c, channels/chan_unistim.c, channels/chan_local.c,
+ channels/chan_sip.c, main/channel_internal_api.c,
+ main/features.c, include/asterisk/channel.h, apps/app_queue.c:
+ Convert AST_FLAG_ANSWERED_ELSEWHERE usage to
+ AST_CAUSE_ANSWERED_ELSEWHERE This was essentially duplicated
+ functionality where normal channels used
+ AST_CAUSE_ANSWERED_ELSEWHERE while local channels and queues used
+ AST_FLAG_ANSWERED_ELSEWHERE. This removes the flag and converts
+ that usage into AST_CAUSE_ANSWERED_ELSEWHER usage. Review:
+ https://reviewboard.asterisk.org/r/1944 (closes issue
+ ASTERISK-19865) Patch-by: Birger Harzenetter
+
+2012-06-04 22:12 +0000 [r368500] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Relay proper SIP responses on calling
+ side. Revision 351130 broke corect HANGUPCAUSE setting for the
+ 404 case in chan_sip. Other cases were also potentially broken.
+ This patch fixes the relaying of causes to be what they used to
+ be. (closes issue ASTERISK-19914) Reported by Pavel Troller
+ Tested by Walter Doekes (via a reviewboard test to be committed
+ later) Patches: chan_sip.diff uploaded by Pavel Troller (license
+ #6302) ........ Merged revisions 368498 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368499 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-04 21:18 +0000 [r368472] Richard Mudgett <rmudgett@digium.com>
+
+ * /, UPGRADE.txt: Document BLINDTRANSFER behavior change. (issue
+ ASTERISK-19322) (closes issue ASTERISK-19875) Reported by: call
+ ........ Merged revisions 368469 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368470 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-04 20:53 +0000 [r368435-368467] Mark Michelson <mmichelson@digium.com>
+
+ * contrib/editors/asterisk.vim: Also have vim syntax-highlight
+ type=network.
+
+ * contrib/editors/asterisk.vim: Add vim syntax highlighting for
+ type=line, type=phone, and type=application. (closes issue
+ ASTERISK-19800) Reported by: Billy Chia Patches:
+ asterisk.vim.patch uploaded by Billy Chia (license #6381)
+
+ * main/channel.c, apps/app_mixmonitor.c: Remove some extra
+ debugging I forgot to remove in the merge of Digium phone
+ support.
+
+ * /: Remove automerge properties.
+
+ * /, contrib/realtime/mysql/voicemail_messages.sql,
+ main/presencestate.c (added), main/config.c, main/channel.c,
+ include/asterisk/callerid.h, include/asterisk/file.h,
+ main/manager.c, channels/chan_skinny.c,
+ include/asterisk/event_defs.h, include/asterisk/sip_api.h
+ (added), tests/test_voicemail_api.c (added), main/features.c,
+ apps/app_voicemail.exports.in, main/app.c, main/message.c,
+ channels/sip/include/sip.h, main/pbx.c, channels/chan_sip.c,
+ include/asterisk/presencestate.h (added),
+ include/asterisk/config.h, include/asterisk/app_voicemail.h
+ (added), configs/manager.conf.sample, apps/app_queue.c,
+ include/asterisk/manager.h, include/asterisk/app.h,
+ funcs/func_presencestate.c (added), include/asterisk/message.h,
+ main/file.c, main/callerid.c, main/event.c,
+ include/asterisk/pbx.h, tests/test_config.c,
+ channels/chan_sip.exports.in (added), apps/app_mixmonitor.c,
+ main/asterisk.c, apps/app_voicemail.c: Merge changes dealing with
+ support for Digium phones. Presence support has been added. This
+ is accomplished by allowing for presence hints in addition to
+ device state hints. A dialplan function called PRESENCE_STATE has
+ been added to allow for setting and reading presence. Presence
+ can be transmitted to Digium phones using custom XML elements in
+ a PIDF presence document. Voicemail has new APIs that allow for
+ moving, removing, forwarding, and playing messages. Messages have
+ had a new unique message ID added to them so that the APIs will
+ work reliably. The state of a voicemail mailbox can be obtained
+ using an API that allows one to get a snapshot of the mailbox. A
+ voicemail Dialplan App called VoiceMailPlayMsg has been added to
+ be able to play back a specific message. Configuration hooks have
+ been added. Configuration hooks allow for a piece of code to be
+ executed when a specific configuration file is loaded by a
+ specific module. This is useful for modules that are dependent on
+ the configuration of other modules. chan_sip now has a public
+ method that allows for a custom SIP INFO request to be sent
+ mid-dialog. Digium phones use this in order to display progress
+ bars when files are played. Messaging support has been expanded a
+ bit. The main visible difference is the addition of an AMI action
+ MessageSend. Finally, a ParkingLots manager action has been added
+ in order to get a list of parking lots.
+
+2012-06-04 19:46 +0000 [r368421] Richard Mudgett <rmudgett@digium.com>
+
+ * 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/
+ ........ Merged revisions 368405 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368407 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-02 21:13 +0000 [r368359] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/utils.h, res/res_http_websocket.exports.in
+ (added), include/asterisk/http_websocket.h (added), main/utils.c,
+ res/res_http_websocket.c (added): Add res_http_websocket module
+ which implements the WebSocket protocol according to RFC 6455.
+ Review: https://reviewboard.asterisk.org/r/1952/
+
+2012-06-01 23:53 +0000 [r368311] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_stack.c: Fix deadlock when Gosub used with alternate
+ dialplan switches. Attempting to remove a channel from
+ autoservice with the channel lock held will result in deadlock. *
+ Restructured gosub_exec() to not call ast_parseable_goto() and
+ ast_exists_extension() with the channel lock held. (closes issue
+ ASTERISK-19764) Reported by: rmudgett Tested by: rmudgett
+ ........ Merged revisions 368308 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368310 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-01 20:42 +0000 [r368268-368269] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: Improve SDP offer/answer RFC compliance
+ Asterisk should not accept SDP offers that contain unknown RTP
+ profiles (for audio/video streams) or unknown top-level media
+ types. When it does, it answers with an SDP that does not match
+ the offer properly, and this will nearly always result in a
+ broken call. This patch causes such offers to be rejected.
+ Review: https://reviewboard.asterisk.org/r/1811/
+
+ * /, 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/ ........ Merged
+ revisions 368218 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368267 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-06-01 18:20 +0000 [r368181-368221] Terry Wilson <twilson@digium.com>
+
+ * configs/config_test.conf.sample (added): Add missing config for
+ config API test
+
+ * main/udptl.c, include/asterisk/utils.h,
+ include/asterisk/astobj2.h, configure.ac,
+ include/asterisk/config.h, main/astobj2.c, main/config.c,
+ Makefile, include/asterisk/config_options.h (added), configure,
+ main/asterisk.exports.in, apps/app_skel.c, main/config_options.c
+ (added), tests/test_config.c, makeopts.in,
+ configs/app_skel.conf.sample (added),
+ include/asterisk/stringfields.h: Add new config-parsing framework
+ This framework adds a way to register the various options in a
+ config file with Asterisk and to handle loading and reloading of
+ that config in a consistent and atomic manner. Review:
+ https://reviewboard.asterisk.org/r/1873/
+
+2012-06-01 13:04 +0000 [r368143] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample,
+ channels/sip/include/sip.h: 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. Review:
+ https://reviewboard.asterisk.org/r/1954
+
+2012-06-01 03:30 +0000 [r368094] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, funcs/func_channel.c: Add documentation to function CHANNEL
+ for options echocan_mode and buffers The ability to set
+ "echocan_mode" and "buffers" through the dialplan was added to
+ chan_dahdi some time ago. This patch adds some documentation to
+ func_channel. (Closes issue ASTERISK-19911) Reported by: Dale
+ Noll Tested by: Michael L. Young Patches:
+ asterisk-19911-branch18.diff uploaded by Michael L. Young
+ (license 5026) Review: https://reviewboard.asterisk.org/r/1949/
+ ........ Merged revisions 368092 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368093 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-31 18:39 +0000 [r368052] Richard Mudgett <rmudgett@digium.com>
+
+ * 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,
+ apps/app_queue.c, channels/chan_iax2.c, pbx/pbx_config.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 ........ Merged
+ revisions 368039 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 368042 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-30 18:08 +0000 [r367908-367982] Richard Mudgett <rmudgett@digium.com>
+
+ * /: Use the DEADLOCK_AVOIDANCE() macro instead. (issue
+ ASTERISK-19854) ........ Merged revisions 367980 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 367981 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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 ........ Merged revisions 367976 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 367978 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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 ........ Merged revisions 367906 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 367907 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-29 18:40 +0000 [r367845] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_skinny.c: AST-2012-008: Fix remote crash
+ vulnerability in chan_skinny When a skinny session is
+ unregistered, the corresponding device pointer is set to NULL in
+ the channel private data. If the client was not in the on-hook
+ state at the time the connection was closed, the device pointer
+ can later be dereferened 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 (licesen 6283) ........
+ Merged revisions 367844 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-25 16:33 +0000 [r367783] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_iax2.c: AST-2012-007: Fix IAX receiving HOLD
+ without suggested MOH class crash. * Made schedule_delivery() set
+ the received frame f->data.ptr to NULL if the datalen is zero. *
+ Fix queue_signalling() memcpy() size error. * Made
+ queue_signalling() not use C++ keyword variable names. (closes
+ issue ASTERISK-19597) Reported by: mgrobecker Patches:
+ jira_asterisk_19597_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: rmudgett, Michael L. Young ........ Merged
+ revisions 367781 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 367782 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-25 02:31 +0000 [r367732] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, channels/chan_sip.c: Fix pvt_sip for inbound call to use
+ peer's allowtransfer setting The pvt_sip allowtransfer was not
+ being set to that of the peer's setting. Therefore, the global
+ allowtransfer setting was being used instead which would lead to
+ calls not being transfered if the global setting was set to 'no'
+ despite the setting on the peer being 'yes' and vice versa, calls
+ would be allowed to transfer even if the peer's setting was 'no'
+ but the global setting was 'yes'. (Closes issue ASTERISK-19856)
+ Reported by: Jacek Tested by: Michael L. Young, Jacek Patches:
+ issue-asterisk-19856-branch10-v3.diff uploaded by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/1923/ ........ Merged
+ revisions 367730 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 367731 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-24 23:52 +0000 [r367693] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, /, apps/app_queue.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/ ........ Merged
+ revisions 367678 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 367679 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-24 18:56 +0000 [r367640] Jonathan Rose <jrose@digium.com>
+
+ * main/rtp_engine.c, channels/chan_sip.c,
+ include/asterisk/rtp_engine.h: chan_sip: fix problem
+ directmediapermit/deny uses the wrong address When remotely
+ bridging calls with directmedia, Asterisk would check the address
+ of the peers/users holding directmedia ACLs (set via
+ directmediapermit/directmediadeny) instead of the bridged peer.
+ This is similar to r366547, but trunk specific and involves
+ changes to the rtpengine instead of just chan_sip. (closes issue
+ AST-876) review: https://reviewboard.asterisk.org/r/1924/
+
+2012-05-24 13:33 +0000 [r367563] Matthew Jordan <mjordan@digium.com>
+
+ * /, apps/app_confbridge.c: Fix crash in ConfBridge when user
+ announcement is played for more than 2 users A patch introduced
+ in r354938 made it so that ConfBridge would not attempt to play
+ sound files if those files did not exist. Unfortunately,
+ ConfBridge uses the same underlying function, play_sound_helper,
+ to playback both sound files and numbers to callers. When a
+ number is being played back, the name of the sound file is
+ expected to be NULL. This NULL value was passed into a function
+ that tested for the existance of a sound file and is not tolerant
+ to NULL file names, causing a crash. This patch fixes the
+ behavior, such that if a sound file does not exist we do not
+ attempt to play it, but we only attempt that check if the a sound
+ file was specified in the first place. If a sound file was not
+ specified, we use the 'play number' logic in the helper function.
+ (closes issue ASTERISK-19899) Reported by: Florian Gilcher Tested
+ by: Florian Gilcher patches: asterisk-19899.diff uploaded by
+ mjordan (license 6283) ........ Merged revisions 367562 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-24 00:36 +0000 [r367477-367520] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/iax2-parser.c: Made use IAX frame cache only for
+ cacheable frame types.
+
+ * 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. ........ Merged
+ revisions 367469 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 367470 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-23 20:39 +0000 [r367419] Jonathan Rose <jrose@digium.com>
+
+ * main/pbx.c: logger: Fix a potential callid reference leak
+ discovered in development Uncovered a nasty reference leak while
+ I was writing some changes to chan_dahdi/sig_analog. Slapped
+ myself around a bit after seeing that I performed the unchecked
+ return causing this problem.
+
+2012-05-23 20:30 +0000 [r367418] Mark Michelson <mmichelson@digium.com>
+
+ * main/tcptls.c, /: Only call SSL_CTX_free if DO_SSL is defined.
+ Thanks to Paul Belanger for pointing out this error. ........
+ Merged revisions 367416 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 367417 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-23 13:46 +0000 [r367376] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: Re-add
+ LastMsgsSent value for SIP peers 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. The
+ value was previously removed for Asterisk 10; however, since it
+ was still present in Asterisk 1.8 and still useful for reporting
+ purposes, it was decided to re-add the value. This patch re-adds
+ the 'LastMsgsSent' field in the response to an AMI/CLI 'sip show
+ peer [peer]' command, and 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 ........ Merged revisions
+ 367362 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 367369 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-22 17:29 +0000 [r367274-367309] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, /, include/asterisk/cel.h,
+ main/channel_internal_api.c, include/asterisk/channel.h,
+ main/cel.c, main/asterisk.c: 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/
+ ........ Merged revisions 367292 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 367299 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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/ ........ Merged
+ revisions 367266 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 367267 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-21 22:45 +0000 [r367227] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Made ast_queue_hangup() and
+ ast_queue_hangup_with_cause() lock instead of trylock. It made no
+ sense to trylock the channel and then unconditionally lock the
+ channel right after.
+
+2012-05-21 20:35 +0000 [r367189] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_iax2.c: Make chan_iax2 reject cause code
+ indications correctly If chan_iax2 does not reject the
+ PVT_CAUSE_CODE frames, the cause will not be stored properly.
+
+2012-05-21 20:31 +0000 [r367163-367183] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/callerid.h, channels/chan_sip.c,
+ main/callerid.c: Revert revision 367163. This should have been
+ committed to my team trunk-digiumphones branch instead of trunk.
+
+ * include/asterisk/callerid.h, channels/chan_sip.c,
+ main/callerid.c: Add "send to voicemail" Digium phone
+ functionality to Asterisk. This change accommodates two methods
+ by which calls can be directed to a user's voicemail. * Incoming
+ calls can be redirected to any user's voicemail. * Established
+ calls can be blind transferred to any user's voicemail. Digium
+ phones indicate the desire to direct a call to voicemail by using
+ a Diversion header with a reason parameter of "send_to_vm". This
+ patch adds the "send_to_vm" reason as a valid redirecting reason.
+ In addition, chan_sip.c has been modified to update redirecting
+ information on the transferred channel by reading a Diversion
+ header on a REFER request. (closes issue AST-871) Reported by
+ Malcolm Davenport Review: https://reviewboard.asterisk.org/r/1925
+
+2012-05-21 17:39 +0000 [r367124] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/astobj2.h: Minor documentation change
+
+2012-05-18 19:39 +0000 [r367080] Jonathan Rose <jrose@digium.com>
+
+ * configs/queues.conf.sample, CHANGES, apps/app_queue.c: app_queue:
+ Per Member ringinuse option and deprecation of ignorebusy Adds a
+ number of methods for controlling the setting of 'ringinuse'
+ which is basically the same concept as the old ignorebusy
+ setting, only now the per member setting always controls whether
+ or not the member is actually ringed while in use. A CLI command
+ and a manager action have been added to change a given queue
+ member's ringinuse option while Asterisk is running and the an
+ argument has been added for adding members with deliberately set
+ ringinuse in queues.conf Some effort has been made to ensure
+ compatability with dialplans and databases still referring to
+ 'ignorebusy'. (issue ASTERISK-19536) reported by: Philippe
+ Lindheimer Review: https://reviewboard.asterisk.org/r/1919/
+
+2012-05-18 17:54 +0000 [r367010-367029] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_dahdi.c, /, main/say.c: Address MISSING_BREAK
+ static analysis reports some more. This addresses core findings 4
+ and 6. Moises Silva helped me by stating that a break could be
+ safely added to the case where it is added in chan_dahdi.c In
+ say.c, I have added a comment indicating that static analysis
+ complains but that it is currently unknown if this is correct.
+ This fixes all core findings of this type. (closes issue
+ ASTERISK-19662) reported by Matthew Jordan ........ Merged
+ revisions 367027 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 367028 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h:
+ 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) ........ Merged revisions 367002 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 367003 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-18 15:51 +0000 [r366917-366955] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_dahdi.c, /, channels/chan_sip.c, funcs/func_odbc.c,
+ main/cli.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 * chan_dahdi: free
+ a frame that was allocated by the dsp layer if we choose not to
+ process that frame (issue ASTERISK-19665) Reported by: Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/1922/ ........
+ Merged revisions 366944 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 366948 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/netsock2.c, res/res_rtp_asterisk.c, main/pbx.c,
+ res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
+ apps/app_page.c, /, funcs/func_dialgroup.c, channels/chan_sip.c,
+ apps/app_record.c, res/res_calendar_caldav.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/cli.c,
+ main/editline/tokenizer.c, main/data.c, channels/chan_dahdi.c,
+ funcs/func_odbc.c, main/features.c, apps/app_minivm.c,
+ main/editline/readline.c, channels/sip/config_parser.c,
+ main/xmldoc.c, res/res_calendar.c, apps/app_voicemail.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 ........ Merged revisions
+ 366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 366881 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-18 14:27 +0000 [r366896] Jonathan Rose <jrose@digium.com>
+
+ * channels/sip/dialplan_functions.c: chan_sip: Fix a small
+ TEST_FRAMEWORK related error that prevents compiling Introduced
+ with r366842, a function call made only with TEST_FRAMEWORK
+ enabled was missing an argument since the function arguments were
+ changed.
+
+2012-05-18 14:21 +0000 [r366843-366888] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/sip/config_parser.c: Reorder and renumber tests
+ appropriately It appears that a patch did not apply properly when
+ adding tests 12 and 13 and test 11 was duplicated. These tests
+ have been reordered and renumbered such that they make sense.
+ ........ Merged revisions 366882 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 366884 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/channel.c: Make the new SIP_CAUSE backend behave more like
+ the original SIP_CAUSE There was a slight discrepancy in the
+ behaviors of the old SIP_CAUSE and the new SIP_CAUSE/HANGUPCAUSE
+ when a channel had been originated and had not yet been answered.
+ This caused the noload_res_srtp_attempt_srtp test to fail since
+ the SIP_CAUSE variable was never actually set. This behavior has
+ been restored.
+
+2012-05-17 16:28 +0000 [r366842] Jonathan Rose <jrose@digium.com>
+
+ * include/asterisk/logger.h, main/channel.c,
+ channels/sip/include/dialog.h, main/pbx.c, channels/chan_sip.c,
+ main/channel_internal_api.c, main/logger.c,
+ include/asterisk/channel.h, CHANGES, channels/sip/include/sip.h,
+ main/cli.c: logger: Adds additional support for call id logging
+ and chan_sip specific stuff This patch improves the handling of
+ call id logging significantly with regard to transfers and adding
+ APIs to better handle specific aspects of logging. Also, changes
+ have been made to chan_sip in order to better handle the creation
+ of callids and to enable the monitor thread to bind itself to a
+ particular call id when a dialog is determined to be related to a
+ callid. It then unbinds itself before returning to normal
+ monitoring. review: https://reviewboard.asterisk.org/r/1886/
+
+2012-05-17 13:21 +0000 [r366746] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_dahdi.c, /, res/res_calendar_ews.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 ........ Merged
+ revisions 366740 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 366741 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-16 18:00 +0000 [r366663-366700] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/astobj2.h: Remove missed idx parameter to some
+ ao2 global holder macros.
+
+ * include/asterisk/astobj2.h, tests/test_astobj2.c, main/astobj2.c:
+ Change ao2 global array to ao2 global object holder. Review:
+ https://reviewboard.asterisk.org/r/1921/
+
+2012-05-15 23:41 +0000 [r366599] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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.
+ ........ Merged revisions 366597 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 366598 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-15 19:36 +0000 [r366462-366546] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_local.c: The predial routine must be run on the
+ local;1 channel. When ast_call() operates on a local channel, it
+ copies a lot of things from the local;1 channel to the local;2
+ channel. This includes among other things, channel variables and
+ party id information. Other reasons it was a bad idea to run
+ predial on the local;2 channel: 1) The channel has not been
+ completely setup. The ast_call() completes the setup. 2) The
+ local;2 caller and connected line party information is opposite
+ to any other channels predial runs on. (And it hasn't been setup
+ yet.) * Partially back out -r366183 by removing the chan_local
+ implementation of the struct ast_channel_tech.pre_call callback.
+
+ * CHANGES, apps/app_followme.c: Add predial support to FollowMe.
+ Like the new predial feature for Dial. This adds the same b/B
+ options to FollowMe. Review:
+ https://reviewboard.asterisk.org/r/1910/
+
+ * channels/chan_local.c: Make chan_local use the API call instead
+ of inlining its own version.
+
+2012-05-14 20:15 +0000 [r366413] Mark Michelson <mmichelson@digium.com>
+
+ * /, pbx/dundi-parser.c: Fix two more coverity constant expression
+ result findings. These correspond to findings 0 and 1 in the core
+ findings of ASTERISK-19649. After contacting Mark Spencer, he was
+ unsure of what the intent behind these lines of code were, so
+ they are being axed. For Asterisk 1.8 and 10, the output of
+ debugging DUNDi frames will not be changed, but for trunk the
+ "Retry" portion will be omitted since it does not properly
+ distinguish retransmissions from initial frames. (closes issue
+ ASTERISK-19649) Reported by Matthew Jordan ........ Merged
+ revisions 366409 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 366412 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-14 19:44 +0000 [r366408] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_unistim.c, apps/app_dial.c, main/rtp_engine.c,
+ channels/chan_vpb.cc, channels/chan_sip.c, UPGRADE.txt,
+ channels/chan_gtalk.c, channels/chan_console.c,
+ channels/chan_iax2.c, apps/app_queue.c, apps/app_followme.c,
+ channels/chan_oss.c, channels/chan_jingle.c, main/channel.c,
+ channels/chan_phone.c, main/dial.c, channels/chan_misdn.c,
+ channels/chan_skinny.c, funcs/func_frame_trace.c,
+ main/features.c, channels/chan_h323.c, main/file.c,
+ channels/chan_alsa.c, configs/sip.conf.sample,
+ include/asterisk/frame.h, channels/chan_mgcp.c: Commit framework
+ for HANGUPCAUSE (replacement for SIP_CAUSE) This is the starting
+ point for the Asterisk 11: Who Hung Up work and provides a
+ framework which will allow channel drivers to report the types of
+ hangup cause information available in SIP_CAUSE without incurring
+ the overhead of the MASTER_CHANNEL dialplan function. The initial
+ implementation only includes cause generation for chan_sip and
+ does not include cause code translation utilities. This change
+ deprecates SIP_CAUSE and replaces its method of reporting cause
+ codes with the new framework. This change also deprecates the
+ 'storesipcause' option in sip.conf. Review:
+ https://reviewboard.asterisk.org/r/1822/ (Closes issue SWP-4221)
+
+2012-05-14 19:27 +0000 [r366401] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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 ........ Merged revisions
+ 366389 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 366390 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-14 13:42 +0000 [r366351] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * configure, configure.ac, autoconf/ast_pkgconfig.m4 (added): Macro
+ AST_PKG_CONFIG_CHECK to use chkconfig AST_PKG_CONFIG_CHECK:
+ Similar to AST_EXT_LIB_CHECK, but simply uses pkg-config data.
+ This simple version only uses pkg-config(1)'s tests. This commit
+ also uses the macro to test for GTK2 and GMIME (instead of the
+ current direct usage of pkg-config). Review:
+ https://reviewboard.asterisk.org/r/1906/
+
+2012-05-12 00:03 +0000 [r366298] Russell Bryant <russell@russellbryant.com>
+
+ * /, addons/format_mp3.c: format_mp3: Fix a possible crash in
+ 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
+ ........ Merged revisions 366296 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 366297 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-10 23:49 +0000 [r366183-366242] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /: * Made ast_change_name() hold the channels
+ container lock while changing the channel name. * Eliminate
+ redundant list not empty check in clone_variables(). ........
+ Merged revisions 366240 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 366241 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * apps/app_dial.c: Tweak app_dial predial documentation.
+
+ * apps/app_dial.c, main/channel.c, channels/chan_local.c,
+ include/asterisk/channel.h: Run predial routine on local;2
+ channel where you would expect. Before this patch, the predial
+ routine executes on the ;1 channel of a local channel pair.
+ Executing predial on the ;1 channel of a local channel pair is of
+ limited utility. Any channel variables set by the predial routine
+ executing on the ;1 channel will not be available when the local
+ channel executes dialplan on the ;2 channel. * Create
+ ast_pre_call() and an associated pre_call() technology callback
+ to handle running the predial routine. If a channel technology
+ does not provide the callback, the predial routine is simply run
+ on the channel. Review: https://reviewboard.asterisk.org/r/1903/
+
+2012-05-10 20:56 +0000 [r366169] Kinsey Moore <kmoore@digium.com>
+
+ * 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, 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: 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) ........ Merged
+ revisions 366167 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 366168 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-10 18:35 +0000 [r366126] Jonathan Rose <jrose@digium.com>
+
+ * main/pbx.c, channels/sig_analog.c, /, channels/chan_sip.c,
+ funcs/func_lock.c, main/features.c, main/acl.c,
+ channels/iax2-provision.c, apps/app_queue.c,
+ channels/chan_iax2.c, res/ael/ael.flex, funcs/func_devstate.c,
+ main/asterisk.c, main/xmldoc.c, apps/app_voicemail.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/ ........ Merged
+ revisions 366094 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 366106 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-10 16:22 +0000 [r366062] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Close the proper tcptls_session when
+ session creation fails. (issue AST-998) Reported by: Thomas
+ Arimont Tested by: Thomas Arimont ........ Merged revisions
+ 366052 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 366053 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-10 15:57 +0000 [r366007-366051] Jonathan Rose <jrose@digium.com>
+
+ * /, funcs/func_cdr.c, main/features.c, apps/app_disa.c,
+ apps/app_chanspy.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/
+ ........ Merged revisions 366048 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 366049 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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) ........ Merged
+ revisions 365989 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 365990 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-09 19:26 +0000 [r366002] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * Makefile: pass BUILD_CFLGAS and BUILD_LDFLAGS to menuselect Allow
+ menuselect to get its set of CFLAGS and LDFLAGS through the
+ environment of Make: make BUILD_CFLAGS="whatever"
+ BUILD_LDFLAGS="whatever" Review:
+ https://reviewboard.asterisk.org/r/1907/
+
+2012-05-09 17:58 +0000 [r365951] Richard Mudgett <rmudgett@digium.com>
+
+ * configs/followme.conf.sample, apps/app_followme.c: Improve
+ FollowMe accept/decline DTMF string matching. If you hit the
+ wrong DTMF digit trying to accept/decline a FollowMe call, you
+ had to wait for the prompt to repeat to try again. * Make
+ FollowMe compare the last DTMF digits received to the
+ accept/decline matching strings.
+
+2012-05-09 16:36 +0000 [r365913] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Prevent sip_pvt refleak when an
+ ast_channel outlasts its corresponding sip_pvt. chan_sip was
+ coded under the assumption that a SIP dialog with an owner
+ channel will always be destroyed after the owner channel has been
+ hung up. However, there are situations where the SIP dialog can
+ time out and auto destruct before the corresponding channel has
+ hung up. A typical example of this would be if the 'h' extension
+ in the dialplan takes a long time to complete. In such cases,
+ __sip_autodestruct() would complain about the dialog being auto
+ destroyed with an owner channel still in place. The problem is
+ that even once the owner channel was hung up, the sip_pvt would
+ still be linked in its ao2_container because nothing would ever
+ unlink it. The fix for this is that if __sip_autodestruct() is
+ called for a sip_pvt that still has an owner channel in place,
+ the destruction is rescheduled for 10 seconds in the future. This
+ will continue until the owner channel is finally hung up. (closes
+ issue ASTERISK-19425) reported by David Cunningham Patches:
+ ASTERISK-19425.patch uploaded by Mark Michelson (License #5049)
+ (closes issue ASTERISK-19455) reported by Dean Vesvuio Tested by
+ Dean Vesvuio ........ Merged revisions 365896 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 365898 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-09 02:35 +0000 [r365766-365856] Richard Mudgett <rmudgett@digium.com>
+
+ * configs/followme.conf.sample, UPGRADE.txt, apps/app_followme.c:
+ Keep answered FollowMe calls until call accepted or last step
+ times out.
+
+ * apps/app_followme.c: Put winning FollowMe outgoing call on hold
+ if the caller put it on hold. The FollowMe caller call leg is
+ usually answered and listening to MOH. The caller could put the
+ call on hold while FollowMe is looking for a winner. The winning
+ outgoing call is now immediately placed on hold if the caller has
+ put the call on hold before the winning call was selected.
+
+ * apps/app_followme.c: Restructure how the FollowMe outgoing
+ channel list is handled.
+
+ * apps/app_followme.c: Addendum to -r365766. Since it is no longer
+ allocated.
+
+ * apps/app_followme.c: Make FollowMe findmeexec() put the list head
+ on the stack instead of mallocing it. Why this tiny struct was
+ malloced instead of the 28k struct in the last change is beyond
+ me. Just doing my part to help stamp out sillyness.
+
+2012-05-08 21:46 +0000 [r365751] Sean Bright <sean@malleable.com>
+
+ * apps/app_externalivr.c: Add interrupt ('I') command to
+ ExternalIVR. Sending the 'I' command from an external process
+ will cause the current playlist to be cleared, including stopping
+ any audio file that is currently playing. This is useful when you
+ want to interrupt audio playback only when specific DTMF is
+ entered by the caller.
+
+2012-05-08 21:41 +0000 [r365633-365749] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_followme.c: Make FollowMe app_exec() not declare a 28k
+ struct on the stack. Helping to stamp out stack abuse.
+
+ * apps/app_followme.c: Simplify findmeexec() parameter passing.
+
+ * /, 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 fm_args.namerecloc[]. This fixes unexpected
+ namerecloc filename length restriction. ........ Merged revisions
+ 365692 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 365701 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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. ........ Merged revisions 365631 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 365632 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-08 15:57 +0000 [r365576] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Send more accurate identification
+ information in dialog-info SIP NOTIFYs. This uses the calling
+ channel's caller ID and connected line information to populate
+ the remote and local identities in the dialog-info NOTIFY when an
+ extension is ringing. There is a bit of an oddity here, and that
+ is that we seed the remote target with the To header of the
+ outbound call rather than the from header. This is because it was
+ reported that seeding with the from header caused hints to be
+ broken with certain SNOM devices. A comment has been added to the
+ code to explain this. (closes issue ASTERISK-16735) reported by
+ Maciej Krajewski patches: local_remote_hint2.diff uploaded by
+ Mark Michelson (license #5049) 16735_tweak1.diff uploaded by Mark
+ Michelson (license #5049) Tested by Niccolo Belli ........ Merged
+ revisions 365574 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 365575 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-07 20:08 +0000 [r365532] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: Change comment to use local channel name
+ designators in features.c
+
+2012-05-07 18:58 +0000 [r365480] Matthew Jordan <mjordan@digium.com>
+
+ * main/pbx.c, apps/app_voicemail.c: Fix channel opaquification
+ slip-up in r365477 Those channels are opaque now...
+
+2012-05-07 18:51 +0000 [r365479] Richard Mudgett <rmudgett@digium.com>
+
+ * /, tests/test_config.c: Fix type punned compiler warning in
+ test_config.c ........ Merged revisions 365476 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 365478 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-07 18:42 +0000 [r365477] Matthew Jordan <mjordan@digium.com>
+
+ * main/pbx.c, /, apps/app_voicemail.c: Support VoiceMail d() option
+ when extension does not exist in channel's context The VoiceMail
+ d([c]) option is documented to accept digits for a new extension
+ in context <c>, if played during the greeting. This option works
+ fine if the extension being redirected to has an extension with
+ the same initial digit in the channel's current context. If that
+ digit did not happen to exist in some extension, a dialplan match
+ would fail and the user would not be redirected. This patch fixes
+ it such that if the <c> option is used, the extensions are
+ matched in that context as opposed to the caller's original
+ context. (closes issue ASTERISK-18243) Reported by: mjordan
+ Tested by: mjordan Review:
+ https://reviewboard.asterisk.org/r/1892 ........ Merged revisions
+ 365474 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 365475 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-04 22:17 +0000 [r365400] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c, funcs/func_aes.c, main/features.c,
+ 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: 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) ........ Merged revisions 365398
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 365399 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-04 17:38 +0000 [r365356] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_local.c, /: Fix local channel chains optimizing
+ themselves out of a call. * Made chan_local.c:check_bridge()
+ check the return value of ast_channel_masquerade(). In long
+ chains of local channels, the masquerade occasionally fails to
+ get setup because there is another masquerade already setup on an
+ adjacent local channel in the chain. * Made the outgoing local
+ channel (the ;2 channel) flush one voice or video frame per
+ optimization attempt. * Made sure that the outgoing local channel
+ also does not have any frames in its queue before the masquerade.
+ * Made do the masquerade immediately to minimize the chance that
+ the outgoing channel queue does not get any new frames added and
+ thus unconditionally flushed. * Made block indication -1 (Stop
+ tones) event when the local channel is going to optimize itself
+ out. When the call is answered, a chain of local channels pass
+ down a -1 indication for each bridge. This blizzard of -1 events
+ really slows down the optimization process. (closes issue
+ ASTERISK-16711) Reported by: Alec Davis Tested by: rmudgett, Alec
+ Davis Review: https://reviewboard.asterisk.org/r/1894/ ........
+ Merged revisions 365313 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 365320 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-04 15:52 +0000 [r365300] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_rtp_asterisk.c, /: Fix core FINDING 2, FINDING 3, and
+ FINDING 4 from Coverity's CONSTANT_EXPRESSION_RESULT report.
+ These three all are in RTP code that attempts to print the number
+ of sequence number cycles in an RTCP RR report. The code was
+ masking out the upper 16 bits and then shifting the number right
+ by 16 bits. This led to an all zero result in all cases. The fix
+ is to do the shift without the bit masking. (issue
+ ASTERISK-19649) ........ Merged revisions 365298 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 365299 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-03 19:36 +0000 [r365248] Michael L. Young <elgueromexicano@gmail.com>
+
+ * tests/test_security_events.c: Update security events unit tests
+ The security events framework API was changed in Asterisk 10 but
+ the unit tests were not updated at the same time. This patch does
+ the following: * Adds two more security events that were added to
+ the API * Add challenge, received_challenge and received_hash in
+ the inval_password security event unit test (Closes issue
+ ASTERISK-19760) Reported by: Michael L. Young Tested by: Michael
+ L. Young Patches: issue-asterisk-19760-trunk.diff uploaded by
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/1897/
+
+2012-05-03 18:43 +0000 [r365213] Sean Bright <sean@malleable.com>
+
+ * CHANGES: Update documentation references in CHANGES to reflect
+ the correct pages on the wiki. The current CHANGES file refers to
+ doc/ in many places and those files no longer exist.
+
+2012-05-03 15:05 +0000 [r365161] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh323.c, /,
+ addons/ooh323c/src/h323/H323-MESSAGES.h,
+ addons/ooh323c/src/h323/H323-MESSAGESEnc.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) ........ Merged revisions
+ 365159 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 365160 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-03 14:47 +0000 [r365158] Sean Bright <sean@malleable.com>
+
+ * apps/app_externalivr.c, CHANGES: Add IPv6 support to ExternalIVR.
+ Review: https://reviewboard.asterisk.org/r/1896/
+
+2012-05-03 14:35 +0000 [r365157] Alexandr Anikin <may@telecom-service.ru>
+
+ * /, 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) ........ Merged
+ revisions 365143 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 365155 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-02 17:43 +0000 [r365084] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c, /, main/cel.c: Multiple revisions
+ 365006,365068 ........ r365006 | twilson | 2012-05-02 10:49:03
+ -0500 (Wed, 02 May 2012) | 12 lines 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/
+ ........ r365068 | twilson | 2012-05-02 12:02:39 -0500 (Wed, 02
+ May 2012) | 11 lines 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/
+ ........ Merged revisions 365006,365068 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 365083 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-02 15:59 +0000 [r365011] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c: Save the address on which a MESSAGE was
+ received, so it can be used in MESSAGE() This is useful in cases
+ where chan_sip may be listening on multiple addresses.
+
+2012-05-02 02:51 +0000 [r364966] Matthew Jordan <mjordan@digium.com>
+
+ * /, main/audiohook.c: Only log a failure to get read/write samples
+ from factories if it didn't happen In audiohook_read_frame_both,
+ anytime samples are obtained from the read/write factories a
+ debug statement is logged stating that samples were not obtained
+ from the factories. This statement used to only occur if
+ option_debug was turned on and no samples were obtained; in some
+ refactoring when the option_debug statement was removed, the
+ "else" clause was removed as well. This patch makes it so that
+ those debug log statements only occur if the condition leading up
+ to them actually happened. ........ Merged revisions 364965 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-01 23:23 +0000 [r364915] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Remove a function that has been marked
+ unused since Asterisk 1.6.0. The reason I'm removing this is that
+ Coverity reported a STRAY_SEMICOLON issue here. Since the
+ function has been unused for so long, I just elected to remove it
+ altogether. (closes issue ASTERISK-19660)
+
+2012-05-01 23:21 +0000 [r364910] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/astobj2.c: Fixed __ao2_ref() validating user_data twice.
+ (closes issue ASTERISK-19755) Reported by: Gunther Kelleter
+ Patches: ao2_ref.patch (license #6372) patch uploaded by Gunther
+ Kelleter ........ Merged revisions 364902 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 364903 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-01 23:11 +0000 [r364901] Mark Michelson <mmichelson@digium.com>
+
+ * /, funcs/func_volume.c: Fix Coverity-reported ARRAY_VS_SINGLETON
+ error. As it turned out, this wasn't a huge deal. We were calling
+ ast_app_parse_options() for a set of options of which none took
+ arguments. The proper thing to do for this case is to pass NULL
+ for the "args" parameter here. We were instead passing a
+ seemingly-randomly chosen char * from the function. While this
+ would never get written to, you can rest assured things would
+ have gotten bad had new options (which took arguments) been added
+ to func_volume. (closes issue ASTERISK-19656) ........ Merged
+ revisions 364899 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 364900 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-01 22:00 +0000 [r364846] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_local.c, /: * Fix error path resouce leak in
+ local_request(). * Restructure local_request() to reduce
+ indentation. ........ Merged revisions 364840 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 364845 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-01 21:49 +0000 [r364844] Jason Parker <jparker@digium.com>
+
+ * main/manager.c, /: Prevent a potential crash when using manager
+ hooks. Found by me while poking at DPMA-127. ........ Merged
+ revisions 364841 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 364842 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-01 19:10 +0000 [r364788] Kinsey Moore <kmoore@digium.com>
+
+ * /, apps/app_confbridge.c: Play conf-placeintoconf message to the
+ correct channel Correct the code in app_confbridge to play the
+ conf-placeintoconf message to the marked user entering the bridge
+ instead of to the conference while the marked user hears silence.
+ (closes issue ASTERISK-19641) Reported-by: Mark A Walters
+ ........ Merged revisions 364786 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 364787 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-01 18:29 +0000 [r364785] Jonathan Rose <jrose@digium.com>
+
+ * /, main/app.c: Fix bad check in voicemail functions for
+ ast_inboxcount2_func Check looks for ast_inboxcount_func instead
+ of ast_inboxcount2_func on ast_inboxcount2_func calls. (closes
+ issue ASTERISK-19718) Reported by: Corey Farrell Patches:
+ ast_app_inboxcount2-null-refcheck.patch uploaded by Corey Farrell
+ (license 5909) ........ Merged revisions 364769 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 364777 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-30 19:51 +0000 [r364708] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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) ........ Merged revisions 364706
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 364707 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-30 17:17 +0000 [r364654] Mark Murawki <markm@intellasoft.net>
+
+ * /, main/logger.c: Merged revisions 364635 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r364635 | markm | 2012-04-30 11:51:12 -0400 (Mon, 30 Apr 2012) |
+ 10 lines 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 ........ ........ Merged revisions 364650 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-30 16:59 +0000 [r364652] Alexandr Anikin <may@telecom-service.ru>
+
+ * /, addons/ooh323cDriver.c: Fix use freed pointer in return value
+ from call thread (issue ASTERISK-19663) Reported by: Matt Jordan
+ Patches: ASTERISK-19663-ooh323.patch (License #5415) ........
+ Merged revisions 364649 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 364651 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-29 19:50 +0000 [r364580] Matthew Jordan <mjordan@digium.com>
+
+ * formats/format_ilbc.c, /, formats/format_sln.c,
+ formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c,
+ formats/format_g723.c, formats/format_h263.c,
+ formats/format_h264.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: 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 ........ Merged revisions 364578 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 364579 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-29 02:23 +0000 [r364537] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, apps/confbridge/conf_config_parser.c: Fix configuring custom
+ sound_leader_has_left in confbridge.conf The configuration option
+ to specify a custom sound_leader_has_left file for a conference
+ bridge was not being parsed. This patch fixes it so that a custom
+ sound file will now be used. (closes issue ASTERISK-19771)
+ Reported by: Pawel Kuzak Tested by: Pawel Kuzak, Michael L. Young
+ Patches: leaderhasleft_sound.dpatch uploaded by Pawel Kuzak
+ (license 6380) Review: https://reviewboard.asterisk.org/r/1884/
+ ........ Merged revisions 364536 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-28 20:24 +0000 [r364500] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
+ channels/sip/include/sip.h: Add support for lightweight NAT
+ keepalive. If enabled using the keepalive option in sip.conf a
+ small packet will be sent at a regular interval to keep the NAT
+ mapping open. This is lightweight as the remote side does not
+ need to parse and handle a SIP message. (closes issue AST-783)
+ Review: https://reviewboard.asterisk.org/r/1756/
+
+2012-04-28 01:33 +0000 [r364437-364462] Russell Bryant <russell@russellbryant.com>
+
+ * main/md5.c: md5: supress some compiler warnings. md5.c: In
+ function ‘MD5Final’: md5.c:154:2: error: dereferencing
+ type-punned pointer will break strict-aliasing rules
+ [-Werror=strict-aliasing] md5.c:155:2: error: dereferencing
+ type-punned pointer will break strict-aliasing rules
+ [-Werror=strict-aliasing] There is an md5 unit test and it still
+ passes.
+
+ * configure, include/asterisk/autoconfig.h.in, res/res_corosync.c,
+ configure.ac: res_corosync: Fix build against corosync 2.0.
+
+ * apps/app_minivm.c: app_minivm: Fix a couple compiler warnings.
+ The warnings were about argv[0] being used uninitialized, which
+ is correct. Just remove setting username to this value, since
+ username is set again before it actually gets used.
+
+ * main/features.c, CHANGES: features: Add FEATURE() and
+ FEATUREMAP() functions. Add two new dialplan functions: FEATURE()
+ and FEATUREMAP(). FEATURE() lets you set some of the
+ configuration options from the [general] section of features.conf
+ on a per-channel basis. FEATUREMAP() lets you customize the key
+ sequence used to activate built-in features, such as blindxfer,
+ and automon. See the built-in documentation for details. Review:
+ https://reviewboard.asterisk.org/r/1871/
+
+2012-04-28 00:31 +0000 [r364436] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, CHANGES: PreDial - Ability to run dialplan on
+ callee and caller channels before Dial. Thanks to Mark Murawski
+ for the initial patch and feature definition. (closes issue
+ ASTERISK-19548) Reported by: Mark Murawski Review:
+ https://reviewboard.asterisk.org/r/1878/ Review:
+ https://reviewboard.asterisk.org/r/1229/
+
+2012-04-27 22:54 +0000 [r364397] Terry Wilson <twilson@digium.com>
+
+ * /, tests/test_config.c (added), main/config.c: Multiple revisions
+ 364365,364369 ........ r364365 | twilson | 2012-04-27 17:31:01
+ -0500 (Fri, 27 Apr 2012) | 11 lines 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/
+ ........ Merged revisions 364340 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r364369 | twilson | 2012-04-27 17:33:10 -0500 (Fri, 27 Apr 2012)
+ | 2 lines Add missing test_config.c ........ Merged revisions
+ 364365,364369 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-27 22:11 +0000 [r364343] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Don't attempt to make use of the
+ dynamic_exclude_static ACL if DNS lookup fails. (closes issue
+ ASTERISK-18321) Reported by Dan Lukes Patches:
+ ASTERISK-18321.patch by Mark Michelson (license #5049) ........
+ Merged revisions 364341 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 364342 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-27 19:30 +0000 [r364287] Matthew Jordan <mjordan@digium.com>
+
+ * /, include/asterisk/time.h: Prevent overflow in calculation in
+ ast_tvdiff_ms on 32-bit machines The method ast_tvdiff_ms
+ attempts to calculate the difference, in milliseconds, between
+ two timeval structs, and return the difference in a 64-bit
+ integer. Unfortunately, it assumes that the long tv_sec/tv_usec
+ members in the timeval struct are large enough to hold the
+ calculated values before it returns. On 64-bit machines, this
+ might be the case, as a long may be 64-bits. On 32-bit machines,
+ however, a long may be less (32-bits), in which case, the
+ calculation can overflow. This overflow caused significant
+ problems in MixMonitor, which uses the method to determine if an
+ audio factory, which has not presented audio to an audiohook, is
+ merely late in providing said audio or will never provide audio.
+ In an overflow situation, the audiohook would incorrectly
+ determine that an audio factory that will never provide audio is
+ merely late instead. This led to situations where a MixMonitor
+ never recorded any audio. Note that this happened most frequently
+ when that MixMonitor was started by the ConfBridge application
+ itself, or when the MixMonitor was attached to a Local channel.
+ (issue ASTERISK-19497) Reported by: Ben Klang Tested by: Ben
+ Klang Patches: 32-bit-time-overflow-10-2012-04-26.diff (license
+ #6283) by mjordan (closes issue ASTERISK-19727) Reported by: Mark
+ Murawski Tested by: Michael L. Young Patches:
+ 32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan)
+ (closes issue ASTERISK-19471) Reported by: feyfre Tested by:
+ feyfre (issue ASTERISK-19426) Reported by: Johan Wilfer Review:
+ https://reviewboard.asterisk.org/r/1889/ ........ Merged
+ revisions 364277 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 364285 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-27 18:59 +0000 [r364260] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Allow SIP pvts involved in Replaces
+ transfers to fall out of reference sooner Unref the SIP pvt
+ stored in the refer structure as soon as it is no longer needed
+ so that the pvt and associated file descriptors can be freed
+ sooner. This change makes a reference decrement unnecessary in
+ code that handles SIP BYE/Also transfers which should not touch
+ the reference anyway. (Closes issue ASTERISK-19579) Reported by:
+ Maciej Krajewski Tested by: Maciej Krajewski ........ Merged
+ revisions 364258 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 364259 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-27 14:45 +0000 [r364205] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Allow for reloading SRTP crypto keys
+ within the same SIP dialog As a continuation of the patch in
+ r356604, which allowed for the reloading of SRTP keys in
+ re-INVITE transfer scenarios, this patch addresses the more
+ common case where a new key is requested within the context of a
+ current SIP dialog. This can occur, for example, when certain
+ phones request a SIP hold. Previously, once a dialog was
+ associated with an SRTP object, any subsequent attempt to process
+ crypto keys in any SDP offer - either the current one or a new
+ offer in a new SIP request - were ignored. This patch changes
+ this behavior to only ignore subsequent crypto keys within the
+ current SDP offer, but allows future SDP offers to change the
+ keys. (issue ASTERISK-19253) Reported by: Thomas Arimont Tested
+ by: Thomas Arimont Review:
+ https://reviewboard.asteriskorg/r/1885/ ........ Merged revisions
+ 364203 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 364204 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-27 12:58 +0000 [r364164] Stefan Schmidt <sst@sil.at>
+
+ * res/res_calendar_icalendar.c, /, res/res_calendar_caldav.c: fix a
+ wrong behavior of alarm timezones in caldav and icalendar when an
+ alarm doesnt use utc. This change uses the same timezone from the
+ start time. ........ Merged revisions 364163 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-26 21:11 +0000 [r364082-364110] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_directed_pickup.c: Update Pickup application
+ documentation. (With feeling this time.) ........ Merged
+ revisions 364108 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 364109 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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
+ ........ Merged revisions 364060 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 364065 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-26 19:33 +0000 [r364048] Terry Wilson <twilson@digium.com>
+
+ * /, main/asterisk.c: Add more constness to the end_buf pointer in
+ the netconsole issue ASTERISK-18308 Review:
+ https://reviewboard.asterisk.org/r/1876/ ........ Merged
+ revisions 364046 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 364047 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-26 13:59 +0000 [r363989] Olle Johansson <oej@edvina.net>
+
+ * apps/app_queue.c: Code formatting fixes.
+
+2012-04-26 13:31 +0000 [r363988] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Fix reference leaks involving SIP
+ Replaces transfers The reference held for SIP blind transfers
+ using the Replaces header in an INVITE was never freed on success
+ and also failed to be freed in some error conditions. This caused
+ a file descriptor leak since the RTP structures in use at the
+ time of the transfer were never freed. This reference leak and
+ another relating to subscriptions in the same code path have now
+ been corrected. (closes issue ASTERISK-19579) ........ Merged
+ revisions 363986 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 363987 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-26 09:48 +0000 [r363936] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, channels/chan_sip.c: chan_sip: [general] maxforwards, not
+ checked for a value greater than 255 The peer maxforwards is
+ checked for both '< 1' and '> 255', but the default 'maxforwards'
+ in the [general] section is only checked for '< 1' alecdavis
+ (license 585) Reported by: alecdavis Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1888/ ........ Merged
+ revisions 363934 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 363935 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-26 03:12 +0000 [r363689-363877] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_directed_pickup.c: Update Pickup application
+ documentation. (Even better) ........ Merged revisions 363875
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 363876 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * apps/app_directed_pickup.c: * Put more information in
+ pickup_exec() LOG_NOTICE. * Delay duplicating a string on the
+ stack in pickup_exec().
+
+ * /, apps/app_directed_pickup.c: Update Pickup application
+ documentation. ........ Merged revisions 363788 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 363789 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/chan_dahdi.c, /, channels/sig_pri.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 ........ Merged revisions 363730
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 363734 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.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) ........ Merged revisions 363687 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 363688 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-25 13:57 +0000 [r363480-363637] Olle Johansson <oej@edvina.net>
+
+ * apps/app_queue.c: Add documentation Thanks Tilghman!
+
+ * apps/app_queue.c: Formatting changes only
+
+ * apps/app_followme.c, apps/app_queue.c: Use the DEFINED value for
+ musicclass length. For some reason, features.c has it's own
+ definition. Should propably be fixed too.
+
+ * main/channel.c, configs/asterisk.conf.sample, CHANGES,
+ include/asterisk/options.h, main/asterisk.c: Make it possible to
+ change the minimum DTMF duration in asterisk.conf Asterisk has a
+ setting for the minimum allowed DTMF. If we get shorter DTMF
+ tones, these will be changed to the minimum on the outbound call
+ leg. (closes issue ASTERISK-19772) Review:
+ https://reviewboard.asterisk.org/r/1882/ Reported by: oej Tested
+ by: oej Patches by: oej Thanks to the reviewers. 1.8 branch for
+ this patch: agave-dtmf-duration-asterisk-conf-1.8
+
+ * main/say.c: Formatting fixes Developer guidelines are important.
+
+ * main/channel.c: Formatting fixes Found a small amount of curly
+ brackets in my hotel room here in Denmark. I hereby donate them
+ to the Asterisk project.
+
+2012-04-25 01:26 +0000 [r363377-363430] Richard Mudgett <rmudgett@digium.com>
+
+ * /, 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 ........ Merged revisions 363428 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 363429 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, main/features.c: Hangup affected channel in error paths of
+ bridge_call_thread(). ........ Merged revisions 363375 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 363376 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-24 17:52 +0000 [r363335] Terry Wilson <twilson@digium.com>
+
+ * /, main/asterisk.c: OpenBSD doesn't have rawmemchr, use strchr
+ (closes issue ASTERISK-19758) Reported by: Barry Miller Tested
+ by: Terry Wilson Patches: 362758-diff uploaded by Barry Miller
+ (license 5434) ........ Merged revisions 362868 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 362869 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-23 17:05 +0000 [r363269] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, apps/app_queue.c: Make app_dial and app_queue
+ use new macro and gosub calls. * Simplify some code in app_dial
+ and app_queue by calling ast_app_exec_macro() and
+ ast_app_exec_sub(). * Fix minor locking issue in app_dial for
+ post-answer macro/gosub MACRO/GOSUB_RESULT=GOTO: handling.
+
+2012-04-23 16:08 +0000 [r363215] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, main/astfd.c: On some platforms, O_RDONLY is not a flag to be
+ checked, but merely the absence of O_RDWR and O_WRONLY. The POSIX
+ specification does not mandate how these 3 flags must be
+ specified, only that one of the three must be specified in every
+ call. ........ Merged revisions 363209 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 363212 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-23 14:48 +0000 [r363159] Jonathan Rose <jrose@digium.com>
+
+ * main/manager.c, /: AST-2012-004: Fix an error that allows AMI
+ users to run shell commands sans authorization. As detailed in
+ the advisory, AMI users without write authorization for SYSTEM
+ class AMI actions were able to run system commands by going
+ through other AMI commands which did not require that
+ authorization. Specifically, GetVar and Status allowed users to
+ do this by setting their variable/s options to the SHELL or EVAL
+ functions. Also, within 1.8, 10, and trunk there was a similar
+ flaw with the Originate action that allowed users with originate
+ permission to run MixMonitor and supply a shell command in the
+ Data argument. That flaw is fixed in those versions of this
+ patch. (closes issue ASTERISK-17465) Reported By: David Woolley
+ Patches: 162_ami_readfunc_security_r2.diff uploaded by jrose
+ (license 6182) 18_ami_readfunc_security_r2.diff uploaded by jrose
+ (license 6182) 10_ami_readfunc_security_r2.diff uploaded by jrose
+ (license 6182) ........ Merged revisions 363117 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
+ Merged revisions 363141 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 363156 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-23 14:10 +0000 [r363105-363108] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: AST-2012-006: Fix crash in UPDATE
+ handling when no channel owner exists If Asterisk receives a SIP
+ UPDATE request after a call has been terminated and the channel
+ has been destroyed but before the SIP dialog has been destroyed,
+ a condition exists where a connected line update would be
+ attempted on a non-existing channel. This would cause Asterisk to
+ crash. The patch resolves this by first ensuring that the SIP
+ dialog has an owning channel before attempting a connected line
+ update. If an UPDATE request is received and no channel is
+ associated with the dialog, a 481 response is sent. (closes issue
+ ASTERISK-19770) Reported by: Thomas Arimont Tested by: Matt
+ Jordan Patches: ASTERISK-19278-2012-04-16.diff uploaded by Matt
+ Jordan (license 6283) ........ Merged revisions 363106 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 363107 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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 ........
+ Merged revisions 363102 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 363103 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-21 11:45 +0000 [r363045-363046] Russell Bryant <russell@russellbryant.com>
+
+ * res/res_corosync.c: res_corosync: Recover if corosync gets
+ restarted. If corosync gets restarted while Asterisk is running,
+ automatically recover.
+
+ * res/res_corosync.c: res_corosync: reimplement "corosync show
+ members" command. Reimplement the "corosync show members" CLI
+ command using a CPG iterator instead of the cpg_membership_get
+ API call. This will also show all CPG members, including those in
+ groups other than 'asterisk', which may be useful at some point
+ for debugging purposes.
+
+2012-04-21 01:46 +0000 [r362920-362999] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, /: Update app_dial M and U option GOTO return
+ value documentation. ........ Merged revisions 362997 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 362998 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * include/asterisk/app.h, main/app.c, apps/app_stack.c: Fix
+ connected-line/redirecting interception gosubs executing more
+ than intended. * Redo ast_app_run_sub()/ast_app_exec_sub() to use
+ a known return point so execution will stop after the routine
+ returns there. (s@gosub_virtual_context:1) * Create
+ ast_app_exec_macro() and ast_app_exec_sub() to run the macro and
+ gosub application respectively with the parameter string already
+ created.
+
+ * main/rtp_engine.c: Move debug message in
+ ast_rtp_instance_early_bridge_make_compatible(). Move debug
+ message in ast_rtp_instance_early_bridge_make_compatible() to be
+ output when what it states has actually happened.
+
+2012-04-20 16:50 +0000 [r362919] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, main/event.c: Add missing payload type to events API The
+ Security Events Framework API was changed while adding the
+ generation of security events in chan_sip. A payload type and
+ name was missed from being added to struct ie_maps. (closes issue
+ ASTERISK-19759) Reported by: Michael L. Young Patches:
+ issue-asterisk-19759.diff uploaded by Michael L. Young (license
+ 5026) ........ Merged revisions 362918 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-20 16:23 +0000 [r362867-362888] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, channels/chan_dahdi.c, channels/chan_local.c,
+ channels/chan_misdn.c, main/rtp_engine.c: Use
+ ast_channel_lock_both() where it was inlined before. The
+ CHANNEL_DEADLOCK_AVOIDANCE() feature of preserving where the
+ channel lock was originally obtained is overkill where
+ ast_channel_lock_both() was inlined.
+
+ * main/pbx.c: * Add more information to some messages in
+ __ast_pbx_run(). * Simplify some dialplan priority setting code
+ in ast_explicit_goto() because of opaquification.
+
+2012-04-20 14:50 +0000 [r362817] Terry Wilson <twilson@digium.com>
+
+ * /, 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) ........
+ Merged revisions 362815 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 362816 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-20 00:57 +0000 [r362779] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, UPGRADE.txt, include/asterisk/channel.h, CHANGES,
+ channels/sig_pri.c, funcs/func_callerid.c: Add original party id
+ and reason support. ISDN ETSI PTP and Q.SIG (And SS7 in future)
+ have support for reporting who was the original redirecting party
+ of a call. * Added support for the original redirecting party and
+ reason to the REDIRECTING function and the system core as well as
+ to the stubbed locations in sig_pri.c. Review:
+ https://reviewboard.asterisk.org/r/1829/
+
+2012-04-19 22:01 +0000 [r362731] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * funcs/func_version.c, /: Fix documentation for
+ ${VERSION(ASTERISK_VERSION_NUM)}. ........ Merged revisions
+ 362729 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 362730 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-19 21:14 +0000 [r362682] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, tests/test_linkedlists.c, tests/test_poll.c: Add leading and
+ trailing backslashes A couple of unit tests did not have have
+ leading or trailing backslashes when setting their test category
+ resulting in a warning message being displayed. Added the
+ backslash where needed. ........ Merged revisions 362680 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 362681 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-19 21:01 +0000 [r362679] Richard Mudgett <rmudgett@digium.com>
+
+ * /, configs/queues.conf.sample: Update membermacro and membergosub
+ documentation in queues.conf.sample. ........ Merged revisions
+ 362677 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 362678 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-19 19:05 +0000 [r362635] Terry Wilson <twilson@digium.com>
+
+ * addons/chan_ooh323.c, apps/app_alarmreceiver.c,
+ channels/iax2-provision.c, res/snmp/agent.c: Convert some
+ strncpys to ast_copy_string Review:
+ https://reviewboard.asterisk.org/r/1732/
+
+2012-04-19 16:10 +0000 [r362588] Sean Bright <sean@malleable.com>
+
+ * /, apps/app_externalivr.c: Prevent a crash in ExternalIVR when
+ the 'S' command is sent first. If the first command sent from an
+ ExternalIVR client is an 'S' command, we were blindly removing
+ the first element from the play list and deferencing it, even if
+ it was NULL. This corrects that and also locks appropriately in
+ one place. (issue ASTERISK-17889) Reported by: Chris Maciejewski
+ ........ Merged revisions 362586 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 362587 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-19 14:35 +0000 [r362538] Terry Wilson <twilson@digium.com>
+
+ * /, main/asterisk.c: Handle multiple commands per connection via
+ netconsole Asterisk would accept multiple NULL-delimited CLI
+ commands via the netconsole socket, but would occasionally miss a
+ command due to the command not being completely read into the
+ buffer. This patch ensures that any partial commands get moved to
+ the front of the read buffer, appended to, and properly sent.
+ (closes issue ASTERISK-18308) Review:
+ https://reviewboard.asterisk.org/r/1876/ ........ Merged
+ revisions 362536 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 362537 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-19 02:40 +0000 [r362497] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_unistim.c, /, main/tdd.c, main/jitterbuf.c,
+ apps/app_sms.c, main/stdtime/localtime.c, utils/extconf.c,
+ addons/chan_mobile.c, main/format_pref.c, main/asterisk.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. * format_pref: 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 ........ Merged revisions 362485 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 362496 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-18 17:03 +0000 [r362432] Michael L. Young <elgueromexicano@gmail.com>
+
+ * tests/test_security_events.c: Fix building security events test
+ The Security Events Framework API changed in trunk to support
+ IPv6. This broke the building of the security events test which
+ was based around IPv4. This patches fixes the build by changing
+ the test to conform to the new changes. (related to issue
+ ASTERISK-19447) Review: https://reviewboard.asterisk.org/r/1874/
+
+2012-04-18 16:41 +0000 [r362430] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, /, channels/sig_pri.c: 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 ........ Merged revisions 362428 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 362429 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-17 21:23 +0000 [r362365-362380] Matthew Jordan <mjordan@digium.com>
+
+ * /, main/format_pref.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/ ........ Merged
+ revisions 362377 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * res/res_rtp_asterisk.c, /, res/res_agi.c, res/res_musiconhold.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/ ........ Merged
+ revisions 362362 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 362364 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-17 21:10 +0000 [r362363] Jonathan Rose <jrose@digium.com>
+
+ * 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/ ........ Merged
+ revisions 362354 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 362357 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-17 21:08 +0000 [r362358-362361] Matthew Jordan <mjordan@digium.com>
+
+ * main/manager.c, /, main/asterisk.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. (issue ASTERISK-19655) Reported by: Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........
+ Merged revisions 362359 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 362360 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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/ ........ Merged
+ revisions 362355 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 362356 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-17 18:57 +0000 [r362307] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * channels/chan_unistim.c, cdr/cdr_sqlite3_custom.c,
+ funcs/func_env.c, res/res_phoneprov.c, channels/chan_gtalk.c,
+ cdr/cdr_pgsql.c, res/res_http_post.c, res/res_musiconhold.c,
+ res/res_jabber.c, res/res_format_attr_celt.c,
+ channels/chan_dahdi.c, funcs/func_groupcount.c,
+ apps/app_osplookup.c, funcs/func_odbc.c, main/ast_expr2f.c,
+ apps/app_minivm.c, channels/chan_alsa.c, codecs/codec_resample.c,
+ formats/format_h264.c, res/res_format_attr_silk.c,
+ res/res_config_ldap.c, main/ast_expr2.fl,
+ res/res_config_sqlite3.c, channels/chan_sip.c,
+ channels/vcodecs.c, codecs/codec_g726.c, main/data.c,
+ res/res_corosync.c, channels/chan_h323.c, codecs/codec_dahdi.c,
+ funcs/func_callerid.c, main/asterisk.c, res/res_odbc.c: Avoid
+ cppcheck warnings; removing unused vars and a bit of cleanup.
+ Patch by: junky Review: https://reviewboard.asterisk.org/r/1743/
+
+2012-04-17 18:29 +0000 [r362306] Matthew Jordan <mjordan@digium.com>
+
+ * /, formats/format_sln.c, formats/format_vox.c,
+ formats/format_wav.c, formats/format_pcm.c,
+ formats/format_wav_gsm.c, formats/format_siren14.c,
+ formats/format_gsm.c, formats/format_g719.c,
+ formats/format_siren7.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
+ ........ Merged revisions 362304 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 362305 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-17 15:00 +0000 [r362266] Michael L. Young <elgueromexicano@gmail.com>
+
+ * /, channels/chan_sip.c: Turn off warning message when bind
+ address is set to any. When a bind address is set to an ANY
+ address (udpbindport=::), a warning message is displayed stating
+ that "Address remapping activated in sip.conf but we're using
+ IPv6, which doesn't need it. Please remove 'localnet' and/or
+ 'externaddr' settings." But if one is running dual stack, we
+ shouldn't be told to turn those settings off. This patch checks
+ if the bind address is an ANY address or not. The warning message
+ will now only be displayed if the bind address is NOT an ANY
+ address and IPv6 is being used. Also, updated the copyright year.
+ (closes issue ASTERISK-19456) Reported by: Michael L. Young
+ Tested by: Michael L. Young Patches: chan_sip_ipv6_message.diff
+ uploaded by Michael L. Young (license 5026) ........ Merged
+ revisions 362253 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 362264 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-16 21:58 +0000 [r362203-362206] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_dahdi.c, /, channels/chan_agent.c: Fix negative
+ return handling in channel drivers In chan_agent, while handling
+ a channel indicate, the agent channel driver must obtain a lock
+ on both the agent channel, as well as the channel the agent
+ channel is using. To do so, it attempts to lock the other channel
+ first, then unlock the agent channel which is locked prior to
+ entry into the indicate handler. If this unlock fails with a
+ negative return value, which can occur if the object passed to
+ agent_indicate is an invalid ao2 object or is NULL, the return
+ value is passed directly to strerror, which can only accept
+ positive integer values. In chan_dahdi, the return value of
+ dahdi_get_index is used to directly index into the sub-channel
+ array. If dahd_get_index returns a negative value, it would use
+ that value to index into the array, which could cause an invalid
+ memory access. If dahdi_get_index returns a negative number, we
+ now default to SUB_REAL. (issue ASTERISK-19655) Reported by: Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........
+ Merged revisions 362204 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 362205 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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 ........
+ Merged revisions 362201 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 362202 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-16 21:20 +0000 [r362200] Michael L. Young <elgueromexicano@gmail.com>
+
+ * main/manager.c, main/security_events.c,
+ channels/sip/security_events.c, CHANGES,
+ include/asterisk/security_events_defs.h: Add IPv6 address support
+ to security events framework. The current Security Events
+ Framework API only supports IPv4 when it comes to generating
+ security events. This patch does the following: * Changes the
+ Security Events Framework API to support IPV6 and updates the
+ components that use this API. * Eliminates an error message that
+ was being generated since the current implementation was treating
+ an IPv6 socket address as if it was IPv4. * Some copyright dates
+ were updated on files touched by this patch. (closes issue
+ ASTERISK-19447) Reported by: Michael L. Young Tested by: Michael
+ L. Young Patches: security_events_ipv6v3.diff uploaded by Michael
+ L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/1777/
+
+2012-04-16 20:17 +0000 [r362153] Matthew Jordan <mjordan@digium.com>
+
+ * formats/format_ilbc.c, /, formats/format_sln.c,
+ formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c,
+ formats/format_g723.c, formats/format_h263.c,
+ formats/format_h264.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: 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/ ........ Merged
+ revisions 362151 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 362152 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-13 16:12 +0000 [r362081-362085] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_forkcdr.c, /: Make ForkCDR e option not set end time of
+ the newly forked CDR log Prior to this patch, ForkCDR's e option
+ would immediately set the end time of the forked CDR to that of
+ the CDR that is being terminated. This resulted in the new CDR's
+ end time being roughly the same as it's beginning time (which is
+ in turn roughly the same as the original's end time). (closes
+ issue ASTERISK-19164) Reported by: Steve Davies Patches:
+ cdr_fork_end.v10.patch uploaded by Steve Davies (license 5012)
+ ........ Merged revisions 362082 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 362084 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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/ ........ Merged
+ revisions 362079 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 362080 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-12 20:08 +0000 [r362043] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * main/srv.c: Convert SRV lookup message to debug level This helps
+ clean up the Asterisk CLI by converting the log message from
+ verbose to debug
+
+2012-04-12 16:29 +0000 [r361998] Richard Mudgett <rmudgett@digium.com>
+
+ * configs/asterisk.conf.sample, UPGRADE.txt, pbx/pbx_config.c,
+ include/asterisk/options.h, main/asterisk.c: Add option to invoke
+ the extensions.conf stdexten using the legacy macro method.
+ ASTERISK-18809 eliminated the legacy macro invocation of the
+ stdexten in favor of the Gosub method without a means of
+ backwards compatibility. (issue ASTERISK-18809) (closes issue
+ ASTERISK-19457) Reported by: Matt Jordan Tested by: rmudgett
+ Review: https://reviewboard.asterisk.org/r/1855/
+
+2012-04-12 16:25 +0000 [r361968-361987] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_iax2.c: Make trunkfreq take effect when set
+ Previously, setting trunkfreq had no effect on initial load or on
+ reload and only ever used the default value. This causes
+ trunkfreq to be used appropriately on initial load and reload.
+ (closes issue ASTERISK-19521) Patch-by: Jaco Kroon ........
+ Merged revisions 361972 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 361981 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * Makefile, build_tools/cflags.xml, /,
+ build_tools/menuselect-deps.in, codecs/gsm/src/k6opt.s,
+ configure, codecs/gsm/Makefile, configure.ac, Makefile.rules,
+ makeopts.in, codecs/lpc10/Makefile: 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) ........ Merged revisions 361955 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 361956 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-11 17:20 +0000 [r361909] Jonathan Rose <jrose@digium.com>
+
+ * /, configs/queues.conf.sample, apps/app_queue.c: Change default
+ value of 'ignorebusy' on Queue members so that behavior is more
+ like 1.8 Prior to this patch, in order to restore that behavior,
+ a function would have to be used on the QueueMember to make the
+ ringinuse option do anything, which is pretty unreasonable.
+ (closes issue ASTERISK-19536) reported by: Philippe Lindheimer
+ Review: https://reviewboard.asterisk.org/r/1860/ ........ Merged
+ revisions 361907 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-10 21:50 +0000 [r361856] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Prevent invalid access of free'd memory
+ if DAHDI channel during an MWI event In the MWI processing loop,
+ when a valid event occurs the temporary caller ID information is
+ deallocated. If a new DAHDI channel is successfully created, the
+ event is passed up to the analog_ss_thread without error and the
+ loop exits. If, however, the DAHDI channel is not created, then
+ the caller ID struct has been free'd, and the gains reset to
+ their previous level. This will almost certainly cause an invalid
+ access to the free'd memory, either in subsequent calls to
+ callerid_free or calls to callerid_feed. * Rework the -r361705
+ patch to better manage the cs and mtd allocated resources. *
+ Fixed use of mwimonitoractive flag to be correct if the
+ mwi_thread() fails to start. ........ Merged revisions 361854
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 361855 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-10 19:58 +0000 [r361659-361805] Matthew Jordan <mjordan@digium.com>
+
+ * /, main/http.c: Fix crash caused by unloading or reloading of
+ res_http_post When unlinking itself from the registered HTTP
+ URIs, res_http_post could inadvertently free all URIs registered
+ with the HTTP server. This patch modifies the unregister method
+ to only free the URI that is actually being unregistered, as
+ opposed to all of them. ........ Merged revisions 361803 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 361804 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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. ........ Merged revisions 361753
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 361754 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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. ........ Merged revisions 361705 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 361706 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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. ........ Merged revisions 361657 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 361658 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-06 22:00 +0000 [r361561-361608] Matthew Jordan <mjordan@digium.com>
+
+ * /, res/res_calendar_ews.c: Fix memory leak in res_calendar_ews
+ when event email address node is empty If the XML calendar data
+ returned by a Microsoft Exchange Web Service specifies an XML
+ Event E-Mail Address ("EmailAddress"), and no e-mail address is
+ provided, a condition existed where an ast_calendar_attendee
+ struct would be allocated but not appended to the list of
+ attendees. Because of that, the memory associated with the
+ attendee would never be freed. This patch frees the memory if no
+ e-mail address is provided. ........ Merged revisions 361606 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 361607 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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. ........ Merged revisions 361558 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 361560 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-06 19:58 +0000 [r361523] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/message.c: Don't add an empty MESSAGE_DATA(key) header if
+ it doesn't already exist. Doing Set(MESSAGE_DATA(key)=) would add
+ an empty key header if the key header did not already exist. If
+ it already existed it would delete it. * Made msg_set_var_full()
+ exit early if the named variable did not already exist and the
+ value to set is empty. ........ Merged revisions 361522 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-06 18:19 +0000 [r361476] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_unistim.c, main/pbx.c, /, channels/chan_sip.c,
+ funcs/func_strings.c, formats/format_ogg_vorbis.c,
+ channels/console_video.c, 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, channels/console_gui.c,
+ funcs/func_channel.c, apps/app_sms.c, addons/chan_mobile.c,
+ apps/app_chanspy.c, main/xmldoc.c, channels/chan_mgcp.c,
+ res/res_config_sqlite3.c, res/res_clioriginate.c,
+ apps/app_voicemail.c: Add missing newlines to CLI logging
+ ........ Merged revisions 361471 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 361472 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-06 16:33 +0000 [r361429] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * bridges/bridge_builtin_features.c, /, funcs/func_sysinfo.c,
+ bridges/bridge_multiplexed.c: Multiple revisions 361403,361412
+ ........ r361403 | pabelanger | 2012-04-06 12:24:36 -0400 (Fri,
+ 06 Apr 2012) | 2 lines Fix typo in svn:keywords ........ r361412
+ | pabelanger | 2012-04-06 12:27:30 -0400 (Fri, 06 Apr 2012) | 2
+ lines Fix typo in svn:keywords ........ Merged revisions
+ 361403,361412 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 361422 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-06 15:50 +0000 [r361382] Russell Bryant <russell@russellbryant.com>
+
+ * /, configs/rpt.conf.sample (removed),
+ configs/usbradio.conf.sample (removed), apps/rpt_flow.pdf
+ (removed): Remove a few more files related to chan_usbradio and
+ app_rpt. ........ Merged revisions 361380 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 361381 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-06 14:02 +0000 [r361334] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Fix a typo in the warning messages for an
+ ignored media stream Added a '\n' to the warning messages when we
+ ignore a media stream due to the port number being '0'. (closes
+ issue ASTERISK-19646) Reported by: Badalian Vyacheslav ........
+ Merged revisions 361332 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 361333 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-06 13:32 +0000 [r361331] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_dial.c, /: Remove unnecessary error message in
+ app_dial.c The error message for failure to stop autoservice
+ after a gosub or macro call during a dial was removed for macro
+ while Asterisk 1.4 was still being actively developed. The
+ corresponding gosub error message was never removed. (closes
+ issue ASTERISK-19551) ........ Merged revisions 361329 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 361330 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-05 17:22 +0000 [r361092-361279] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_meetme.c: Fix MusicOnHold in MeetMe so that it always
+ uses the class if it's been defined There were a few instances of
+ restarting music on hold in meetme that would cause Asterisk to
+ revert to the default class of music on hold for no adequate
+ reason. Review: https://reviewboard.asterisk.org/r/1844/ ........
+ Merged revisions 361269 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 361270 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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) ........ Merged revisions
+ 361210 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 361211 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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) ........ Merged revisions 361201 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 361208 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/channel.c, pbx/pbx_loopback.c, addons/chan_ooh323.c, /,
+ channels/chan_sip.c, main/app.c, pbx/pbx_realtime.c,
+ apps/app_externalivr.c, channels/chan_iax2.c,
+ res/res_fax_spandsp.c, apps/app_milliwatt.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)
+ ........ Also add from the patch the portion in res_fax_spandsp
+ that didn't apply to 1.8 Merged revisions 361142 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 (closes issue
+ ASTERISK-19540) ........ Merged revisions 361143 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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/
+ ........ Merged revisions 361090 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 361091 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-03 20:14 +0000 [r361042] Kinsey Moore <kmoore@digium.com>
+
+ * /, apps/app_transfer.c: Fix the display of documentation for
+ Transfer This came up while fixing documentation generation for
+ many other cases where the argument separator was not being
+ displayed properly. Now that it is displayed properly, it shows
+ up in the wrong place for Transfer since the '/' is only required
+ if Tech is present. (related to issue ASTERISK-18168) ........
+ Merged revisions 361040 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 361041 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-04-03 20:03 +0000 [r361038-361039] Mark Murawki <markm@intellasoft.net>
+
+ * include/asterisk/manager.h: Fix dev-mode compiler warning about
+ gnu_printf (related to ASTERISK-19575)
+
+ * main/channel.c, main/manager.c, main/utils.c,
+ include/asterisk/channel.h, include/asterisk/strings.h, CHANGES,
+ include/asterisk/manager.h: Allow the Hangup manager action to
+ match channels by regex * Hangup now can take a regular
+ expression as the Channel option. If you want to hangup multiple
+ channels, use /regex/ as the Channel option. Existing behavior to
+ hanging up a single channel is unchanged, but if you pass a
+ regex, the manager will send you a list of channels back that
+ were hung up. (closes issue ASTERISK-19575) Reported by: Mark
+ Murawski Tested by: Mark Murawski
+
+2012-04-02 22:27 +0000 [r360994] Kinsey Moore <kmoore@digium.com>
+
+ * /, 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)
+ ........ Merged revisions 360987 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 360993 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-30 21:38 +0000 [r360935] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/logger.c: Fix logger deadlock on Asterisk shutdown. The
+ logger_thread() had an exit path that failed to release the
+ logmsgs list lock. * Make logger_thread() exit path unlock the
+ logmsgs list lock. * Made ast_log() not queue any messages to the
+ logmsgs list if the close_logger_thread flag is set. (issue
+ ASTERISK-19463) Reported by: Matt Jordan ........ Merged
+ revisions 360933 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 360934 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-29 23:36 +0000 [r360872-360886] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/features.c: Fix potential race condition during call
+ pickup. Prior to this patch, a connected line update was queued
+ during call pickup and then an answer frame was queued. The
+ original caller would presumably then have his connected line
+ updated and then the call would be answered. In actuality, the
+ answer frame was not how the call ended up being answered.
+ Rather, an odd section in app_dial that checks if the called
+ channel's state is up. The result is that the order of the
+ connected line update and the answer were variable. In most
+ cases, this wasn't actually a bad thing. However, if the 'I'
+ option was passed to dial, the connected line update would be
+ inhibited. The fix is to queued the connected line after the
+ answer frame is queued. This way the race in app_dial is between
+ two conditions resulting in an answer. This way the connected
+ line update occurs after the answer every time. (closes issue
+ ASTERISK-19183) Reported by: Thomas Arimont Tested by: Thomas
+ Arimont Mark Michelson Patches: ASTERISK-19183.patch uploaded by
+ Mark Michelson (license 5049) ........ Merged revisions 360884
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 360885 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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)
+ ........ Merged revisions 360862 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 360863 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-29 21:57 +0000 [r360827] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/astobj2.h, main/astobj2.c: Misc changes to make
+ astobj2 enhancement diffs easier to follow. * Rename astobj2 API
+ parameter funcname to func. * Rename astobj2 API iterator
+ parameter to iter. * Update some documentation for OBJ_MULTIPLE.
+
+2012-03-29 20:01 +0000 [r360785-360787] Jonathan Rose <jrose@digium.com>
+
+ * include/asterisk/logger.h, main/dial.c, main/pbx.c,
+ include/asterisk/bridging.h, main/features.c, main/logger.c,
+ CHANGES, apps/app_mixmonitor.c, configs/logger.conf.sample:
+ Introducing the log message unique call identifiers feature Log
+ messages will now display a call number that they are tied to
+ (ordered for calls based on when they started). This feature is
+ made to be minimally invasive without requiring changes to many
+ of the existing log messages. These IDs won't show up for verbose
+ messages on CLI (but they will in log files) This is currently in
+ phase II of production, see more about this feature on the wiki
+ --
+ https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging
+ Review: https://reviewboard.asterisk.org/r/1823/
+
+ * include/asterisk/logger.h, main/dial.c, main/pbx.c, /,
+ include/asterisk/bridging.h, main/features.c, main/logger.c,
+ CHANGES, apps/app_mixmonitor.c, configs/logger.conf.sample:
+ undoing 360785 due to merging mistake
+
+ * include/asterisk/logger.h, main/dial.c, main/pbx.c, /,
+ include/asterisk/bridging.h, main/features.c, main/logger.c,
+ CHANGES, apps/app_mixmonitor.c, configs/logger.conf.sample:
+ Introducing the log message unique call identifiers feature Log
+ messages will now display a call number that they are tied to
+ (ordered for calls based on when they started). This feature is
+ made to be minimally invasive without requiring changes to many
+ of the existing log messages. These IDs won't show up for verbose
+ messages on CLI (but they will in log files) This is currently in
+ phase II of production, see more about this feature on the wiki
+ --
+ https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging
+ Review: https://reviewboard.asterisk.org/r/1823/
+
+2012-03-28 19:39 +0000 [r360724] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_jingle.c, addons/chan_ooh323.c,
+ cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c,
+ channels/chan_gtalk.c, apps/confbridge/conf_config_parser.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/ ........ Merged
+ revisions 358978 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 358989 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-27 18:44 +0000 [r360673] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Make a debug message regarding
+ subscription changes more accurate. I was getting confused during
+ some testing why Asterisk was saying that a subscription was
+ being added when it was clearly being removed. This fixes that
+ confusion. ........ Merged revisions 360625 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 360672 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-27 17:13 +0000 [r360626-360627] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/astobj2.h, tests/test_astobj2.c, main/astobj2.c:
+ Add global ao2 array container. Global ao2 objects must always
+ exist after initialization because there is no access control to
+ obtain another reference to the global object. It is expected
+ that module configuration could use these new API calls to
+ replace an active configuration parameter object with an updated
+ configuration parameter object. With these new API calls, the
+ global object could be replaced, removed, or referenced without
+ the risk of someone using a stale global object pointer. Review:
+ https://reviewboard.asterisk.org/r/1824/
+
+ * main/astobj2.c: Attempt to be more helpful when using a bad ao2
+ object pointer.
+
+2012-03-27 14:43 +0000 [r360576] Jonathan Rose <jrose@digium.com>
+
+ * /, configure: Updates config with bootstrap where I changed
+ configure.ac in r360488 (issue ASTERISK-17842) Reported by: Bryon
+ Clark ........ Merged revisions 360574 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 360575 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-26 21:22 +0000 [r360536] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * main/dnsmgr.c, /: Convert ast_verb() to ast_debug() and increase
+ log level Rather then flood the CLI with verbose messages, we've
+ changed the level to debug. This will help keep the CLI clean.
+
+2012-03-26 19:49 +0000 [r360490] Jonathan Rose <jrose@digium.com>
+
+ * /, 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) ........ Merged revisions
+ 360488 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 360489 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-24 23:49 +0000 [r360359-360415] Russell Bryant <russell@russellbryant.com>
+
+ * funcs/func_curl.c, /: func_curl: Fix leak of an ast_str in error
+ handling code path. ........ Merged revisions 360413 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 360414 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/chan_iax2.c: chan_iax2: Use OBJ_NODATA to be a bit more
+ explicit. This is just a minor code cleanup change. These uses of
+ ao2_callback() would never return anything since the callbacks
+ always returned 0. However, be more explicit that no returned
+ results are wanted by specifying OBJ_NODATA.
+
+ * /, 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. ........ Merged revisions 360363 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 360364 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, apps/app_jack.c: app_jack: fix datastore memory leak in error
+ handling path. ........ Merged revisions 360360 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 360361 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, main/ast_expr2.h, res/ael/ael.tab.c, main/ast_expr2.y,
+ main/ast_expr2f.c, res/ael/ael_lex.c, res/ael/ael.tab.h,
+ main/ast_expr2.c: Multiple revisions 360356-360357 ........
+ r360356 | russell | 2012-03-23 22:33:36 -0400 (Fri, 23 Mar 2012)
+ | 6 lines 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. ........ r360357 |
+ russell | 2012-03-23 22:34:39 -0400 (Fri, 23 Mar 2012) | 6 lines
+ 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. ........ Merged revisions 360356-360357 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 360358 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-24 00:40 +0000 [r360264-360311] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /, channels/sig_pri.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. ........ Merged
+ revisions 360309 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 360310 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, channels/chan_sip.c: Add missing initialization of
+ update_redirecting in chan_sip.c ........ Merged revisions 360262
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 360263 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-22 21:25 +0000 [r360227] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_dial.c, include/asterisk/utils.h, main/features.c,
+ main/utils.c, CHANGES, apps/app_queue.c: Adds F option to Bridge
+ application Similar to dial and queue F option. (Closes issue
+ ASTERISK-19282) Reported by: To Patches: bridge_f-v3.diff
+ uploaded by To (license 6347) Review:
+ https://reviewboard.asterisk.org/r/1825/
+
+2012-03-22 19:51 +0000 [r360190] Kinsey Moore <kmoore@digium.com>
+
+ * main/udptl.c, main/stdtime/test.c, main/autoservice.c,
+ main/rtp_engine.c, main/frame.c, main/fskmodem_float.c,
+ main/sha1.c, main/say.c, main/ecdisa.h, main/utils.c,
+ main/devicestate.c, main/taskprocessor.c, main/indications.c,
+ main/enum.c, main/config.c, main/loader.c, main/term.c,
+ main/cli.c, main/io.c, main/ulaw.c, main/channel.c, main/dial.c,
+ main/manager.c, main/tdd.c, main/strcompat.c, main/plc.c,
+ main/features.c, main/logger.c, main/fskmodem_int.c, main/app.c,
+ main/stdtime/localtime.c, main/image.c, main/dns.c,
+ main/message.c, main/md5.c, main/sched.c, main/lock.c,
+ main/pbx.c, main/dnsmgr.c, main/slinfactory.c, main/translate.c,
+ main/jitterbuf.c, main/cel.c, main/chanvars.c, main/netsock.c,
+ main/srv.c, main/privacy.c, main/fixedjitterbuf.c, main/file.c,
+ main/callerid.c, main/event.c, main/astmm.c, main/audiohook.c,
+ main/cygload.c, main/fixedjitterbuf.h, main/asterisk.c,
+ main/xmldoc.c, main/dsp.c, main/timing.c: Kill off red blobs in
+ most of main/* Everything still compiled after making these
+ changes, so I assume these whitespace-only changes didn't break
+ anything (and shouldn't have).
+
+2012-03-21 14:55 +0000 [r360140] Jonathan Rose <jrose@digium.com>
+
+ * /, contrib/scripts/install_prereq: Update install_prereq script
+ to include missing GSM library for debian amd move SQLite3.
+ (closes issue ASTERISK-19367) Reported by: Andrew Latham Patches:
+ debian_install_prereq.diff uploaded by Andrew Latham (license
+ 5985) ........ Merged revisions 360138 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 360139 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-21 14:47 +0000 [r360137] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * /, configure, configure.ac: Also detect gmime 2.6 Also detect
+ gmime version 2.6 (Michael Biebl) Signed-off-by: Tzafrir Cohen
+ (License #5035) <tzafrir.cohen@xorcom.com> ........ Merged
+ revisions 360087 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 360098 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-21 13:31 +0000 [r360089] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Ensure Asterisk sends a BYE when pending
+ on the final response to a re-INVITE When Asterisk detects a
+ hangup and cannot send a BYE due to a pending INVITE, it sets the
+ pendingbye flag and waits for the final response to that INVITE.
+ When the response is received, it transmits the BYE. If, however,
+ that INVITE request is a pending re-INVITE, it needs to first
+ send a CANCEL request to terminate the pending re-INVITE. In that
+ circumstance, Asterisk was, in some scenarios, clearing the
+ pendingbye flag after processing the CANCEL request and not
+ checking for a pending BYE when receiving the final 487 response
+ to the INVITE. This patch ensures that if the pendingbye flag is
+ set, it is honored regardless of the nature of the INVITE request
+ currently in flight. (closes issue ASTERISK-19365) Reported by:
+ Thomas Arimont Tested by: Thomas Arimont Patches:
+ bugASTERISK-19365_2012_03_08.patch uploaded by mjordan (license
+ 6283) Review: https://reviewboard.asterisk.org/r/1807 ........
+ Merged revisions 360086 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 360088 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-20 20:42 +0000 [r360036] Kinsey Moore <kmoore@digium.com>
+
+ * /, apps/app_echo.c: Prevent Echo() from relaying control, null,
+ and modem frames Echo()'s description states that it echoes
+ audio, video, and DTMF except for # while it actually echoes any
+ frame that it receives other than DTMF #. This was causing frame
+ storms in the test suite in some circumstances where Echo() was
+ attached to both ends of a pair of local channels and control
+ frames were being periodically generated. Echo()'s behavior and
+ description have been modifed so that it only echoes media and
+ non-# DTMF frames. ........ Merged revisions 360033 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 360034 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-20 18:17 +0000 [r359983] Sean Bright <sean@malleable.com>
+
+ * /, UPGRADE.txt, channels/chan_iax2.c, include/asterisk/manager.h:
+ chan_iax2: Correct spelling of 'Port' header in IAX2 PeerStatus
+ AMI Events The PeerStatus event for IAX2 channels currently
+ includes a header named Post which should have been Port. Post
+ was removed and the AMI version has been updated to 1.3. ........
+ Merged revisions 359982 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-20 17:31 +0000 [r359942-359981] Richard Mudgett <rmudgett@digium.com>
+
+ * main/data.c, main/pbx.c, main/manager.c, /, main/features.c,
+ include/asterisk/manager.h, main/db.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/ ........ Merged
+ revisions 359979 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 359980 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * res/res_mutestream.c: Convert MuteAudio documentation to XML. *
+ Added missing error exits with cause in manager_mutestream(). *
+ Cleaned up manager_mutestream() and func_mute_write(). * Some
+ whitespace and comment cleanup.
+
+2012-03-16 21:00 +0000 [r359905] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_chanspy.c: Prevent chanspy from binding to zombie
+ channels This patch addresses a bug with chanspy on local
+ channels which roughly 50% of the time would create a situation
+ where chanspy can latch onto a zombie channel, keeping the zombie
+ alive forever and causing the channel doing the spying to never
+ be able to hang up. (closes issue ASTERISK-19493) Reported by:
+ lvl Review: https://reviewboard.asterisk.org/r/1819/ ........
+ Merged revisions 359892 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 359898 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-16 20:37 +0000 [r359904] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/app.h, main/app.c: Simplify some code in
+ ast_app_run_sub(). * Remove unnnecessary const from const char *
+ const var declaration in the ast_app_run_macro() and
+ ast_app_run_sub() prototypes. The second const is unnecessary.
+
+2012-03-16 15:38 +0000 [r359857] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h, CHANGES:
+ Revert the pre-dial addition. The code may be just fine, but it
+ had not received a "ship it!" on review board yet.
+
+2012-03-16 08:27 +0000 [r359811] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, channels/sip/include/sip.h: Missed lastinvite CSeq int to
+ uint32_t change from Review:
+ https://reviewboard.asterisk.org/r/1699/ ........ Merged
+ revisions 359809 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 359810 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-15 20:11 +0000 [r359772] Mark Murawki <markm@intellasoft.net>
+
+ * main/pbx.c: Fix warning from commit r359705 (predial options for
+ app_dial)
+
+2012-03-15 19:11 +0000 [r359708] Matthew Jordan <mjordan@digium.com>
+
+ * /, main/utils.c: Fix remotely exploitable stack overflow in HTTP
+ manager There exists a remotely exploitable stack buffer overflow
+ in HTTP digest authentication handling in Asterisk. The
+ particular method in question is only utilized by HTTP AMI. When
+ parsing the digest information, the length of the string is not
+ checked when it is copied into temporary buffers allocated on the
+ stack. This patch fixes this behavior by parsing out pre-defined
+ key/value pairs and avoiding unnecessary copies to the stack.
+ (closes issue ASTERISK-19542) Reported by: Russell Bryant Tested
+ by: Matt Jordan ........ Merged revisions 359706 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 359707 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-15 18:58 +0000 [r359705] Mark Murawki <markm@intellasoft.net>
+
+ * apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h, CHANGES: Add
+ options PreDial options 'b' and 'B' to app_dial * Added 'b' and
+ 'B' options to Dial. These options will allow you to run
+ last-minute dialplan on the caller and callee channels while the
+ Dial application is executing, but before the call is started.
+ For example you can use the 'b' option to run dialplan on the
+ callee channel to get the name of the newly created channel right
+ away. Review: https://reviewboard.asterisk.org/r/1229/ (closes
+ issue: ASTERISK-19548) Reported by: Mark Murawski Tested by: Mark
+ Murawski, Stefan Schmidt
+
+2012-03-15 18:55 +0000 [r359704] Matthew Jordan <mjordan@digium.com>
+
+ * /, 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
+ ........ Merged revisions 359656 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 359694 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-15 18:34 +0000 [r359651] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * channels/chan_sip.c: Remove unused variable ‘srch’ Missed on the
+ previous commit
+
+2012-03-15 18:32 +0000 [r359644] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, /, apps/app_queue.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/
+ ........ Merged revisions 359609 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 359620 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-15 17:36 +0000 [r359607] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * channels/chan_sip.c: Remove some dead code found in
+ _sip_show_peers() Review:
+ https://reviewboard.asterisk.org/r/1696/
+
+2012-03-15 00:54 +0000 [r359456-359560] Russell Bryant <russell@russellbryant.com>
+
+ * /, channels/chan_iax2.c: chan_iax2: Fix use of uninitialized
+ sockaddr_in in try_transfer(). Initialize a struct sockaddr_in in
+ try_transfer() so that the code isn't (potentially) trying to
+ read from it while uninitialized. ........ Merged revisions
+ 359558 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 359559 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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. ........ Merged
+ revisions 359508 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 359509 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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. ........ Merged
+ revisions 359486 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 359491 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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). ........ Merged revisions
+ 359457 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 359458 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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. ........ Merged revisions 359452 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 359454 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-14 22:38 +0000 [r359455] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /, channels/chan_agent.c,
+ include/asterisk/channel.h: 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. ........ Merged revisions 359451
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 359453 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-14 18:56 +0000 [r359406] Matthew Jordan <mjordan@digium.com>
+
+ * tests/test_jitterbuf.c (added): Add tests for main/jitterbuf.c
+ This patch adds unit tests for main/jitterbuf.c. This includes
+ checking for the following: * Nominal insertion and retrieval of
+ frames * Insertion and retrieval of frames where the frames are
+ inserted out of order with respect to the previous frame *
+ Insertion and retrieval of frames where some number of frames
+ that would occur in the expected sequence are instead dropped *
+ Insertion and retrieval of frames with an arrival time that does
+ not occur at the same rate as the surrounding frames *
+ Resynchronization of the jitter buffer when an inserted frame
+ breaks the resynchronization threshold * Overfilling of the
+ jitter buffer For each of the tests, both JB_TYPE_VOICE and
+ JB_TYPE_CONTROL permutations exist. Review:
+ https://reviewboard.asterisk.org/r/1815 (issue: ASTERISK-18964)
+ Reported by: Kris Shaw Tested by: Kris Shaw, Matt Jordan
+
+2012-03-14 18:12 +0000 [r359360] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/channel_internal.h: Three copies of the file
+ contents in channel_internal.h are a bit excessive.
+
+2012-03-14 17:48 +0000 [r359359] Matthew Jordan <mjordan@digium.com>
+
+ * /, main/jitterbuf.c: Fix incorrect jitter buffer overflow due to
+ missed resynchronizations When a change in time occurs, such that
+ the timestamps associated with frames being placed into an
+ adaptive jitter buffer (implemented in jitterbuf.c) are
+ significantly different then the previously inserted frames, the
+ jitter buffer checks to see if it needs to be resynched to the
+ new time frame. If three consecutive packets break the threshold,
+ the jitter buffer resynchs itself to the new timestamps. This
+ currently only occurs when history is calculated, and hence only
+ on JB_TYPE_VOICE frames. JB_TYPE_CONTROL frames, on the other
+ hand, are never passed to the history calculations. Because of
+ this, if the jump in time is greater then the maximum allowed
+ length of the jitter buffer, the JB_TYPE_CONTROL frames are
+ dropped and no resynchronization occurs. Alterntively, if the
+ overfill logic is not triggered, the JB_TYPE_CONTROL frame will
+ be placed into the buffer, but with a time reference that is not
+ applicable. Subsequent JB_TYPE_VOICE frames will quickly trigger
+ the overflow logic until reads from the jitter buffer reach the
+ errant JB_TYPE_CONTROL frame. This patch allows JB_TYPE_CONTROL
+ frames to resynch the jitter buffer. As JB_TYPE_CONTROL frames
+ are unlikely to occur in multiples, it perform the
+ resynchronization on any JB_TYPE_CONTROL frame that breaks the
+ resynch threshold. Note that this only impacts chan_iax2, as
+ other consumers of the adaptive jitter buffer use the abstract
+ jitter buffer API, which does not use JB_TYPE_CONTROL frames.
+ Review: https://reviewboard.asterisk.org/r/1814/ (closes issue
+ ASTERISK-18964) Reported by: Kris Shaw Tested by: Kris Shaw, Matt
+ Jordan Patches: jitterbuffer-2012-2-26.diff uploaded by Kris Shaw
+ (license 5722) ........ Merged revisions 359356 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 359358 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-14 17:39 +0000 [r359357] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, main/channel.c, /: Fix Dial m and r options and
+ forked calls generating warnings for voice frames. When connected
+ line support was added, the wait_for_answer() variable single
+ changed its meaning slightly. Unfortunately, the places where
+ single was used did not necessarily get updated to reflect that
+ change. Also audio/video frames were sent to all forked calls
+ when the endpoints were never made compatible. * Don't pass
+ audio/video media frames when the channels have not been made
+ compatible. * Added handling of AST_CONTROL_SRCCHANGE to
+ app_dial.c. * Fixed app_dial.c passing on AST_CONTROL_HOLD
+ because that frame can also pass a requested MOH class. (closes
+ issue ASTERISK-16901) Reported by: Chris Gentle (closes issue
+ ASTERISK-17541) Reported by: clint Review:
+ https://reviewboard.asterisk.org/r/1805/ ........ Merged
+ revisions 359344 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 359355 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-14 14:40 +0000 [r359306] Matthew Jordan <mjordan@digium.com>
+
+ * include/asterisk/astobj2.h: Force non-inlining of
+ ao2_iterator_destroy when TEST_FRAMEWORK is enabled In r357272,
+ astobj2 was changed to automatically enable REF_DEBUG when the
+ TEST_FRAMEWORK flag was enabled. Unfortunately, some compilers
+ (gcc 4.5.1 at least) will attempt to inline ao2_iterator_destroy
+ in handle_astobj2_test. This by itself is not a problem;
+ unfortunately, the compiler believes that there is a code path
+ wherein an object allocated on the stack will be free'd. As
+ warnings are treated as errors, this prevents compilation of
+ astobj2. This patch works around that by adding the noinline
+ attribue to ao2_iterator_destroy, but only if the TEST_FRAMEWORK
+ flag is enabled. Preventing inlining is only needed for the test
+ method defined in astobj2, which is also only enabled if
+ TEST_FRAMEWORK is enabled.
+
+2012-03-14 10:56 +0000 [r359052-359261] Russell Bryant <russell@russellbryant.com>
+
+ * include/asterisk/logger.h, /, main/logger.c: Fix bogus
+ reads/writes of console log levels in asterisk.c This patch
+ updates the NUMLOGLEVELS define in logger.h to 32, to match the
+ fact that logger.c implements 32 log levels (because of the
+ custom log level stuff). asterisk.c uses this define to size an
+ array of levels per remote console. This array is modified in
+ ast_console_toggle_loglevel(), which is called by the "logger set
+ level" CLI command. While the documentation for the CLI command
+ doesn't make it terribly obvious, you can use this CLI command to
+ toggle a custom log level on a remote console, as well. However,
+ doing so led to an invalid array index in asterisk.c. This array
+ is read from any time a log message is written to a console. So,
+ all custom log level messages resulted in a bogus read if a
+ remote console was connected. ........ Merged revisions 359259
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 359260 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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. ........ Merged revisions
+ 359211 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 359212 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, main/sched.c: Fix inaccurate sizeof() in sched.c. This code
+ just needed sizeof(int), not sizeof(int *). ........ Merged
+ revisions 359157 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 359162 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, utils/astman.c: Fix incorrect sizeof() in astman. ........
+ Merged revisions 359116 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 359117 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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.
+ ........ Merged revisions 359110 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 359114 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, res/res_adsi.c: Fix broken usage of sizeof() in res_adsi.
+ ........ Merged revisions 359088 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 359091 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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. ........ Merged revisions 359069 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 359072 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, main/md5.c: Fix incorrect sizeof() on a pointer in MD5Final().
+ ........ Merged revisions 359059 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 359060 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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. ........
+ Merged revisions 359056 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 359057 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/chan_usbradio.c (removed), /, channels/xpmr (removed),
+ build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, makeopts.in,
+ apps/app_rpt.c (removed): 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/ ........ Merged
+ revisions 359050 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 359051 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-13 21:24 +0000 [r359011] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/channel_internal.h (added): Add missing
+ channel_internal.h ...again.
+
+2012-03-13 21:18 +0000 [r358997] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, channels/sig_pri.c: Add ability
+ for chan_dahdi ISDN to block connected line updates per span.
+ Added new chan_dahdi.conf colp_send option parameter to block
+ connected line updates per span. (closes issue ASTERISK-17025)
+ Reported by: Michael Smith
+
+2012-03-13 20:43 +0000 [r358907-358993] Terry Wilson <twilson@digium.com>
+
+ * /, main/features.c: Fix setting CDR variables in the hangup
+ extension A previous CDR fix for setting CDR variables during a
+ bridge via custom dialplan features broke setting CDR variables
+ in the hangup extension. This patch fixes the issue. Review:
+ https://reviewboard.asterisk.org/r/1794/ ........ Merged
+ revisions 358978 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 358989 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * include/asterisk/devicestate.h, /, channels/chan_sip.c,
+ tests/test_devicestate.c, main/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/ ........ Merged
+ revisions 358943 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 358944 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * apps/app_voicemail.c: Fix IMAP storage compilation after
+ opaquification changes (closes issue ASTERISK-19513)
+
+ * channels/chan_unistim.c, main/autoservice.c,
+ channels/chan_vpb.cc, channels/chan_local.c, main/rtp_engine.c,
+ res/res_musiconhold.c, bridges/bridge_multiplexed.c,
+ apps/app_followme.c, main/indications.c, main/cli.c,
+ main/channel.c, channels/chan_phone.c, channels/chan_dahdi.c,
+ channels/sig_analog.c, main/manager.c, main/features.c,
+ apps/app_dumpchan.c, res/res_agi.c, main/app.c,
+ apps/app_confbridge.c, apps/app_externalivr.c, main/bridging.c,
+ apps/app_parkandannounce.c, apps/app_dial.c, main/pbx.c,
+ channels/chan_sip.c, channels/chan_bridge.c,
+ main/channel_internal_api.c, channels/chan_agent.c,
+ apps/app_disa.c, include/asterisk/channel.h,
+ apps/app_talkdetect.c, apps/app_queue.c, apps/app_speech_utils.c,
+ apps/app_channelredirect.c, main/file.c, res/snmp/agent.c,
+ apps/app_macro.c, apps/app_stack.c, apps/app_chanspy.c,
+ apps/app_mixmonitor.c: Finalize ast_channel opaquification
+ Review: https://reviewboard.asterisk.org/r/1786/
+
+2012-03-13 17:01 +0000 [r358858-358861] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c: Fix crash caused by opaquification change
+ -r356042. The set_format() function was more subtle in how it
+ modified the struct ast_channel readtrans/writetrans values. *
+ Fixed ast_activate_generator() conversion correctly. (closes
+ issue ASTERISK-19434) Reported by: Birger Harzenetter Tested by:
+ rmudgett
+
+ * main/format.c: Use struct copy instead of memcpy().
+
+2012-03-13 08:06 +0000 [r358812] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * res/ael/pval.c, funcs/func_dialplan.c, /, tests/test_gosub.c,
+ utils/ael_main.c, apps/app_stack.c, utils/conf2ael.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/ ........ Merged
+ revisions 358810 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 358811 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-12 17:01 +0000 [r358766] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * channels/chan_unistim.c, contrib/unistimLang/ru.po (added),
+ contrib/unistimLang/ru.po.utf8 (added),
+ configs/unistim.conf.sample, UPGRADE.txt, CHANGES,
+ contrib/unistimLang/en.po (added), contrib/unistimLang (added):
+ Massive changes in chan_unistim channel driver. Include many
+ fixes in channel driver operation and add additional
+ functionality: * Added ability to use multiple lines on phone, so
+ for one device in configuration multiple lines can be defined, it
+ allows to have multiple calls on one phone, callwaiting and
+ switching between calls. * Added ability for translation
+ on-screen menu to multiple languages. Tested on Russian
+ languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO
+ 8859-4, ISO 8859-5, ISO 2022-JP. Language controlled by
+ 'language' and on-screen menu of phone * Other described in
+ CHANGES file Testing done by issue tracker users: ibercom,
+ scsiborg, idarwin, TeknoJuce, c0rnoTa. Tested on production
+ system by Jonn Taylor (jonnt) using phone models: Nortel i2004,
+ 1120E and 1140E. (closes issue ASTERISK-16890) Review:
+ https://reviewboard.asterisk.org/r/1243/
+
+2012-03-10 20:06 +0000 [r358730] Joshua Colp <jcolp@digium.com>
+
+ * configs/confbridge.conf.sample, main/dial.c, apps/app_page.c,
+ apps/confbridge/include/confbridge.h, apps/app_confbridge.c,
+ include/asterisk/dial.h, CHANGES,
+ apps/confbridge/conf_config_parser.c: Transition app_page to
+ using app_confbridge internally for the conference bridge portion
+ of paging. This also adds a new 'announcement' option to
+ ConfBridge user profiles. Review:
+ https://reviewboard.asterisk.org/r/1754/
+
+2012-03-08 17:48 +0000 [r358646-358691] Sean Bright <sean@malleable.com>
+
+ * apps/app_dial.c, apps/app_directory.c, apps/app_queue.c: Resolve
+ a few more cases of variable shadowing.
+
+ * channels/chan_phone.c, channels/chan_skinny.c,
+ channels/chan_agent.c, pbx/pbx_lua.c, pbx/pbx_dundi.c,
+ channels/chan_gtalk.c, pbx/pbx_config.c, channels/chan_oss.c,
+ apps/confbridge/conf_config_parser.c: Eliminate a bunch of shadow
+ warnings.
+
+ * include/asterisk/linkedlists.h: Add some underscores in a few of
+ our llist macros to reduce name collisions.
+
+2012-03-08 16:59 +0000 [r358645] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: Make transfer not ignore port information
+ with SIP. Attempting to transfer with SIP to an address like
+ 1XXXXX@ip.ad.re.ss:5061 would fail because port would be cut from
+ the host string and ignored. This simply keeps chan_sip from
+ cutting off the port number during these kinds of transfers.
+ (closes issue ASTERISK-19321) Reported by: Federico Alves Review:
+ https://reviewboard.asterisk.org/r/1790/diff/#index_header
+ ........ Merged revisions 358643 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 358644 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-08 16:21 +0000 [r358609-358622] Sean Bright <sean@malleable.com>
+
+ * Makefile, configure, configure.ac, makeopts.in: Add
+ --enable-dev-mode=strict to configure. Passing -Wshadow to gcc
+ enables shadow warnings. From the gcc manual: Warn whenever a
+ local variable or type declaration shadows another variable,
+ parameter, type, or class member (in C++), or whenever a built-in
+ function is shadowed. Asterisk will not currently compile with
+ this option set, but a number of bugs have been discovered by
+ enabling this flag on specific files. The long-term goal is to
+ eliminate all of the suspect code that causes this warning to be
+ emitted.
+
+ * Makefile: Whitespace only change to the Makefile
+
+2012-03-07 21:28 +0000 [r358576] Terry Wilson <twilson@digium.com>
+
+ * cel/cel_odbc.c, configs/cel_odbc.conf.sample: Handle numeric
+ columns for eventtype properly in cel_odbc Patch also implements
+ correct handling of datetime2 and datetimeoffset new datatypes in
+ SQL Server 2008 and 2008 R2. (closes issue ASTERISK-17548)
+ Review: https://reviewboard.asterisk.org/r/1160/ Review:
+ https://reviewboard.asterisk.org/r/1804/
+
+2012-03-07 18:33 +0000 [r358532] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_ss7.c: Change directly setting _softhangup in
+ sig_ss7.c to use ast_softhangup_nolock(). Update to: (issue
+ ASTERISK-19372) ........ Merged revisions 358530 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 358531 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-07 16:16 +0000 [r358486] Sean Bright <sean@malleable.com>
+
+ * /, codecs/codec_dahdi.c: Return g729 and g723.1 frames with the
+ number of samples set properly. If the wctc4xxp returns more than
+ a single packet, we need to update the number of samples in the
+ returned frame accordingly. Acked-by: Shaun Ruffell
+ <sruffell@digium.com> ........ Merged revisions 358484 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 358485 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-07 15:19 +0000 [r358437-358444] Terry Wilson <twilson@digium.com>
+
+ * /, configs/cdr_adaptive_odbc.conf.sample: Set snarkiness = 0 in
+ cdr_adaptive_odbc.conf.sample ........ Merged revisions 358438
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 358441 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * cel/cel_odbc.c, /, cdr/cdr_adaptive_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. ........ Merged revisions
+ 358435 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 358436 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-06 17:47 +0000 [r358262-358379] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Fix ring cadance setup for outgoing
+ calls on FXS ports. * Fix referencing the wrong variable in
+ chan_dahdi.c:my_set_cadence(). Thanks to Sean Bright for
+ compiling with -Wshadow and finding this bug. ........ Merged
+ revisions 358377 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 358378 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, CHANGES:
+ Add dialtone_detect option for analog incoming calls. For analog
+ lines, enables Asterisk to use dialtone detection per channel if
+ an incoming call was hung up before it was answered. If dialtone
+ is detected, the call is hung up. no: Disabled. (Default) yes:
+ Look for dialtone for 10000 ms after answer. <number>: Look for
+ dialtone for the specified number of ms after answer. always:
+ Look for dialtone for the entire call. Dialtone may return if the
+ far end hangs up first. dialtone_detect=yes dialtone_detect=5000
+ dialtone_detect=always (closes issue ASTERISK-19316) Reported by:
+ Jeremy Pepper Patch by: Jeremy Pepper Tested by: rmudgett,Jeremy
+ Pepper Review: https://reviewboard.asterisk.org/r/1737/
+
+ * /, 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 ........ Merged revisions 358278 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 358284 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * include/asterisk/channel.h: Make usage of
+ DECLARE_STRINGFIELD_SETTERS_FOR() not look so odd.
+
+ * channels/chan_dahdi.c, channels/sig_ss7.h, /, channels/sig_ss7.c:
+ 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 ........
+ Merged revisions 358260 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 358261 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-05 19:06 +0000 [r358216] Jonathan Rose <jrose@digium.com>
+
+ * main/manager.c, /: Eliminate double close of file descriptor in
+ manager.c The process_output function in manager.c attempted to
+ call fclose and close immediately afterwards. Since fclose
+ implies close, this resulted in a potential double free on file
+ descriptors. This patch changes that behavior and also adds error
+ checking to fclose and close depending on which was deemed
+ necessary. Also error messages. Thanks to Rosen Iliev for
+ pointing out the location of the problem. (closes issue
+ ASTERISK-18453) Reported By: Jaco Kroon Review:
+ https://reviewboard.asterisk.org/r/1793/ ........ Merged
+ revisions 358214 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 358215 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-05 16:44 +0000 [r358164] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Defer sending the connected line reinvite
+ if a reinvite is already in progress. (issue ASTERISK-19355)
+ Reported by: tomaso (closes issue AST-825) ........ Merged
+ revisions 358162 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 358163 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-05 16:00 +0000 [r358117] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Ensure Asterisk acknowledges ACKs to 4xx
+ on Replaces errors Asterisk was not setting pendinginvite in the
+ upper half of handle_request_invite such that the 4xx was
+ retransmitted repeatedly even though an ack was received for
+ every retransmission. (closes issue ASTERISK-19303) Reported by:
+ Jon Tsiros Patches: fix-19303.patch uploaded by Jeremiah Gowdy
+ (license 6358) ........ Merged revisions 358115 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 358116 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-05 11:20 +0000 [r358082] Sean Bright <sean@malleable.com>
+
+ * configs/iax.conf.sample: Tab to spaces and text change.
+
+2012-03-02 23:29 +0000 [r357999-358038] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_usbradio.c, /, channels/xpmr/xpmr.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. ........ Merged revisions 358029 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 358033 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /: Correct some set-but-unused variable warnings in the mISDN
+ library. (from kpfleming's commit to trunk r356292) ........
+ Merged revisions 358011 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 358017 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, channels/xpmr/xpmr.c: Make chan_usbradio compile under dev
+ mode x=++x and x=x=1? Really? ........ Merged revisions 357986
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 357987 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-02 21:06 +0000 [r357942] Kinsey Moore <kmoore@digium.com>
+
+ * /, main/ccss.c, tests/test_event.c, main/event.c,
+ include/asterisk/strings.h: 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/
+ ........ Merged revisions 357940 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 357941 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-02 18:38 +0000 [r357896] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /, channels/sig_pri.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 ........ Merged revisions 357894 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 357895 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-02 16:57 +0000 [r357861] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_queue.c: Adds a transfer callee on hangup option (like
+ with Dial option F) to queues. This should (and does in my
+ testing) act just like the Dial option of the same name. This
+ allows a queue member to be transfered to the next priority (no
+ args), or to a context/extension/priority similar to goto (with
+ args context^extension^priority) when a caller hangs up on them.
+ (closes issue ASTERISK-19283) Reported by: To Patches:
+ queue_f-v3.diff uploaded by To (license 6347) Review:
+ https://reviewboard.asterisk.org/r/1785/
+
+2012-03-02 16:26 +0000 [r357834] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_chanspy.c: Remove bad usage of goto in ChanSpy
+ next_channel().
+
+2012-03-02 16:19 +0000 [r357821] Sean Bright <sean@malleable.com>
+
+ * configs/iax.conf.sample: Beef up the IAX2 sample configuration a
+ bit and fix some formatting issues.
+
+2012-03-02 16:03 +0000 [r357814-357815] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_chanspy.c: Fix channel reference leak in ChanSpy. *
+ Fix next_channel() channel reference leak in ChanSpy. (closes
+ issue ASTERISK-19461) Reported by: Irontec Patches:
+ app_chanspy_iteartor_next_unref.patch (license #6213) patch
+ uploaded by Irontec (issue ASTERISK-17515) ........ Merged
+ revisions 357809 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 357810 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/chan_usbradio.c: Fix compile error from latest channel
+ opaquification change.
+
+2012-03-02 16:00 +0000 [r357813] Sean Bright <sean@malleable.com>
+
+ * /, channels/chan_iax2.c: The default value for mohinterpret is
+ the empty string, so when resetting to default values don't
+ explicitly set the value to "default." ........ Merged revisions
+ 357811 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 357812 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-03-02 01:33 +0000 [r357774-357775] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Fix race condition that can cause important
+ control frames (such as a hangup) to be missed. This takes two
+ actions. 1. Move the reading of the alertpipe in __ast_read() to
+ immediately before the removal of frames from the readq. This
+ means we won't do something silly like read from the alertpipe,
+ then ignore the fact that there's a frame to get from the readq
+ since channel's fdno is the AST_TIMING_FD. 2. When
+ ast_settimeout() sets the rate to 0 and the timingfunc to NULL,
+ if the channel's fdno is the AST_TIMING_FD, then set the fdno to
+ -1. This is because if the rate is 0 and the timingfunc is NULL,
+ it means that the channel's timing fd is being invalidated, so
+ any pending reads should not occur. This may actually solve more
+ issues than the referenced one below, but it's not known at this
+ time for sure. (closes issue ASTERISK-19223) reported by
+ Frank-Michael Wittig Review:
+ https://reviewboard.asterisk.org/r/1779 ........ Merged revisions
+ 357761 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 357762 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/chan_dahdi.c: Fix compilation error due to typo during
+ channel opaquification.
+ s/ast_channel_fd_set/ast_channel_internal_fd_set/g
+
+2012-03-01 22:09 +0000 [r357721] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_unistim.c, apps/app_dahdibarge.c,
+ main/autoservice.c, addons/chan_ooh323.c, channels/chan_vpb.cc,
+ apps/app_meetme.c, channels/console_video.c,
+ channels/chan_gtalk.c, channels/chan_iax2.c, main/cli.c,
+ main/channel.c, channels/chan_phone.c, channels/chan_dahdi.c,
+ channels/sig_analog.c, channels/chan_skinny.c, main/features.c,
+ apps/app_dumpchan.c, channels/sig_ss7.c, channels/chan_mgcp.c,
+ main/pbx.c, channels/chan_sip.c, main/channel_internal_api.c,
+ channels/chan_agent.c, apps/app_dahdiras.c,
+ include/asterisk/channel.h, apps/app_queue.c, channels/sig_pri.c,
+ channels/chan_jingle.c, channels/chan_misdn.c, apps/app_flash.c,
+ funcs/func_channel.c, apps/app_directed_pickup.c, main/file.c,
+ channels/chan_h323.c, res/snmp/agent.c, main/dsp.c: Opaquify
+ ast_channel typedefs, fd arrays, and softhangup flag Review:
+ https://reviewboard.asterisk.org/r/1784/
+
+2012-03-01 14:22 +0000 [r357673] Kinsey Moore <kmoore@digium.com>
+
+ * /, main/acl.c: Prevent outbound SIP NOTIFY packets from
+ displaying a port of 0 In the change from 1.6.2 to 1.8,
+ ast_sockaddr was introduced which changed the behavior of
+ ast_find_ourip such that port number was wiped out. This caused
+ the port in internip (which is used for Contact and Call-ID on
+ NOTIFYs) to be 0. This change causes ast_find_ourip to be
+ port-preserving again. (closes issue ASTERISK-19430) ........
+ Merged revisions 357665 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 357667 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-29 20:41 +0000 [r357621] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, main/utils.c, include/asterisk/stringfields.h: Update
+ stringfield documentation for removed second va_list in favor of
+ va_copy. In r320946, the second va_list that was passed to
+ ast_string_field_build_va and friends, was removed. This patch
+ updates the documentation to reflect that. ........ Merged
+ revisions 357620 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-29 20:31 +0000 [r357610] Sean Bright <sean@malleable.com>
+
+ * res/res_agi.c, CHANGES: Add IPv6 support to FastAGI. Review:
+ https://reviewboard.asterisk.org/r/1774/ Reviewed by: Simon
+ Perreault, Mark Michelson
+
+2012-02-29 19:48 +0000 [r357577] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * apps/app_dial.c, /: Fix copying of CDR(accountcode) to local
+ channels. In r203638, during the addition of the Channel Event
+ Logging, in mid-2009, this got broken in trunk and ended up in
+ asterisk 1.8 and higher. This fixes so the CDR(accountcode) from
+ the calling channel is available to dialed channels again as well
+ as showing up properly in the CDR's. (closes issue
+ ASTERISK-19384) Reported by: jamicque Patches: accountcode.patch
+ (License #6033) by jamicque Review:
+ https://reviewboard.asterisk.org/r/1775/ Reviewed by: Richard
+ Mudgett ........ Merged revisions 357575 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 357576 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-29 16:52 +0000 [r357542] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c, addons/chan_ooh323.c,
+ funcs/func_strings.c, channels/console_video.c,
+ apps/app_alarmreceiver.c, channels/chan_iax2.c, main/cli.c,
+ channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/chan_skinny.c, apps/app_dumpchan.c, main/features.c,
+ apps/app_amd.c, channels/sig_ss7.c, apps/app_dial.c, main/pbx.c,
+ include/asterisk/utils.h, funcs/func_timeout.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, channels/chan_misdn.c,
+ apps/app_macro.c, apps/app_zapateller.c, apps/app_mixmonitor.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, main/autochan.c, apps/app_followme.c,
+ main/cdr.c, main/channel.c, main/dial.c, channels/chan_phone.c,
+ apps/app_osplookup.c, apps/app_setcallerid.c, main/manager.c,
+ bridges/bridge_builtin_features.c, apps/app_minivm.c,
+ res/res_agi.c, main/app.c, apps/app_confbridge.c, apps/app_rpt.c,
+ main/message.c, channels/chan_mgcp.c, apps/app_parkandannounce.c,
+ apps/app_while.c, funcs/func_dialplan.c, channels/chan_sip.c,
+ res/res_fax.c, main/channel_internal_api.c, pbx/pbx_lua.c,
+ channels/chan_console.c, channels/sig_pri.c, apps/app_queue.c,
+ channels/chan_oss.c, channels/chan_jingle.c,
+ channels/chan_usbradio.c, funcs/func_blacklist.c,
+ main/abstract_jb.c, channels/chan_h323.c, main/file.c,
+ res/snmp/agent.c, apps/app_sms.c, apps/app_stack.c,
+ funcs/func_callerid.c: Opaquify ast_channel structs and lists
+ Review: https://reviewboard.asterisk.org/r/1773/
+
+2012-02-28 22:31 +0000 [r357460-357503] Jonathan Rose <jrose@digium.com>
+
+ * /, configs/sip.conf.sample, UPGRADE-1.8.txt: 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) ........ Merged revisions 357490
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 357497 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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) ........ Merged
+ revisions 357455 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 357458 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-28 21:26 +0000 [r357436] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, apps/app_stack.c: Correctly reset the dialplan priority. When
+ the stack frame is allocated, we save the address to which we
+ should return, when the Gosub returns. However, if we just want
+ to restore the priority, then we need to subtract 1 before
+ setting it. Otherwise, when a Gosub goes to a nonexistent
+ address, it will skip a priority in the dialplan. This is because
+ when we return from an application, the PBX increments the
+ priority for us. ........ Merged revisions 357416 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 357421 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-28 21:01 +0000 [r357409] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Use more reasonable cause code when
+ rejecting incoming call waiting calls. (closes issue
+ ASTERISK-19397) Reported by: Birger Harzenetter Patches:
+ nochannel-cause.patch (license #5870) patch uploaded by Birger
+ Harzenetter ........ Merged revisions 357407 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 357408 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-28 20:43 +0000 [r357406] Jonathan Rose <jrose@digium.com>
+
+ * /, UPGRADE-10.txt: revision 357386 -- oops, accidentally made it
+ 10.3 to 10.4 instead of 10.2 to 10.3 (issue ASTERISK-19352)
+ reported by: jamicque ........ Merged revisions 357405 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-28 20:34 +0000 [r357404] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, res/res_musiconhold.c, apps/app_queue.c: Fix
+ REF_DEBUG compile errors.
+
+2012-02-28 20:33 +0000 [r357358-357403] Jonathan Rose <jrose@digium.com>
+
+ * /, UPGRADE-10.txt, UPGRADE-1.8.txt: Moves UPGRADE.txt notes from
+ r357356 to a new section specific to 1.8.12 (issue
+ ASTERISK-19352) reported by: jamicque ........ Merged revisions
+ 357386 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 357400 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, UPGRADE-1.8.txt: Adds UPGRADE.txt notes to r357266 indicating
+ changes to transport option (issue ASTERISK-19352) Reported by:
+ jamicque ........ Merged revisions 357356 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 357357 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-28 19:55 +0000 [r357355] Sean Bright <sean@malleable.com>
+
+ * include/asterisk/netsock2.h: Documentation update. There is no
+ AST_SOCKADDR_UNSPEC.
+
+2012-02-28 19:37 +0000 [r357354] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_page.c: Remove dupliate 'i' option table entry in
+ app_page.c. (closes issue ASTERISK-19310) Reported by: Makoto Dei
+ Patches: app_page-duplicate-i-option.patch (license #5027) patch
+ uploaded by Makoto Dei ........ Merged revisions 357352 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 357353 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-28 18:52 +0000 [r357319] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/sip/security_events.c: Add a security event for the
+ case where fake authentication challenge is sent. ........ Merged
+ revisions 357318 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-28 18:46 +0000 [r357317] Richard Mudgett <rmudgett@digium.com>
+
+ * main/tcptls.c, channels/chan_sip.c, include/asterisk/tcptls.h:
+ Convert struct ast_tcptls_session_instance to finally use the ao2
+ object lock.
+
+2012-02-28 18:23 +0000 [r357288] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: Changes transport option in sip.conf so
+ that using multiple instances doesn't stack. Prior to this patch,
+ Using "transport=" multiple times would cause them to add to one
+ another like allow/deny. This patch changes that behavior to
+ simply use the transport option specified last. Also, if no
+ transport option is applied now, the default will automatically
+ be UDP. (closes ASTERISK-19352) Reported by: jamicque Patches:
+ asterisk-19352-transport-warning-message-v1.patch uploaded by
+ Michael L. Young (license 5026)
+ issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes
+ (license 5674) Review:
+ https://reviewboard.asterisk.org/r/1745/diff/#index_header
+ ........ Merged revisions 357266 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 357271 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-28 18:15 +0000 [r357272] Richard Mudgett <rmudgett@digium.com>
+
+ * main/format.c, main/format_cap.c, include/asterisk/astobj2.h,
+ include/asterisk/lock.h, main/astobj2.c: Astobj2 locking
+ enhancement. Add the ability to specify what kind of locking an
+ ao2 object has when it is allocated. The locking could be one of:
+ MUTEX, RWLOCK, or none. New API: ao2_t_alloc_options()
+ ao2_alloc_options() ao2_t_container_alloc_options()
+ ao2_container_alloc_options() ao2_rdlock() ao2_wrlock()
+ ao2_tryrdlock() ao2_trywrlock() The OBJ_NOLOCK and
+ AO2_ITERATOR_DONTLOCK flags have a slight meaning change. They no
+ longer mean that the object is protected by an external
+ mechanism. They mean the lock associated with the object has
+ already been manually obtained by one of the ao2_lock calls. This
+ change is necessary for RWLOCK support since they are not
+ reentrant. Also an operation on an ao2 container may require
+ promoting a read lock to a write lock by releasing the already
+ held read lock to re-acquire as a write lock. Replaced API calls:
+ ao2_t_link_nolock() ao2_link_nolock() ao2_t_unlink_nolock()
+ ao2_unlink_nolock() with the respective ao2_t_link_flags()
+ ao2_link_flags() ao2_t_unlink_flags() ao2_unlink_flags() API
+ calls to be more flexible and to allow an anticipated enhancement
+ to control linking duplicate objects into a container. The
+ changes to format.c and format_cap.c are taking advantange of the
+ new ao2 locking options to simplify the use of the format
+ capabilities containers. Review:
+ https://reviewboard.asterisk.org/r/1554/
+
+2012-02-28 14:47 +0000 [r357178-357214] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, Makefile.rules: Make COMPILE_DOUBLE magic actually work. The
+ build system has some special magic to ensure that if Asterisk is
+ built with --enable-dev-mode *and* DONT_OPTIMIZE, that all the
+ source is still compiled with the optimizer enabled (even though
+ the result will be thrown away), because the compiler is able to
+ find a great deal of coding errors and bugs as a result of
+ running its optimizers. Unfortunately at some point this mode got
+ broken, and the 'throwaway' compile of the code was no longer
+ done with the optimizer enabled. This patch corrects that
+ problem. ........ Merged revisions 357212 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 357213 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/astobj2.c: Trailing whitespace cleanup.
+
+2012-02-28 00:42 +0000 [r357096-357145] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/astobj2.h, tests/test_astobj2.c, main/astobj2.c:
+ Add ability to clone ao2 containers. Occasionally there is a need
+ to put all objects in one container also into another container.
+ Some reasons you might need to do this: 1) You need to
+ reconfigure a container. You would do this by creating a new
+ container with the new configuration and ao2_container_dup the
+ old container into it. Then replace the old container with the
+ new. Then destroy the old container. 2) You need the contents of
+ a container to remain stable while operating on all of the
+ objects. You would do this by creating a cloned container of the
+ original with ao2_container_clone. The cloned container is a
+ snapshot of the objects at the time of the cloning. When done,
+ just destroy the cloned container. Review:
+ https://reviewboard.asterisk.org/r/1746/
+
+ * main/channel.c: Fix ast_channel allocation init setting priority
+ to -1 instead of 1. * Fix opaquification conversion error.
+ (closes issue ASTERISK-19424) Reported by: Jeremy Pepper Patches:
+ asterisk-19424-initialize_priority_regression.diff (license
+ #5026) patch uploaded by Michael L. Young
+
+ * main/channel.c, /: Fix callerid of Originated calls. Thanks to
+ Matt Riddell for tracking this down. (closes issue
+ ASTERISK-19385) Reported by: ornix ........ Merged revisions
+ 357093 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 357095 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-27 19:55 +0000 [r357051] Jonathan Rose <jrose@digium.com>
+
+ * include/asterisk/res_odbc.h, res/res_odbc.c: Converts locking for
+ odbc containers from ast_mutex_lock to ao2_locks.
+
+2012-02-27 17:03 +0000 [r357014] Sean Bright <sean@malleable.com>
+
+ * channels/chan_iax2.c, main/netsock.c: Address comments from Mark
+ Michelson
+
+2012-02-27 16:50 +0000 [r357013] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_dial.c, main/channel.c, include/asterisk/app.h,
+ main/dial.c, main/rtp_engine.c, main/ccss.c, main/features.c,
+ UPGRADE.txt, main/app.c, include/asterisk/channel.h,
+ configs/ccss.conf.sample, apps/app_followme.c, apps/app_queue.c,
+ include/asterisk/ccss.h: Deprecated macro usage for connected
+ line, redirecting, and CCSS This commit adds GoSub alternatives
+ to connected line, redirecting, and CCSS macro hooks so that
+ macro can finally be deprecated. This also adds deprecation
+ warnings for those features when used and in documentation.
+ Review: https://reviewboard.asterisk.org/r/1760/ (closes issue
+ SWP-4256)
+
+2012-02-27 16:31 +0000 [r357005] Sean Bright <sean@malleable.com>
+
+ * include/asterisk/netsock.h, channels/chan_iax2.c, main/netsock.c:
+ Convert netsock.h over to use ast_sockaddrs rather than
+ sockaddr_in and update chan_iax2 to pass in the correct types.
+ chan_iax2 is the only consumer for the various ast_netsock_*
+ functions in trunk at this point, so this feels like a safe
+ change to make.
+
+2012-02-27 16:24 +0000 [r356987] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
+ channels/sip/include/sip.h: Adds an option to sip.conf that
+ prevents diversion headers from being added. send_diversion=no
+ will prevent Diversion headers from being added to SIP requests.
+ This doesn't prevent Diversion from being added with dialplan
+ such as with SIPAddHeader. (closes issue ASTERISK-16862) Reported
+ by: rsw686 Review: https://reviewboard.asterisk.org/r/1769/
+
+2012-02-27 16:12 +0000 [r356966] Sean Bright <sean@malleable.com>
+
+ * channels/chan_iax2.c: There isn't much point in saving off and
+ restoring a value that we never use again.
+
+2012-02-27 16:08 +0000 [r356965] Terry Wilson <twilson@digium.com>
+
+ * /, main/features.c: Copy CDR variables when set during a bridge
+ This patch makes sure amaflags, accountcode, and userfield get
+ copied to the bridge CDR when set during a bridge (like via a
+ custom feature). (closes issue ASTERISK-16990) Review:
+ https://reviewboard.asterisk.org/r/1721/ ........ Merged
+ revisions 356963 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 356964 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-27 15:35 +0000 [r356962] Jonathan Rose <jrose@digium.com>
+
+ * /, res/res_odbc.c: Remove possible segfaults from res_odbc by
+ adding locks around usage of odbc handle (closes issue
+ ASTERISK-19011) Reported by: Walter Doekes Patches:
+ issueA19011_combine_read_and_write_locks_WORK_IN_PROGRESS.patch
+ uploaded by Walter Doekes (license 5674) review:
+ https://reviewboard.asterisk.org/r/1719/ review:
+ https://reviewboard.asterisk.org/r/1622/ ........ Merged
+ revisions 356917 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 356961 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-27 14:57 +0000 [r356881-356916] Sean Bright <sean@malleable.com>
+
+ * include/asterisk/netsock.h, main/netsock.c: Make
+ ast_netsock_set_qos() delegate to ast_set_qos().
+
+ * include/asterisk/netsock.h: Correct typo in deprecation comment.
+
+ * channels/chan_unistim.c, main/udptl.c, channels/chan_skinny.c,
+ include/asterisk/netsock.h, pbx/pbx_dundi.c,
+ channels/chan_mgcp.c: Prefer ast_set_qos() over
+ ast_netsock_set_qos()
+
+ * main/netsock.c: Remove trailing whitespace
+
+2012-02-26 18:25 +0000 [r356848] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooGkClient.c, addons/chan_ooh323.c: Add
+ support change gatekeeper mode or ip per ooh323 reload command
+ (issue ASTERISK-19298) Reported by: Dmitry Melekhov Patches:
+ change_gk_on_reload-1.patch (License #5415)
+
+2012-02-25 17:22 +0000 [r356799] Matthew Jordan <mjordan@digium.com>
+
+ * /, apps/app_voicemail.c: Fix crash in app_voicemail during
+ close_mailbox In r354890, a memory leak in app_voicemail was
+ fixed by properly disposing of the allocated heard/deleted
+ pointers. However, there are situations, particularly when no
+ messages are found in a folder, where these pointers are not
+ allocated and not NULL. In that case, an invalid free would be
+ attempted, which could crash app_voicemail. As there are a number
+ of code paths where this could occur, this patch uses the number
+ of messages detected in the folder before it attempts to free the
+ pointers. This resolves the crash detected in the Asterisk Test
+ Suite's check_voicemail_nominal test. ........ Merged revisions
+ 356797 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 356798 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-24 23:40 +0000 [r356697-356765] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/astobj2.h: astobj2.h comment tweaks.
+
+ * include/asterisk/astobj2.h, main/astobj2.c: astobj2.h
+ documentation updates.
+
+ * /, channels/chan_sip.c, include/asterisk/tcptls.h,
+ channels/sip/include/sip.h: 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/ ........ Merged
+ revisions 356677 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 356690 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-24 17:43 +0000 [r356606-356652] Matthew Jordan <mjordan@digium.com>
+
+ * /, 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. ........ Merged revisions 356650
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 356651 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/sip/sdp_crypto.c, include/asterisk/res_srtp.h,
+ main/rtp_engine.c, /, include/asterisk/rtp_engine.h,
+ res/res_srtp.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) ........ Merged revisions 356604 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 356605 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-24 00:32 +0000 [r356573] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_unistim.c, channels/chan_local.c,
+ addons/chan_ooh323.c, channels/chan_multicast_rtp.c,
+ channels/chan_vpb.cc, main/rtp_engine.c, apps/app_meetme.c,
+ apps/app_dictate.c, apps/app_record.c, apps/app_test.c,
+ bridges/bridge_softmix.c, channels/chan_gtalk.c, apps/app_ices.c,
+ res/res_musiconhold.c, channels/chan_iax2.c,
+ bridges/bridge_multiplexed.c, main/indications.c, main/cli.c,
+ main/channel.c, channels/chan_phone.c, channels/chan_dahdi.c,
+ channels/chan_skinny.c, res/res_agi.c, main/features.c,
+ apps/app_mp3.c, apps/app_dumpchan.c, main/app.c, apps/app_amd.c,
+ channels/chan_alsa.c, apps/app_confbridge.c,
+ addons/chan_mobile.c, main/bridging.c, channels/chan_mgcp.c,
+ apps/app_nbscat.c, main/pbx.c, channels/chan_sip.c,
+ res/res_fax.c, apps/app_festival.c, channels/chan_bridge.c,
+ main/channel_internal_api.c, apps/app_fax.c,
+ apps/app_waitforsilence.c, res/res_adsi.c, channels/chan_agent.c,
+ bridges/bridge_simple.c, include/asterisk/channel.h,
+ channels/chan_console.c, apps/app_talkdetect.c,
+ channels/chan_oss.c, apps/app_speech_utils.c,
+ channels/chan_usbradio.c, channels/chan_jingle.c,
+ channels/chan_misdn.c, funcs/func_channel.c, main/file.c,
+ channels/chan_nbs.c, apps/app_chanspy.c, apps/app_voicemail.c,
+ res/res_calendar.c: Opaquification for ast_format structs in
+ struct ast_channel Review:
+ https://reviewboard.asterisk.org/r/1770/
+
+2012-02-23 20:14 +0000 [r356523] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c, main/features.c: Fix blind transfer
+ parking issues if the dialed extension is not recognized as a
+ parking extension. Custom parking extensions may not be coded
+ such that the first and only extension priority is the Park
+ application. These custom parking extensions will not be
+ recognized as parking extensions. When a call is blind
+ transferred to an extension that is not recognized as a parking
+ extension, the normal blind transfer code causes the transferred
+ channel to start executing dialplan. Calls that get parked in
+ this manner do not know the original channel name that parked the
+ call so the original parker could never be called back if the
+ parked call is not retrieved before the timeout time. The parking
+ space is also announced to the call being parked as a side effect
+ of not knowing the original parking channel. * Fix handling of
+ BLINDTRANSFER channel variable for call parking. * Fixed SIP
+ blind transfer using the wrong dialplan context variable to check
+ for the parking extension. (closes issue ASTERISK-19322) Reported
+ by: aragon Tested by: rmudgett, jparker Review:
+ https://reviewboard.asterisk.org/r/1730/ JIRA AST-766 ........
+ Merged revisions 356521 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 356522 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-23 15:49 +0000 [r356477] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Fix ACK routing for non-2xx responses.
+ When we send an ACK for a 2xx response to an INVITE, we are
+ supposed to use the learned route set. However, when we receive a
+ non-2xx final response to an INVITE, we are supposed to send the
+ ACK to the same place we initially sent the INVITE. We had been
+ doing this up until the changes went in that would build a route
+ set from provisional responses. That introduced a regression
+ where we would use the learned route set under all circumstances.
+ With this change, we now will set the destination of our ACK
+ based on the invitestate. If it is INV_COMPLETED then that means
+ that we have received a non-2xx final response (INV_TERMINATED
+ indicates a 2xx response was received). If it is INV_CANCELLED,
+ then that means the call is being canceled, which means that we
+ should be ACKing a 487 response. The other change introduced here
+ is setting the invitestate to INV_CONFIRMED when we send an ACK
+ *after* the reqprep instead of before. This way, we can tell in
+ reqprep more easily what the invitestate is prior to sending the
+ ACK. (closes issue ASTERISK-19389) reported by Karsten Wemheuer
+ patches: ASTERISK-19389v2.patch uploaded by Mark Michelson
+ (license #5049) (with some slight modifications prior to commit)
+ ........ Merged revisions 356475 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 356476 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-23 03:27 +0000 [r356429] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, apps/app_rpt.c: Multiple revisions 356290,356335,356337
+ ........ r356290 | pabelanger | 2012-02-22 15:20:29 -0500 (Wed,
+ 22 Feb 2012) | 4 lines Fix -Werror=unused-but-set-variable
+ compiler error (gcc 4.6.2) Review:
+ https://reviewboard.asterisk.org/r/1763/ ........ r356335 |
+ pabelanger | 2012-02-22 16:29:25 -0500 (Wed, 22 Feb 2012) | 2
+ lines Add back strsep() function for previous commit ........
+ r356337 | pabelanger | 2012-02-22 16:36:37 -0500 (Wed, 22 Feb
+ 2012) | 2 lines Missed one strsep() function ........ Merged
+ revisions 356290,356335,356337 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 356428 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-23 01:53 +0000 [r356397] Terry Wilson <twilson@digium.com>
+
+ * tests/test_substitution.c, tests/test_utils.c: Fix some tests
+ that didn't get opaquification changes Review:
+ https://reviewboard.asterisk.org/r/1766/
+
+2012-02-23 00:56 +0000 [r356366] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel_internal_api.c: Revert some apparently accidental
+ spacing changes.
+
+2012-02-22 21:22 +0000 [r356314] Terry Wilson <twilson@digium.com>
+
+ * /, include/asterisk/calendar.h, main/loader.c,
+ res/res_calendar.c: Track module use count for res_calendar If
+ the res_calendar module was followed immediately by one of the
+ calendar tech modules and "core stop gracefully" was run,
+ Asterisk would crash. This patch adds use count tracking for
+ res_calendar so that it is unloaded after the tech modules when
+ shutting down gracefully. It is now not possible to unload all
+ the of the calendar modules via "module unload res_calednar.so",
+ but it is still possible to unload them all via "module unload -h
+ res_calendar.so". Review:
+ https://reviewboard.asterisk.org/r/1752/ ........ Merged
+ revisions 356291 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 356297 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-22 21:10 +0000 [r356292] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c:
+ Correct some set-but-unused variable warnings in the mISDN
+ library.
+
+2012-02-22 17:34 +0000 [r356259] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_misdn.c: Fix chan_misdn after the lastest
+ opaquification changes It now compiles, but there are some
+ unrelated warnings for set but unused variables.
+
+2012-02-22 14:54 +0000 [r356216] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 356215 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r356215 | mjordan | 2012-02-22 08:53:53 -0600
+ (Wed, 22 Feb 2012) | 32 lines Merged revisions 356214 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r356214 | mjordan | 2012-02-22 08:50:20 -0600 (Wed, 22 Feb 2012)
+ | 27 lines 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 00:35 +0000 [r356152-356183] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, main/channel_internal_api.c,
+ include/asterisk/channel.h: Rename
+ ast_channel_emulate_dtmf_digit* funcs The accessors names for the
+ "emulate_dtmf_digit" field on the ast_channel are misleading.
+ Change them to ast_channel_dtmf_digit_to_emulate*.
+
+ * main/channel.c, main/framehook.c, res/res_monitor.c: Fix some
+ opaquification-related compiler warnings (closes issue
+ ASTERISK-19419) PseudoReview - seanbright on IRC
+
+2012-02-21 11:17 +0000 [r356111] Sean Bright <sean@malleable.com>
+
+ * /, channels/chan_iax2.c: Make 'iax2 show callnumber usage' output
+ make sense when an IP is passed in. ........ Merged revisions
+ 356107 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 356108 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-21 04:31 +0000 [r356075] Kinsey Moore <kmoore@digium.com>
+
+ * /, main/ccss.c: Add missing newline to ccss state change
+ notification Move along, nothing to see here... ........ Merged
+ revisions 356074 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-20 23:43 +0000 [r356042] Terry Wilson <twilson@digium.com>
+
+ * main/udptl.c, apps/app_dahdibarge.c, addons/chan_ooh323.c,
+ cdr/cdr_sqlite3_custom.c, channels/chan_local.c,
+ main/rtp_engine.c, apps/app_playtones.c, apps/app_record.c,
+ apps/app_sayunixtime.c, apps/app_test.c, main/devicestate.c,
+ apps/app_alarmreceiver.c, apps/app_chanisavail.c,
+ apps/app_ices.c, channels/chan_iax2.c,
+ bridges/bridge_multiplexed.c, main/cli.c, channels/chan_dahdi.c,
+ channels/sig_analog.c, main/framehook.c, channels/chan_skinny.c,
+ main/features.c, apps/app_dumpchan.c, pbx/pbx_realtime.c,
+ channels/chan_alsa.c, apps/app_externalivr.c, main/bridging.c,
+ channels/sig_ss7.c, apps/app_milliwatt.c, cdr/cdr_manager.c,
+ apps/app_dial.c, main/pbx.c, funcs/func_timeout.c,
+ apps/app_privacy.c, channels/chan_bridge.c, apps/app_echo.c,
+ apps/app_softhangup.c, apps/app_fax.c, apps/app_dahdiras.c,
+ channels/chan_agent.c, apps/app_disa.c, bridges/bridge_simple.c,
+ include/asterisk/channel.h, apps/app_talkdetect.c,
+ apps/app_transfer.c, main/cel.c, res/res_monitor.c,
+ apps/app_playback.c, apps/app_speech_utils.c,
+ channels/chan_misdn.c, apps/app_sendtext.c, funcs/func_channel.c,
+ funcs/func_cdr.c, channels/sip/dialplan_functions.c,
+ apps/app_macro.c, apps/app_zapateller.c, main/audiohook.c,
+ apps/app_chanspy.c, apps/app_voicemail.c, apps/app_cdr.c,
+ res/res_calendar.c, channels/chan_unistim.c,
+ channels/chan_multicast_rtp.c, channels/chan_vpb.cc,
+ apps/app_meetme.c, main/ccss.c, apps/app_dictate.c,
+ apps/app_authenticate.c, apps/app_readexten.c,
+ channels/chan_gtalk.c, res/res_musiconhold.c,
+ apps/app_followme.c, main/channel.c, main/cdr.c,
+ channels/chan_phone.c, main/dial.c, main/manager.c,
+ apps/app_osplookup.c, bridges/bridge_builtin_features.c,
+ res/res_agi.c, apps/app_minivm.c, main/app.c,
+ apps/app_confbridge.c, main/image.c, apps/app_directory.c,
+ main/message.c, apps/app_ivrdemo.c, addons/chan_mobile.c,
+ apps/app_rpt.c, cdr/cdr_custom.c, apps/app_parkandannounce.c,
+ channels/chan_mgcp.c, apps/app_while.c, res/res_rtp_asterisk.c,
+ apps/app_read.c, channels/chan_sip.c, apps/app_festival.c,
+ res/res_fax.c, cdr/cdr_syslog.c, apps/app_waitforsilence.c,
+ main/channel_internal_api.c, res/res_adsi.c, pbx/pbx_lua.c,
+ funcs/func_jitterbuffer.c, channels/chan_console.c,
+ apps/app_queue.c, channels/sig_pri.c, channels/chan_oss.c,
+ channels/chan_jingle.c, channels/chan_usbradio.c,
+ apps/app_channelredirect.c, apps/app_forkcdr.c, apps/app_flash.c,
+ main/abstract_jb.c, main/file.c, channels/chan_h323.c,
+ include/asterisk/sched.h, res/snmp/agent.c, apps/app_sms.c,
+ channels/chan_nbs.c, funcs/func_callerid.c, apps/app_verbose.c,
+ apps/app_stack.c: ast_channel opaquification of pointers and
+ integral types Review: https://reviewboard.asterisk.org/r/1753/
+
+2012-02-20 18:40 +0000 [r355903-355999] Sean Bright <sean@malleable.com>
+
+ * /, 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) ........ Merged revisions 355997 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 355998 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/chan_dahdi.c, /: This was a LOG_NOTICE, so roll it back.
+ ........ Merged revisions 355952 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 355953 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/chan_dahdi.c, /: Change some debug messages from
+ LOG_DEBUG to ast_debug. ........ Merged revisions 355949 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 355950 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, channels/chan_iax2.c: Add some boilerplate documentation for
+ IAXVAR and IAXPEER. ........ Merged revisions 355904 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 355905 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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. ........
+ Merged revisions 355901 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 355902 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-18 08:02 +0000 [r355852] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_ss7.h, /, channels/sig_analog.h, channels/sig_pri.c,
+ channels/sig_ss7.c: 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/ ........ Merged
+ revisions 355850 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 355851 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-17 22:03 +0000 [r355795] Sean Bright <sean@malleable.com>
+
+ * configs/iax.conf.sample, /, channels/chan_iax2.c: Don't allow
+ trunkfreq to be greater than 1000ms. ........ Merged revisions
+ 355793 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 355794 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-17 19:56 +0000 [r355749] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/asterisk.c: Non-verbose output should always go to the
+ remote console, regardless of the previous level.
+
+2012-02-17 19:35 +0000 [r355748] Sean Bright <sean@malleable.com>
+
+ * /, channels/chan_iax2.c: Pass the correct value to
+ ast_timer_set_rate() for IAX2 trunking. IAX2 uses the trunkfreq
+ variable to determine how often to send trunk packets, but this
+ value is in milliseconds while ast_timer_set_rate() expects the
+ rate argument to be ticks per second. So we divide 1000 by
+ trunkfreq and pass that in instead. With a default of 20ms, this
+ change makes IAX2 send trunk packets every 20ms instead of every
+ 50ms. Tracked down by myself and Bob Wienholt. ........ Merged
+ revisions 355746 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 355747 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-17 19:22 +0000 [r355745] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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
+ ........ Merged revisions 355732 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 355733 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-16 22:00 +0000 [r355667] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * apps/app_rpt.c: Fix channel opaquification for app_rpt
+
+2012-02-16 20:03 +0000 [r355624] Sean Bright <sean@malleable.com>
+
+ * /, main/audiohook.c: Revert a change to
+ audio_audiohook_write_list that had no affect. When I made this
+ change initially, I was under the false impression that the
+ audiohooks structure remained on the channel after all of the
+ hooks had been detached. This is not the case, ast ast_read takes
+ care of removing the audiohooks structure if the lists are empty.
+ ........ Merged revisions 355622 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 355623 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-16 19:51 +0000 [r355576-355621] Richard Mudgett <rmudgett@digium.com>
+
+ * /, 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 ........
+ Merged revisions 355608 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 355620 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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. ........ Merged revisions 355574 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 355575 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-15 19:29 +0000 [r355450-355531] Sean Bright <sean@malleable.com>
+
+ * /, channels/chan_iax2.c: When IAX2 debugging is enabled, make
+ sure to log 'apathetic' messages too. ........ Merged revisions
+ 355529 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 355530 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * build_tools/cflags.xml, channels/chan_iax2.c: Remove IAX_OLD_FIND
+ from chan_iax2.
+
+ * /, 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. ........ Merged revisions 355448 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 355449 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-14 20:27 +0000 [r355413] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * utils/refcounter.c, main/pbx.c, funcs/func_timeout.c,
+ include/asterisk/autoconfig.h.in, utils/hashtest.c, UPGRADE.txt,
+ CHANGES, main/config.c, configs/logger.conf.sample,
+ main/loader.c, include/asterisk/logger.h, main/manager.c,
+ main/logger.c, utils/ael_main.c, utils/hashtest2.c,
+ codecs/codec_dahdi.c, main/stdtime/localtime.c, main/asterisk.c,
+ addons/res_config_mysql.c: Re-commit the verbose branch. This
+ change permits each verbose destination (consoles, logger) to
+ have its own concept of what the verbosity level is. The big
+ feature here is that the logger will now be able to capture a
+ particular verbosity level without condemning each console to
+ need to suffer that level of verbosity. Additionally, a stray
+ 'core set verbose' will no longer change what will go to the log.
+ Review: https://reviewboard.asterisk.org/r/1599/
+
+2012-02-14 19:29 +0000 [r355321-355376] Richard Mudgett <rmudgett@digium.com>
+
+ * /, 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 ........ Merged revisions 355365 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 355375 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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/ ........ Merged
+ revisions 355319 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 355320 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-14 16:28 +0000 [r355274] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Properly invert the return of a strncmp
+ call. This was causing identification that should have been made
+ private to be public. (closes issue AST-814) reported by Patrick
+ Anderson Patches: chan_sip.c.diff uploaded by Patrick Anderson
+ (license 5430) ........ Merged revisions 355268 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 355271 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-14 15:58 +0000 [r355230] Jason Parker <jparker@digium.com>
+
+ * /, configs/cdr_sqlite3_custom.conf.sample: Don't enable sqlite3
+ CDRs by default in sample configs. ........ Merged revisions
+ 355228 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 355229 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-14 13:35 +0000 [r355184] Sean Bright <sean@malleable.com>
+
+ * /, channels/chan_iax2.c: Clear the high order bit from the
+ destination call number before sending. send_apathetic_reply
+ takes the incoming frame's source call number as the destination
+ call number for the outgoing frame. If the incoming frame was a
+ full frame, then the high order bit of the source call number is
+ set and will be interpreted as a retransmit when sent back out as
+ the destination call number. ........ Merged revisions 355182
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 355183 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-14 09:58 +0000 [r355138] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c, /: call manager_event only if there is not
+ null channel structure (Closes issue ASTERISK-19298) Reported by:
+ robinfood Patches: issue19298.patch uploaded by may213 (License
+ #5415) ........ Merged revisions 355136 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 355137 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-14 00:43 +0000 [r355102] Russell Bryant <russell@russellbryant.com>
+
+ * res/res_agi.c, CHANGES: res_agi: Add AGIEXITONHANGUP variable.
+ This patch adds a variable AGIEXITONHANGUP for res_agi. If this
+ variable is set to "yes" on a channel, AGI() will exit
+ immediately once a channel hangup has been detected. This was the
+ behavior of AGI() in Asterisk 1.4 and earlier and is still
+ desired by some people. Review:
+ https://reviewboard.asterisk.org/r/1734/
+
+2012-02-13 22:04 +0000 [r355055-355058] Richard Mudgett <rmudgett@digium.com>
+
+ * pbx/pbx_spool.c, /: Fix occasional incorrectly delayed call-file
+ execution. Since the dir timestamp is available at one second
+ resolution, we cannot know if it was updated within the same
+ second after we scanned it. Therefore, we will force another scan
+ if the dir was just modified. * Changed to force another scan if
+ the directory was just modified. (closes issue ASTERISK-19081)
+ Reported by: Knut Bakke Review:
+ https://reviewboard.asterisk.org/r/1688/ ........ Merged
+ revisions 355056 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 355057 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/chan_misdn.c: Fix compile error from most recent
+ ast_channel opaquification installment.
+
+2012-02-13 19:56 +0000 [r355011] Joshua Colp <jcolp@digium.com>
+
+ * /, pbx/pbx_config.c: Only allow one 'dialplan reload' to execute
+ at a time as otherwise they would share the same common local
+ context list. (closes issue AST-758) ........ Merged revisions
+ 355009 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 355010 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-13 17:27 +0000 [r354968] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c, addons/chan_ooh323.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, pbx/pbx_realtime.c, channels/chan_alsa.c,
+ apps/app_dial.c, main/pbx.c, apps/app_fax.c,
+ channels/chan_agent.c, include/asterisk/channel.h,
+ apps/app_talkdetect.c, main/cel.c, channels/chan_misdn.c,
+ funcs/func_channel.c, apps/app_macro.c, apps/app_chanspy.c,
+ res/res_calendar.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, main/cdr.c,
+ main/channel.c, main/dial.c, channels/chan_phone.c,
+ main/manager.c, apps/app_osplookup.c,
+ bridges/bridge_builtin_features.c, res/res_agi.c,
+ apps/app_minivm.c, apps/app_confbridge.c, apps/app_directory.c,
+ addons/chan_mobile.c, apps/app_rpt.c, apps/app_parkandannounce.c,
+ channels/chan_mgcp.c, apps/app_while.c, funcs/func_dialplan.c,
+ channels/chan_sip.c, res/res_fax.c, main/channel_internal_api.c,
+ pbx/pbx_lua.c, channels/sig_pri.c, apps/app_queue.c,
+ channels/chan_oss.c, channels/chan_jingle.c,
+ apps/app_directed_pickup.c, main/file.c, channels/chan_h323.c,
+ res/snmp/agent.c, pbx/pbx_dundi.c, channels/chan_nbs.c,
+ apps/app_stack.c, apps/app_verbose.c: Opaquify char * and char[]
+ in ast_channel Review: https://reviewboard.asterisk.org/r/1733/
+
+2012-02-13 17:25 +0000 [r354964] Richard Mudgett <rmudgett@digium.com>
+
+ * res/res_config_pgsql.c, /, configs/extconfig.conf.sample: Fix
+ reconnecting to pgsql database after connection loss. There can
+ only be one database connection in res_config_pgsql just like
+ res_config_sqlite. If the connection is lost, the connection may
+ not get reestablished to the same database if the res_pgsql.conf
+ and extconfig.conf files are inconsistent. * Made only use the
+ configured database from res_pgsql.conf. * Fixed potential buffer
+ overwrite of last[] in config_pgsql(). (closes issue
+ ASTERISK-16982) Reported by: german aracil boned Review:
+ https://reviewboard.asterisk.org/r/1731/ ........ Merged
+ revisions 354953 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 354959 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-13 16:42 +0000 [r354939] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_confbridge.c: Don't try to play sound files that do
+ not exist. (closes issue ASTERISK-19188) Reported by: slesru
+ ........ Merged revisions 354938 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-10 22:44 +0000 [r354903] Jason Parker <jparker@digium.com>
+
+ * /, apps/app_voicemail.c: Fix a voicemail memory leak with
+ heard/deleted messages. open_mailbox() was changed quite a long
+ time ago to allocate this memory. close_mailbox() should have
+ been changed to be responsible for freeing it. ........ Merged
+ revisions 354889 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 354890 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-10 18:08 +0000 [r354837] Richard Mudgett <rmudgett@digium.com>
+
+ * main/manager.c, /: Fix AMI Redirect ExtraChannel not redirecting
+ to the same exten and context. The astman_get_header() never
+ returns NULL so the check by the code for NULL would never fail.
+ (closes issue ASTERISK-16974) Reported by: Nuno Borges Patches:
+ 0018325.patch (license #6116) patch uploaded by Nuno Borges
+ (modified) ........ Merged revisions 354835 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 354836 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-10 14:51 +0000 [r354799] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_voicemail.c: Fix IMAP app_voicemail compilation issue
+ introduced in r354429 This simply fixes the compilation issue
+ introduced in r354429 by re-adding the 'quote' variable. (closes
+ issue ASTERISK-19337) Reported by: John Taylor
+
+2012-02-09 22:06 +0000 [r354751] Terry Wilson <twilson@digium.com>
+
+ * /, funcs/func_cdr.c: Note that CDRs are immutable once a bridge
+ is torn down CDRs cannot be modified after a bridge is torn down,
+ (e.g. after Dial() returns) even though the CDR() function may be
+ called. Since modifying the CDR code to change this behavior
+ could very easily break all kinds of things, this patch just
+ documents this limitation. (closes issues ASTERISK-16923) Review:
+ https://reviewboard.asterisk.org/r/1720/ ........ Merged
+ revisions 354749 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 354750 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-09 20:52 +0000 [r354657-354704] Kinsey Moore <kmoore@digium.com>
+
+ * /, 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/
+ ........ Merged revisions 354702 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 354703 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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/ ........ Merged
+ revisions 354655 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 354656 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-09 18:14 +0000 [r354597] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c, channels/sip/include/config_parser.h,
+ channels/sip/utils.c (added), configs/sip.conf.sample, CHANGES,
+ channels/sip/config_parser.c, channels/sip/include/sip.h,
+ channels/sip/include/sip_utils.h: Add auto_force_rport and
+ auto_comedia NAT options This patch adds the auto_force_rport and
+ auto_comedia NAT options. It also converts the nat= setting to a
+ list of comma-separated combinable options: no, force_rport,
+ comedia, auto_force_rport, and auto_comedia. nat=yes remains as
+ an undocumented option equal to "force_rport,comedia". The first
+ instance of 'yes' or 'no' in the list stops parsing and overrides
+ any previously set options. If an auto_* option is specified with
+ its non-auto_ counterpart, the auto setting takes precedence.
+ This patch builds upon the patch posted to ASTERISK-17860 by JIRA
+ user pedro-garcia. (closes issue ASTERISK-17860) Review:
+ https://reviewboard.asterisk.org/r/1698/
+
+2012-02-09 17:17 +0000 [r354552] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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 ........ Merged revisions
+ 354545 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 354546 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-09 17:09 +0000 [r354544-354549] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Clean-up of minor formatting issues in
+ r354542/3/4 rmudgett pointed out some formatting issues in the
+ check-in for ASTERISK-19290. This cleans those up. Review:
+ https://reviewboards.asterisk.org/r/1722/ ........ Merged
+ revisions 354547 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 354548 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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 ........
+ Merged revisions 354542 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 354543 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-09 03:09 +0000 [r354497-354498] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/chan_misdn.c: Fix some compile
+ problems from the 'cppcheck' patch.
+
+ * /, 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 ........ Merged revisions 354495 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 354496 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-09 02:28 +0000 [r354494] Russell Bryant <russell@russellbryant.com>
+
+ * main/channel.c, /: Remove some unnecessary locking from
+ ast_hangup(). This patch removes some unnecessary locking of the
+ channels container in ast_hangup(). The reason this came up is
+ that this lock can very quickly block the entire system. If any
+ of the channel cleanup code decides to block, it causes a problem
+ for the whole system. For example, when audiohooks get destroyed,
+ if that blocks for a while waiting on the mixmonitor thread to
+ exit because it's busy blocking on some I/O, it causes a problem
+ for many other threads in the meantime. Review:
+ https://reviewboard.asterisk.org/r/1712/ ........ Merged
+ revisions 354492 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 354493 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-08 21:29 +0000 [r354459] Kevin P. Fleming <kpfleming@digium.com>
+
+ * res/res_ais.c (removed), contrib/scripts/install_prereq: Revision
+ 354046 added res_corosync as a replacement for res_ais, but
+ didn't actually remove res_ais. This commit removes it. In
+ addition, the 'install_prereq' script has been updated to no
+ longer install AIS dependency packages, and instead install
+ Corosync packages instead.
+
+2012-02-08 21:28 +0000 [r354458] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c, contrib/realtime/postgresql/realtime.sql,
+ CHANGES, channels/sip/include/sip.h: Add callbackextension
+ matching & realtime callbackextensions This patch is based on the
+ one by David Vossel, developer extrodinaire, at
+ https://reviewboard.asterisk.org/r/344/. If multiple peers are
+ defined with the same host/port, but differing
+ callbackextensions, it chooses the peer with the matching
+ callbackextension. Since callbackextension creates an outbound
+ registration with the callbackextension as the Contact address,
+ matching an incoming request by that (in addition to the
+ host/port) makes a lot of sense. This patch also adds support for
+ callbackextension to realtime by querying all peers with
+ callbackextensions on reload and adding registrations for them.
+ (closes issue ASTERISK-13456) Review:
+ https://reviewboard.asterisk.org/r/344/ Review:
+ https://reviewboard.asterisk.org/r/1717/
+
+2012-02-08 21:25 +0000 [r354450] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: Restore some variables removed by the
+ 'cppcheck' patch that were actually needed.
+
+2012-02-08 20:49 +0000 [r354429] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * apps/app_dial.c, main/udptl.c, main/pbx.c, addons/chan_ooh323.c,
+ funcs/func_env.c, funcs/func_strings.c, utils/astman.c,
+ main/acl.c, apps/app_disa.c, apps/app_alarmreceiver.c,
+ apps/app_queue.c, channels/chan_iax2.c,
+ addons/ooh323c/src/memheap.c, channels/chan_usbradio.c,
+ channels/chan_dahdi.c, apps/app_osplookup.c,
+ channels/chan_misdn.c, channels/chan_skinny.c, funcs/func_odbc.c,
+ main/ast_expr2f.c, apps/app_minivm.c, formats/format_h263.c,
+ addons/chan_mobile.c, apps/app_chanspy.c, main/ast_expr2.fl,
+ apps/app_voicemail.c: Avoid cppcheck warnings; removing unused
+ vars and a bit of cleanup. Patch by: Clod Patry Review:
+ https://reviewboard.asterisk.org/r/1651
+
+2012-02-08 15:28 +0000 [r354395] Kinsey Moore <kmoore@digium.com>
+
+ * CHANGES: Add CHANGES documentation for the "pri set debug"
+ bitmask change (related to ASTERISK-17159)
+
+2012-02-07 21:33 +0000 [r354360] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c, contrib/realtime/postgresql/realtime.sql:
+ 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/
+ ........ Merged revisions 354348 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 354349 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-07 18:07 +0000 [r354312-354314] Sean Bright <sean@malleable.com>
+
+ * contrib/scripts/live_ast: Continuation of last patch - since
+ LIVE_AST_LD_PATH_EXTRA will now never be empty, don't check for
+ it, instead of check if LD_LIBRARY_PATH is already set and if so,
+ append LIVE_AST_LD_PATH_EXTRA properly.
+
+ * contrib/scripts/live_ast: Include live/usr/lib in the shared
+ library search path to that we pick up libasteriskssl.so at run
+ time when using live_ast.
+
+ * contrib/scripts/live_ast: Whitespace only (remove trailing
+ spaces)
+
+2012-02-07 15:29 +0000 [r354275] Jonathan Rose <jrose@digium.com>
+
+ * /, cdr/cdr_pgsql.c: Fix column duplication bug in module reload
+ for cdr_pgsql. Prior to this patch, attempts to reload
+ cdr_pgsql.so would cause the column list to keep its current data
+ and then add a second copy during the reload. This would cause
+ attempts to log the CDR to the database to fail. This patch also
+ cleans up some unnecessary null checks for ast_free and deals
+ with a few potential locking problems. (closes issue
+ ASTERISK-19216) Reported by: Jacek Konieczny Review:
+ https://reviewboard.asterisk.org/r/1711/ ........ Merged
+ revisions 354263 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 354270 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-06 23:15 +0000 [r354174-354218] Richard Mudgett <rmudgett@digium.com>
+
+ * /, pbx/pbx_config.c: Improved documentation of CLI "dialplan add
+ extension" command. * Documented dialplan add extension
+ <exten>,<priority>,<app(<app-data>)> format. * Allow acceptance
+ of command without the app-data value. There are many
+ applications that do no need any parameters so it is silly to
+ require that field for all commands. * Fixed a couple
+ ast_malloc/ast_free mismatches with ast_add_extension2() calls.
+ (closes issue ASTERISK-19222) Reported by: Andrey Solovyev Tested
+ by: rmudgett ........ Merged revisions 354216 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 354217 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * channels/sig_pri.h: Restore alternate SIG_PRI_DEBUG_DEFAULT
+ meaning.
+
+2012-02-06 20:18 +0000 [r354165] Kinsey Moore <kmoore@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c: Allow more control
+ over the output of pri debug This changes the debuglevel of 'pri
+ set debug' to a bit mask allowing the user to independently
+ select bits of output: 1 libpri internals including state machine
+ 2 Decoded Q.931 messages 4 Decoded Q.921 headers 8 raw hex dump
+ of the full frames Additionally, this ensures that the meaning of
+ "on" does not change and intrudces intense and hex to simplify
+ usage. (closes issue ASTERISK-17159) Original-patch-by: wimpy
+
+2012-02-06 17:33 +0000 [r354120] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Add missing headers to AMI UnParkedCall event
+ to uniquely identify the call. The AMI UnParkedCall event was
+ missing the Parkinglot and Uniqueid headers that the AMI
+ ParkedCall event contains. (closes issue ASTERISK-19240) Reported
+ by: Michael Yara ........ Merged revisions 354116 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 354119 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-06 16:38 +0000 [r354084] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_meetme.c, UPGRADE.txt: Make the 'c' option to MeetMe
+ work even if the 'q' option is used. (closes issue
+ ASTERISK-17053) Reported by: justdave
+
+2012-02-05 10:58 +0000 [r354046] Russell Bryant <russell@russellbryant.com>
+
+ * build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, res/res_corosync.c (added),
+ configure.ac, configs/res_corosync.conf.sample (added), res/ais
+ (removed), UPGRADE.txt, configs/ais.conf.sample (removed),
+ CHANGES, makeopts.in: Replace res_ais with a new module,
+ res_corosync. This patch removes res_ais and introduces a new
+ module, res_corosync. The OpenAIS project is deprecated and is
+ now just a wrapper around Corosync. This module provides the same
+ functionality using the same core infrastructure, but without the
+ use of the deprecated components. Technically res_ais could have
+ been used with an AIS implementation other than OpenAIS, but that
+ is the only one I know of that was ever used. Review:
+ https://reviewboard.asterisk.org/r/1700/
+
+2012-02-03 21:33 +0000 [r354001] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_agent.c: Fixes deadlocks occuring in chan_agent
+ due to r335976 Bad locking order was added to chan_agent to
+ prevent segfaults from having no locking in a patch by irroot.
+ This patch addresses the bad locking order by releasing locks
+ before getting the right locking order to stop deadlocks from
+ occuring when doing multiple interactions with agents. (closes
+ issue ASTERISK-19285) Reported by: Alex Villacis Lasso Review:
+ https://reviewboard.asterisk.org/r/1708/ ........ Merged
+ revisions 353999 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 354000 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-03 16:50 +0000 [r353964] Kinsey Moore <kmoore@digium.com>
+
+ * UPGRADE.txt, cdr/cdr_adaptive_odbc.c,
+ configs/cdr_adaptive_odbc.conf.sample: Support schema selection
+ in cdr_adaptive_odbc Asterisk now supports using ODBC with
+ databases where a single schema must be selected. Previously,
+ INSERTs would fail because they did not take into account extra
+ fields cause by having multiple schemas. This also corrects some
+ SQL resource leaks. (closes issue ASTERISK-17106) Patch-by:
+ Alexander Frolkin Patch-by: Tilgnman Lesher
+
+2012-02-03 16:23 +0000 [r353963] Jonathan Rose <jrose@digium.com>
+
+ * /, res/res_fax.c: Fixes a segfault occuring when performing
+ attended transfer with FAXOPT(gateway)=yes (closes issue
+ ASTERISK-19184) Reported by: Alexandr ........ Merged revisions
+ 353962 from http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-02 22:28 +0000 [r353917] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Ensure entering T.38 passthrough does not
+ cause an infinite loop After R340970 Asterisk was still polling
+ the RTCP file descriptor after RTCP is shut down and removed. If
+ the descriptor happened to have data ready when the removal
+ occured then Asterisk would go into an infinite loop trying to
+ read data that it can never actually access. This change disables
+ the audio RTCP file descriptor for the duration of the T.38
+ transaction. (closes issue ASTERISK-18951) Reported-by: Kristijan
+ Vrban ........ Merged revisions 353915 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 353916 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-02 20:18 +0000 [r353872] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.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 ........ Merged
+ revisions 353867 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 353868 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-02 18:55 +0000 [r353821] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c, /, main/http.c, configs/manager.conf.sample,
+ include/asterisk/manager.h, configs/http.conf.sample: 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 ........ Merged
+ revisions 353770 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 353820 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-02 17:07 +0000 [r353725-353772] Jonathan Rose <jrose@digium.com>
+
+ * /, 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) ........ Merged revisions 353769
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 353771 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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) ........ Merged revisions 353720
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 353721 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-01 19:53 +0000 [r353647-353685] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_unistim.c, channels/chan_multicast_rtp.c,
+ channels/chan_local.c, addons/chan_ooh323.c,
+ channels/chan_vpb.cc, channels/chan_gtalk.c,
+ channels/chan_iax2.c, main/channel.c, channels/chan_phone.c,
+ channels/chan_dahdi.c, channels/sig_analog.c, main/manager.c,
+ pbx/pbx_spool.c, channels/chan_skinny.c, main/features.c,
+ channels/sig_analog.h, channels/chan_alsa.c,
+ apps/app_confbridge.c, addons/chan_mobile.c, channels/sig_ss7.c,
+ channels/chan_mgcp.c, main/pbx.c, channels/sig_ss7.h,
+ channels/chan_sip.c, channels/chan_bridge.c,
+ channels/chan_agent.c, include/asterisk/channel.h,
+ channels/chan_console.c, channels/sig_pri.c, channels/chan_oss.c,
+ channels/chan_usbradio.c, channels/chan_jingle.c,
+ channels/sig_pri.h, channels/chan_misdn.c, channels/chan_h323.c,
+ channels/chan_nbs.c, include/asterisk/pbx.h: Constify some more
+ channel driver technology callback parameters. Review:
+ https://reviewboard.asterisk.org/r/1707/
+
+ * cel/cel_pgsql.c, configs/cel_sqlite3_custom.conf.sample,
+ cel/cel_odbc.c, configs/cel.conf.sample, cel/cel_manager.c,
+ cel/cel_tds.c, configs/cel_pgsql.conf.sample,
+ configs/cel_odbc.conf.sample, main/cel.c,
+ configs/cel_custom.conf.sample: Remove inconsistency in CEL
+ eventtype for user defined events. The CEL eventtype field for
+ ODBC and PGSQL backends should be USER_DEFINED instead of the
+ user defined event name supplied by the CELGenUserEvent
+ application. If the field is output as a number, the user defined
+ name does not have a value and is always output as 21 for
+ USER_DEFINED and the userdeftype field would be required to
+ supply the user defined name. The following CEL backends
+ (cel_odbc, cel_pgsql, cel_custom, cel_manager, and
+ cel_sqlite3_custom) can be independently configured to remove
+ this inconsistency. * Allows cel_manager, cel_custom, and
+ cel_sqlite3_custom to behave the same way. (closes issue
+ ASTERISK-17189) Reported by: Bryant Zimmerman Review:
+ https://reviewboard.asterisk.org/r/1669/
+
+ * main/channel.c, include/asterisk/channel.h: Fix ExtenSpy and
+ simplify the channel search functions. When ast_channel name was
+ opaquified, the channel search functions did not get converted
+ correctly. As a result ExtenSpy which uses a channel iterator
+ search by exten@context could never find anything. * Updated the
+ doxygen documentation for the search functions in channel.h.
+ Review: https://reviewboard.asterisk.org/r/1702/
+
+2012-02-01 15:59 +0000 [r353600] Sean Bright <sean@malleable.com>
+
+ * /, include/asterisk/audiohook.h: Resolve an overlap in the
+ ast_audiohook_flags values. AST_AUDIOHOOK_TRIGGER_WRITE and
+ AST_AUDIOHOOK_WANTS_DTMF were overlapping which may have caused
+ unintended side effects. This patch moves
+ AST_AUDIOHOOK_TRIGGER_WRITE, and updates
+ AST_AUDIOHOOK_TRIGGER_MODE to reflect the original intention.
+ This will affect existing modules that use these flags, so be
+ sure to recompile as necessary. (closes issue ASTERISK-19246)
+ Reported by: feyfre ........ Merged revisions 353598 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 353599 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-01 15:07 +0000 [r353552] Matthew Jordan <mjordan@digium.com>
+
+ * /, contrib/init.d/etc_default_asterisk: Added clarification for
+ the VERBOSITY setting to etc_default_asterisk Clarified that
+ using the VERBOSITY setting in etc_default_asterisk is the same
+ as using the -v command line switch, which causes Asterisk to
+ launch in console mode. (closes issue ASTERISK-17030) Reported
+ by: Jonas ........ Merged revisions 353550 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 353551 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-02-01 00:08 +0000 [r353504] Terry Wilson <twilson@digium.com>
+
+ * /, res/res_calendar.c: Allow res_calendar to be unloaded The
+ calendaring tech modules depend on res_calendar and initially
+ res_calendar just bumped the use count so that it couldn't be
+ unloaded. res_calendar can potentially create many threads and
+ I've seen issues where the Asterisk shutdown has failed where it
+ looked like these threads could be the culprit. This patch adds
+ unload support for res_calendar. Unloading res_calendar will also
+ unload the dependant tech modules as well. (closes issue
+ ASTERISK-16744) Review: https://reviewboard.asterisk.org/r/1657/
+ ........ Merged revisions 353502 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 353503 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-31 17:26 +0000 [r353466] Richard Mudgett <rmudgett@digium.com>
+
+ * main/manager.c, /, include/asterisk/channel.h: 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/
+ ........ Merged revisions 353454 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 353463 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-30 23:58 +0000 [r353418] Terry Wilson <twilson@digium.com>
+
+ * main/dnsmgr.c, /, channels/chan_sip.c, include/asterisk/dnsmgr.h:
+ 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/
+ ........ Merged revisions 353371 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 353397 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-30 22:44 +0000 [r353347-353370] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, channels/chan_sip.c: Merged revisions 353369 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r353369 | alecdavis | 2012-01-31 11:42:28 +1300
+ (Tue, 31 Jan 2012) | 9 lines Merged revisions 353368 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r353368 | alecdavis | 2012-01-31 11:40:40 +1300 (Tue, 31
+ Jan 2012) | 2 lines prevent debug messsges displaying -ve Cseq
+ numbers. Missed in R353320 ........ ................
+
+ * channels/sip/include/dialog.h, /, channels/chan_sip.c,
+ channels/sip/include/sip.h: Merged revisions 353321 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r353321 | alecdavis | 2012-01-31 11:16:22 +1300
+ (Tue, 31 Jan 2012) | 25 lines Merged revisions 353320 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r353320 | alecdavis | 2012-01-31 10:57:49 +1300 (Tue, 31 Jan
+ 2012) | 18 lines 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/ ........
+ ................
+
+2012-01-30 21:34 +0000 [r353262-353319] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile: Correct serious flaw in the top-level Makefile.
+
+ * include/asterisk.h, /, main/Makefile, main/libasteriskssl.c
+ (added), configure.ac, Makefile.moddir_rules, main/ssl.c
+ (removed), addons, CHANGES, include/asterisk/optional_api.h,
+ Makefile, build_tools/mkpkgconfig, configure, main, makeopts.in,
+ build_tools/make_defaults_h, main/libasteriskssl.exports.in
+ (added): Address OpenSSL initialization issues when using
+ third-party libraries. When Asterisk is used with various
+ third-party libraries (CURL, PostgresSQL, many others) that have
+ the ability themselves to use OpenSSL, it is possible for
+ conflicts to arise in how the OpenSSL libraries are initialized
+ and shutdown. This patch addresses these conflicts by 'wrapping'
+ the important functions from the OpenSSL libraries in a new
+ shared library that is part of Asterisk itself, and is loaded in
+ such a way as to ensure that *all* calls to these functions will
+ be dispatched through the Asterisk wrapper functions, not the
+ native functions. This new library is optional, but enabled by
+ default. See the CHANGES file for documentation on how to disable
+ it. Along the way, this patch also makes a few other minor
+ changes: * Changes MODULES_DIR to ASTMODDIR throughout the build
+ system, in order to more closely match what is used during
+ run-time configuration. * Corrects some errors in the configure
+ script where AC_CHECK_TOOLS was used instead of AC_PATH_PROG. *
+ Adds a new variable for linker flags in the build system
+ (DYLINK), used for producing true shared libraries (as opposed to
+ the dynamically loadable modules that the build system produces
+ for 'regular' Asterisk modules). * Moves the Makefile bits that
+ handle installation and uninstallation of the main Asterisk
+ binary into main/Makefile from the top-level Makefile. * Moves a
+ couple of useful preprocessor macros from optional_api.h to
+ asterisk.h. Review: https://reviewboard.asterisk.org/r/1006/
+
+ * /, 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). ........ Merged revisions 353260
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 353261 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-29 22:33 +0000 [r353224] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Allow softkey reject while device onhook.
+ Fixes up softkey endcall. Previous code was a copy of onhook, now
+ allows for endcall softkey to be used while device is still
+ onhook.
+
+2012-01-29 02:45 +0000 [r353177] Russell Bryant <russell@russellbryant.com>
+
+ * /, main/netsock.c: Find even more network interfaces. The
+ previous change made the code look for emN and pciN in addition
+ to what it did originally, which was search for ethN. However, it
+ needed to be looking for pciN#N, so that's what it does now. This
+ also moves the memset() to be before every ioctl(). ........
+ Merged revisions 353175 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 353176 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-28 14:52 +0000 [r353128] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/rtp_engine.c, /: Add 'L16-256' MIME subtype alias for
+ slin16. Asterisk has supported the 'L16' MIME subtype for 16kHz
+ signed linear (PCM) audio for quite some time, but some endpoints
+ refer to it as 'L16-256'. This commit adds this as an alias for
+ the existing format. ........ Merged revisions 353126 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 353127 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-28 04:31 +0000 [r353079] Russell Bryant <russell@russellbryant.com>
+
+ * /, main/netsock.c: Update ast_set_default_eid() to find more
+ network interfaces. As of Fedora 15, ethN is not the name of
+ ethernet interfaces. The names are emN or pciN. Update some code
+ that searched for interfaces named ethN to look for the new
+ names, as well. For more information about why this change was
+ made, see this page: http://domsch.com/blog/?p=455 ........
+ Merged revisions 353077 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 353078 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-27 21:38 +0000 [r352996-353040] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_queue.c: Audit of ao2_iterator_init() usage for v10.
+ Missed one. ........ Merged revisions 353039 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, tests/test_format_api.c: Audit of ao2_iterator_init() usage
+ for v10. Fix double format_cap iterator cleanup. ........ Merged
+ revisions 352992 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-27 19:26 +0000 [r352981] Jonathan Rose <jrose@digium.com>
+
+ * /, res/res_monitor.c: Make failed PauseMonitor and UnpauseMonitor
+ with no valid channel not close AMI session. I also went ahead
+ and took a little time to make sure that the manager value
+ AMI_SUCCESS was used instead of just return 0 being thrown around
+ everywhere since that's how we handle this stuff these days.
+ (closes issue ASTERISK-19249) Reporter: Jamuel Starkey Patches:
+ res_monitor.c-ASTERISK-19249.diff uploaded by Jamuel Starkey
+ (license 5766) ........ Merged revisions 352959 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 352965 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-27 18:47 +0000 [r352957] Richard Mudgett <rmudgett@digium.com>
+
+ * main/pbx.c, /, channels/chan_sip.c,
+ include/asterisk/indications.h, 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: 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/ ........ Merged
+ revisions 352955 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 352956 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-27 15:57 +0000 [r352916] Terry Wilson <twilson@digium.com>
+
+ * res/res_calendar_exchange.c, res/res_calendar_caldav.c,
+ res/res_calendar.c: Add aresult variable for CALENDAR_WRITE This
+ patch adds a CALENDAR_SUCCESS=1/0 variable that is set to show
+ whether or not CALENDAR_WRITE has passed. This patch also adds
+ some debugging for caldav PUT responses and no longer treats
+ responses with no body as an error (as a PUT gets a 201 Created
+ with no body). (closes issue ASTERISK-16903) Reported by: Clod
+ Patry Tested by: Terry Wilson Patches: calendarstatus.diff
+ uploaded by Clod Patry (License #5138), slightly modified by
+ Terry Wilson Review: https://reviewboard.asterisk.org/r/1692/ -
+ This line, and those below, will be ignored-- M
+ res/res_calendar.c M res/res_calendar_exchange.c M
+ res/res_calendar_caldav.c
+
+2012-01-27 00:11 +0000 [r352864] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged
+ revisions 352863 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r352863 | alecdavis | 2012-01-27 13:08:03 +1300
+ (Fri, 27 Jan 2012) | 19 lines Merged revisions 352862 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r352862 | alecdavis | 2012-01-27 13:05:30 +1300 (Fri, 27 Jan
+ 2012) | 12 lines 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 20:44 +0000 [r352821] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c, /: Fix outbound DTMF for inband mode (tell
+ asterisk core to generate DTMF sounds). (Closes issue
+ ASTERISK-19233) Reported by: Matt Behrens Patches:
+ chan_ooh323.c.patch uploaded by Matt Behrens (License #6346)
+ ........ Merged revisions 352807 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 352817 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-26 19:09 +0000 [r352757] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: Copy amaflags to sip_pvt from peer during
+ create_addr_from_peer For whatever reason, we don't have a single
+ function for copying data like this from SIP peers to the SIP
+ pvt. This patch adds the copying of amaflags to the sip_pvt, but
+ it would probably be worth discussing this function along with
+ the others that essentially just copy some amount of data from a
+ peer to a private. (Closes issue ASTERISK-19029) Reported by:
+ Matt Lehner ........ Merged revisions 352755 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 352756 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-26 06:36 +0000 [r352706] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, channels/chan_sip.c: Merged revisions 352705 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r352705 | alecdavis | 2012-01-26 19:33:11 +1300
+ (Thu, 26 Jan 2012) | 27 lines Merged revisions 352704 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r352704 | alecdavis | 2012-01-26 19:27:07 +1300 (Thu, 26 Jan
+ 2012) | 20 lines Cleanup dialog-info+xml Notify dialog Make
+ similar to other Notify messages. sample output: <?xml
+ version="1.0"?> <dialog-info
+ xmlns="urn:ietf:params:xml:ns:dialog-info" version="715"
+ state="full" entity="sip:8523@192.168.x.xx"> <dialog id="8523">
+ <state>terminated</state> </dialog> </dialog-info> Tested with
+ Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585)
+ Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1693/ ........
+ ................
+
+2012-01-25 22:25 +0000 [r352659] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, apps/app_voicemail.c: Fix -Werror=unused-but-set-variable
+ compiler error (gcc 4.6.2) ........ Merged revisions 352643 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 352651 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-25 21:31 +0000 [r352626] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, include/asterisk/version.h (added), main/test.c,
+ build_tools/make_version_h (removed), include/asterisk: Remove
+ "asterisk/version.h" in favor of "asterisk/ast_version.h". A long
+ time ago, in a land far far away, we added
+ "asterisk/ast_version.h", which provides the ast_get_version()
+ and ast_get_version_num() functions. These were added so that
+ modules that needed the version information for the Asterisk
+ instance they were loaded in could actually get it (as opposed
+ the version that they were compiled against). We changed
+ everything in the tree to use the new mechanism (although later
+ main/test.c was added using the old method). However, the old
+ mechanism was never removed, and as a result, new code is still
+ trying to use it. This commit removes asterisk/version.h and
+ replaces it with a header that will generate a compile-time error
+ if you try to use it (the error message tells you which header
+ you should use instead). It also removes the Makefile and
+ build_tools bits that generated the file, and it updates
+ main/test.c to use the 'proper' method of getting the Asterisk
+ version information. This is an API change and thus is being
+ committed for trunk only, but it's a fairly minor one and
+ definitely improves the situation for out-of-tree modules.
+
+2012-01-25 17:33 +0000 [r352565] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Remove some extraneous debugging from
+ registry memleak fix ........ Merged revisions 352551 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 352556 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-25 17:23 +0000 [r352538] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c, CHANGES, main/message.c,
+ channels/sip/include/sip.h: Fixes for sending SIP MESSAGE outside
+ of calls. * Fix authenticate MESSAGE losing custom headers added
+ by the MESSAGE_DATA function in the authorization attempt. * Pass
+ up better From header contents for SIP to use. Now is in the
+ "display-name" <URI> format expected by MessageSend. (Note that
+ this is a behavior change that could concievably affect some
+ people.) * Block user from adding standard headers that are added
+ automatically. (To, From,...) * Allow the user to override the
+ Content-Type header contents sent by MessageSend. * Decrement
+ Max-Forwards header if the user transferred it from an incoming
+ message. * Expand SIP short header names so the dialplan and
+ other code only has to deal with the full names. * Documents what
+ SIP expects in the MessageSend(from) parameter. (closes issue
+ ASTERISK-18992) Reported by: Yuri (closes issue ASTERISK-18917)
+ Reported by: Shaun Clark Review:
+ https://reviewboard.asterisk.org/r/1683/ ........ Merged
+ revisions 352520 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-25 17:02 +0000 [r352519] Terry Wilson <twilson@digium.com>
+
+ * /, 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/ ........
+ Merged revisions 352514 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 352515 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-25 16:54 +0000 [r352517] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/format.c, /, main/format_cap.c, main/format_pref.c:
+ Eliminate unnecessary rebuilds of main/format*.c. These files
+ have no need to include "asterisk/version.h", and doing so forces
+ them to be rebuilt each time a Subversion checkout moves between
+ 'modified' and 'unmodified' states. ........ Merged revisions
+ 352516 from http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-25 16:42 +0000 [r352513] Jonathan Rose <jrose@digium.com>
+
+ * /, configs/sip.conf.sample: Redocuments sip types peer, user,
+ friend in sip.conf.sample There was faulty information in the
+ sample config describing user as a synonym for friend so it has
+ been changed to better elaborate on the differences between the
+ three entity types. (closes issue ASTERISK-15537) Reported by:
+ yarique ........ Merged revisions 352511 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 352512 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-25 01:21 +0000 [r352475] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_vpb.cc: Fix channel opaquification of stringfields
+ for chan_vpb
+
+2012-01-24 22:28 +0000 [r352431] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Don't do a DNS lookup on an outbound
+ REGISTER host if there is an outbound proxy configured. (closes
+ issue ASTERISK-16550) reported by: Olle Johansson ........ Merged
+ revisions 352424 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 352430 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-24 20:37 +0000 [r352377] Jonathan Rose <jrose@digium.com>
+
+ * /, sounds/Makefile: Set core sounds version to 1.4.22. Now that
+ we have the right license for the Russian 1.4.22 sounds as well
+ as the sounds for the Australian English 1.4.22 sounds, we can
+ finally set the sounds to use 1.4.22! (closes issue
+ ASTERISK-18978) Reported by: Cameron Twomey Patches:
+ confbridge.tar.001 uploaded by Cameron Twomey confbridge.tar.002
+ uploaded by Cameron Twomey ........ Merged revisions 352367 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 352373 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-24 20:12 +0000 [r352348] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c, addons/chan_ooh323.c, main/say.c,
+ apps/app_record.c, apps/app_sayunixtime.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/chan_alsa.c, pbx/pbx_realtime.c, apps/app_externalivr.c,
+ apps/app_dial.c, main/pbx.c, apps/app_page.c,
+ channels/chan_bridge.c, apps/app_privacy.c,
+ channels/chan_agent.c, apps/app_disa.c,
+ include/asterisk/channel.h, main/aoc.c, apps/app_talkdetect.c,
+ main/cel.c, res/res_monitor.c, apps/app_playback.c,
+ apps/app_speech_utils.c, channels/chan_misdn.c,
+ funcs/func_channel.c, apps/app_chanspy.c, apps/app_voicemail.c,
+ channels/chan_unistim.c, channels/chan_multicast_rtp.c,
+ apps/app_meetme.c, apps/app_dictate.c, apps/app_authenticate.c,
+ apps/app_readexten.c, apps/app_userevent.c,
+ res/res_musiconhold.c, channels/chan_gtalk.c,
+ apps/app_followme.c, main/cdr.c, main/channel.c,
+ channels/chan_phone.c, main/dial.c, main/manager.c,
+ apps/app_minivm.c, res/res_agi.c, main/app.c,
+ apps/app_confbridge.c, main/image.c, apps/app_directory.c,
+ addons/chan_mobile.c, apps/app_rpt.c, channels/chan_mgcp.c,
+ apps/app_parkandannounce.c, channels/chan_sip.c, res/res_fax.c,
+ main/channel_internal_api.c, channels/chan_console.c,
+ channels/sig_pri.c, apps/app_queue.c, channels/chan_oss.c,
+ funcs/func_global.c, channels/chan_jingle.c,
+ channels/chan_usbradio.c, channels/chan_h323.c, main/file.c,
+ res/snmp/agent.c, channels/chan_nbs.c, apps/app_stack.c,
+ addons/app_saycountpl.c: Opaquify channel stringfields Continue
+ channel opaque-ification by wrapping all of the stringfields.
+ Eventually, we will restrict what can actually set these
+ variables, but the purpose for now is to hide the implementation
+ and keep people from adding code that directly accesses the
+ channel structure. Semantic changes will follow afterward.
+ Review: https://reviewboard.asterisk.org/r/1661/
+
+2012-01-24 17:04 +0000 [r352293] Richard Mudgett <rmudgett@digium.com>
+
+ * /, funcs/func_odbc.c: Fix locking issues with channel datastores
+ in func_odbc.c. * Fixed a potential memory leak when an existing
+ datastore is manually destroyed by inline code instead of calling
+ ast_datastore_free(). (closes issue ASTERISK-17948) Reported by:
+ Archie Cobbs Review: https://reviewboard.asterisk.org/r/1687/
+ ........ Merged revisions 352291 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 352292 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-23 20:31 +0000 [r352229-352232] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/features.c: Fix grammar of comment. ........ Merged
+ revisions 352230 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 352231 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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 ........ Merged revisions
+ 352199 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 352228 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-23 19:22 +0000 [r352166] Matthew Jordan <mjordan@digium.com>
+
+ * /, res/res_fax_spandsp.c: Correctly apply FAXOPT settings (V17,
+ V27, V29) before starting spandsp layer While the FAXOPT function
+ could be used to set the modem capabilities, the input to that
+ function was not being applied correctly to the spandsp layer.
+ This patch applies the current model capabilities before starting
+ the spandsp layer. (closes issue: ASTERISK-16409) Reported by:
+ Kristijan Vrban Tested by: Matt Jordan, Matthew Nicholson
+ Patches: spandsp-modems-1.8.diff uploaded by mnicholson (license
+ 5081) spandsp-modems-10.diff uploaded by mnicholson (license
+ 5081) ........ Merged revisions 352144 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 352149 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-23 18:34 +0000 [r352093-352134] Jonathan Rose <jrose@digium.com>
+
+ * configs/musiconhold.conf.sample, res/res_musiconhold.c, CHANGES:
+ Add an announcement option to music-on-hold - plays sound when
+ put on hold/between songs This is a feature patch which allows an
+ 'announcement' option to be specified in musiconhold.conf which
+ should be set to the name of a sound. If a valid sound is
+ specified for this option, then it will be played on that music
+ on hold class whenever a channel bound to that class is put on
+ hold as well as when Asterisk is able to detect that a song has
+ ended before starting the next song (excludes external players).
+ (closes ASTERISK-18977) Reported by: Timo Teräs Patches:
+ asterisk-moh-announcement.diff uploaded by Timo Teräs (license
+ 5409)
+
+ * CHANGES, apps/app_mixmonitor.c: Adds the ability to stop specific
+ mixmonitors by using unique IDs set at monitor launch. MixMonitor
+ receives a new option i(channel_variable) which stores the unique
+ id at said variable. StopMixMonitor now accepts ID as an optional
+ argument, which if included will make StopMixMonitor specifically
+ target the mixmonitor on that particular channel. CLI commands
+ and AMI actions have been ammended to work with the IDs as well.
+ In addition, monitors across a channel can now be listed be
+ listed via CLI command "mixmonitor list <channel>" which will
+ display all of the mixmonitors active on that channel along with
+ the files they each have open. Created by Sergio González Martín.
+ (closes issue ASTERISK-19096) Reported by: Sergio González Martín
+ Review: https://reviewboard.asterisk.org/r/1643/ Review:
+ https://reviewboard.asterisk.org/r/1682/
+
+2012-01-23 17:36 +0000 [r352092] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Fix sip_cfg.notifycid to be set with the
+ defined enum values. The invalid value used when notifycid was
+ enabled was benign. As far as the code was concerned -1 and 1 are
+ equivalent. (closes issue ASTERISK-19232) Reported by: Eike
+ Kuiper ........ Merged revisions 352090 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 352091 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-21 00:23 +0000 [r352041] Richard Mudgett <rmudgett@digium.com>
+
+ * /, funcs/func_timeout.c, main/app.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 ........ Merged revisions 352029 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 352035 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-21 00:11 +0000 [r352018-352019] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Remove XXX comment that is not necessary.
+ ........ Merged revisions 352016 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 352017 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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/
+ ........ Merged revisions 352014 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 352015 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-20 23:05 +0000 [r351977] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Make CLI sip show channel list the complete
+ route set. (closes issue ASTERISK-16877) Reported by: klaus3000
+ Patches: show-complete-routeset-patch.txt (license #5054) patch
+ uploaded by klaus3000 (modified)
+
+2012-01-20 21:26 +0000 [r351939] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_sip.c, UPGRADE.txt: SIP session timeout AMI event
+ Add an AMI event in the Call category that is issued when a call
+ is terminated due to either RTP stream inactivity or SIP session
+ timer expiration. Event description: Event: SessionTimeout
+ Source: source Channel: channel-name Uniqueid: channel-unique-id
+ `source` can be either RTPTimeout or SIPSessionTimer (closes
+ issue ASTERISK-16467) Patch-by: Kirill Katsnelson
+
+2012-01-20 20:47 +0000 [r351900-351913] Mark Michelson <mmichelson@digium.com>
+
+ * main/features.c, UPGRADE.txt, CHANGES,
+ configs/features.conf.sample: Various parking improvements. *
+ Adds per-parking lot options comebackcontext and comebackdialtime
+ * Makes comebacktoorigin settable per parking lot * Sets a PARKER
+ channel variable when comebacktoorigin is disabled (closes issue
+ ASTERISK-16643) Reported by: Mitch Sharp (bluecrow76) Patches:
+ asterisk-1.6.2.17.2-park-features-comebackcontext-consolidated-v3.diff
+ by Mitch Sharp (bluecrow76) license 5231 with updates by me.
+ Review: https://reviewboard.asterisk.org/r/1674 Review:
+ https://reviewboard.asterisk.org/r/963 Reviewed by Richard
+ Mudgett
+
+ * apps/app_mixmonitor.c: Prevent potential buffer overflow on AMI
+ MixMonitor command. Don't be alarmed. This only affected trunk,
+ and it would have required manager access to your system.
+
+2012-01-20 19:36 +0000 [r351817-351862] Kinsey Moore <kmoore@digium.com>
+
+ * /, codecs/ilbc/iLBC_test.c: More corrections for the ilbc code
+ These changes are in a file that is not compiled by default, and
+ so were missed on earlier checks. ........ Merged revisions
+ 351860 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 351861 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, codecs/ilbc/LPCencode.c, codecs/ilbc/iLBC_decode.c: Restore
+ LSF_check function calls from set/unused variable removal These
+ functions are not noops and modify the array that is passed in.
+ Thanks for the catch Richard. ........ Merged revisions 351818
+ from http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, codecs/ilbc/LPCencode.c, codecs/ilbc/iLBC_decode.c: Remove
+ more set, but unused variables in the ilbc codec GCC 4.6.3 caught
+ these in dev mode as well. ........ Merged revisions 351816 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-20 16:00 +0000 [r351764] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: Adds setting of mwi_from field to
+ check_auth_result check_peer_ok (closes ASTERISK-19057) Reported
+ By: Yuri Patches: 348360chan_sip.diff uploaded by Yuri (license
+ 5242) ........ Merged revisions 351759 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 351762 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-20 16:00 +0000 [r351763] Matthew Jordan <mjordan@digium.com>
+
+ * /, codecs/ilbc/helpfun.c: Remove unused variable 'tmp' from
+ helpfun in ilbc codec gcc version 4.6.2 caught an unused variable
+ in the ilbc codec library. This would prevent compilation with
+ --enable-dev-mode; variable removed. ........ Merged revisions
+ 351760 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 351761 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-20 13:12 +0000 [r351709] Stefan Schmidt <sst@sil.at>
+
+ * /, contrib/asterisk-ng-doxygen: enable doxygen build for files in
+ the channels/sip folder like reqresp_parser.c ........ Merged
+ revisions 351707 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 351708 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-19 23:31 +0000 [r351667] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/reqresp_parser.c: Misc minor
+ fixes in reqresp_parser.c and chan_sip.c. * Fix corner cases in
+ get_calleridname() parsing and ensure that the output buffer is
+ nul terminated. * Make get_calleridname() truncate the name it
+ parses if the given buffer is too small rather than abandoning
+ the parse and not returning anything for the name. Adjusted
+ get_calleridname_test() unit test to handle the truncation
+ change. * Fix get_in_brackets_test() unit test to check the
+ results of get_in_brackets() correctly. * Fix
+ parse_name_andor_addr() to not return the address of a local
+ buffer. This function is currently not used. * Fix potential NULL
+ pointer dereference in sip_sendtext(). * No need to
+ memset(calleridname) in check_user_full() or tmp_name in
+ get_name_and_number() because get_calleridname() ensures that it
+ is nul terminated. * Reply with an accurate response if
+ get_msg_text() fails in receive_message(). This is academic in
+ v1.8 because get_msg_text() can never fail. ........ Merged
+ revisions 351618 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 351646 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-19 22:44 +0000 [r351613] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_rtp_asterisk.c, /: Correct output of RTCP jitter
+ statistics in SR and RR reports Change the RTCP RR and SR
+ generation code to convert Asterisk's internal jitter statistics
+ to be represented in RTP timestamp units based on the rate of the
+ codec in use instead of in seconds. (closes issue ASTERISK-14530)
+ ........ Merged revisions 351611 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 351612 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-19 21:55 +0000 [r351561] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c, include/asterisk/netsock2.h: 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/
+ ........ Merged revisions 351559 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 351560 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-19 21:13 +0000 [r351506] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Prevent crash when an SDP offer is
+ received with an encrypted video stream when support for video is
+ disabled and res_srtp is loaded. (closes issue ASTERISK-19202)
+ Reported by: Catalin Sanda ........ Merged revisions 351504 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 351505 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-18 21:06 +0000 [r351452] Matthew Jordan <mjordan@digium.com>
+
+ * codecs/ilbc/syntFilter.c (added), /, codecs/ilbc/iCBConstruct.h
+ (added), codecs/ilbc/iLBC_test.c (added),
+ codecs/ilbc/syntFilter.h (added), 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/iLBC_define.h (added), codecs/ilbc/FrameClassify.c
+ (added), codecs/ilbc/enhancer.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/gainquant.c (added), codecs/ilbc/hpInput.c (added),
+ codecs/ilbc/hpOutput.c (added), codecs/ilbc/iCBSearch.h (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/ilbc/PATENTS (added),
+ codecs/ilbc/StateSearchW.c (added), codecs/ilbc/hpOutput.h
+ (added), codecs/codec_ilbc.c, 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): 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
+ ........ Merged revisions 351450 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 351451 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-18 16:02 +0000 [r351409] Stefan Schmidt <sst@sil.at>
+
+ * /, channels/chan_sip.c: The get_pai function in chan_sip.c didn't
+ recognized a proper callerid name and number from a
+ P-Asserted-Identity cause the header parsing logic was wrong.
+ Changing the parsing functions to the sip header parsing APIs in
+ reqresp_parser.h solves this problem. Review:
+ https://reviewboard.asterisk.org/r/1673 Reviewed by: wdoekes2 and
+ Mark Michelson ........ Merged revisions 351396 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 351408 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-17 19:45 +0000 [r351360] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * Makefile: Fix support for parallel building with make (-j).
+ Previously make -j <N> would cause a race between doing cleanup
+ of certain files (defaults.h, menuselect, ...) and creating them
+ anew. Add a new target that depends on cleanup only and has a
+ submake doing the rest as command string. This way the cleanup
+ goes first. (closes issue ASTERISK-18751) Tested by: Jeremy
+ Kister Reviewed by: Paul Belanger Review:
+ https://reviewboard.asterisk.org/r/1660
+
+2012-01-17 17:23 +0000 [r351311] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_rtp_asterisk.c, /: Eliminate odd initialization of
+ probation variable. ........ Merged revisions 351306 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 351308 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-17 17:15 +0000 [r351290] Jonathan Rose <jrose@digium.com>
+
+ * res/res_rtp_asterisk.c, /, configs/rtp.conf.sample, CHANGES: 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/
+ ........ Merged revisions 351287 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 351289 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-17 16:56 +0000 [r351288] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Use built-in parsing functions for
+ Contact and Record-Route headers. If a Contact or a Record-Route
+ header had a quoted string with an item in angle brackets, then
+ we would mis-parse it. For instance, "Bob <1234>"
+ <1234@example.org> would be misparsed as having the URI "1234"
+ The fix for this is to use parsing functions from
+ reqresp_parser.h since they are heavily tested and are awesome.
+ (issue ASTERISK-18990) ........ Merged revisions 351284 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 351286 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-17 16:08 +0000 [r351235] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Fix udptl issue with initial INVITE
+ introduced by r351027 When an inital INVITE occurs that contains
+ image media, a channel is not yet associated with the SIP dialog.
+ The file descriptor associated with the udptl session needs to be
+ set in initialize_udptl or in sip_new to account for this
+ scenario. ........ Merged revisions 351233 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 351234 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-17 01:48 +0000 [r351184] Russell Bryant <russell@russellbryant.com>
+
+ * /, channels/chan_sip.c: Merged revisions 351183 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r351183 | russell | 2012-01-16 20:43:19 -0500
+ (Mon, 16 Jan 2012) | 29 lines Merged revisions 351182 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r351182 | russell | 2012-01-16 20:37:03 -0500 (Mon, 16 Jan 2012)
+ | 22 lines 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 21:50 +0000 [r351082-351143] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Ensure ACK retransmit & hangup on non-200
+ response to INVITE When handling a non-2xx final response on an
+ INVITE transaction, we have to keep the transaction around after
+ we send an ACK in case we receive a retransmission of the
+ response so we can re-transmit the ACK, but also tear down the
+ ast_channel as soon as we transmit the ACK. Before this patch, we
+ could fail at both of these things. Calling
+ sip_alreadygone/needdestroy prevented us from keeping the
+ transaction up and retransmitting the ACK, and queueing
+ CONGESTION was not sufficient to cause the channel to be torn
+ down when originating calls via the CLI, for example. This patch
+ queues a hangup with CONGESTION instead of just queueing
+ CONGESTION for these responses and removes the sip_alreadygone
+ and sip_needdestroy calls from handle_response_invite on non-2xx
+ responses. It relies on the hangup calling sip_scheddestroy. For
+ more information, see section 17.1.1.1 of RFC 3261. (closes issue
+ ASTERISK-17717) Review: https://reviewboard.asterisk.org/r/1672/
+ ........ Merged revisions 351130 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 351131 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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) ........ Merged revisions 351080 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 351081 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-16 19:49 +0000 [r351079] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/ast_expr2.y, CHANGES, main/ast_expr2.c: Add ABS() absolute
+ value function to the expression parser.
+
+2012-01-16 19:13 +0000 [r351029] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Create and initialize udptl only when
+ dialog negotiates for image media Prior to this patch, the udptl
+ struct was allocated and initialized when a dialog was associated
+ with a peer that supported T.38, when a new SIP channel was
+ allocated, or what an INVITE request was received. This resulted
+ in any dialog associated with a peer that supported T.38 having
+ udptl support assigned to it, including the UDP ports needed for
+ communication. This occurred even in non-INVITE dialogs that
+ would never send image media. This patch creates and initializes
+ the udptl structure only when the SDP for a dialog specifies that
+ image media is supported, or when Asterisk indicates through the
+ appropriate control frame that a dialog is to support T.38.
+ (closes issue ASTERISK-16698) Reported by: under Tested by:
+ Stefan Schmidt Patches: udptl_20120113.diff uploaded by mjordan
+ (License #6283) (closes issue ASTERISK-16794) Reported by: Elazar
+ Broad Tested by: Stefan Schmidt review:
+ https://reviewboard.asterisk.org/r/1668/ ........ Merged
+ revisions 351027 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 351028 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-16 17:12 +0000 [r350979] Sean Bright <sean@malleable.com>
+
+ * /, main/db.c: Sort the output of 'database showkey' as well. You
+ can pass wildcards (%) to the database CLI commands, so this will
+ sort the returned list of matches. ........ Merged revisions
+ 350978 from http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-16 17:07 +0000 [r350977] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp_engine.c, /: Add missing code to set direct RTP setup
+ information during dialing. ........ Merged revisions 350975 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 350976 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-16 14:31 +0000 [r350939] Sean Bright <sean@malleable.com>
+
+ * /, main/db.c: Sort the output of 'database show' by key. This
+ more closely mimics the behavior of 'database show' before the
+ conversion to sqlite3. ........ Merged revisions 350938 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-15 20:16 +0000 [r350887-350890] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, main/asterisk.c: Allow only one thread at a time to do
+ asterisk cleanup/shutdown. Add locking around the
+ really-really-quit part of the core stop/restart part. Previously
+ more than one thread could be called to do cleanup, causing
+ atexit handlers to be run multiple times, in turn causing
+ segfaults. (issue ASTERISK-18883) Reviewed by: Terry Wilson
+ Review: https://reviewboard.asterisk.org/r/1662/ Review:
+ https://reviewboard.asterisk.org/r/1658/ ........ Merged
+ revisions 350888 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 350889 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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) ........ Merged
+ revisions 350885 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 350886 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-14 16:43 +0000 [r350791-350839] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac,
+ autoconf/libcurl.m4: 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. ........ Merged revisions 350837 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 350838 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, channels/chan_h323.c, addons/chan_mobile.c,
+ res/res_pktccops.c, contrib/scripts/install_prereq: Multiple
+ revisions 350788-350789 ........ r350788 | kpfleming | 2012-01-14
+ 09:22:33 -0600 (Sat, 14 Jan 2012) | 8 lines 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. ........ r350789 |
+ kpfleming | 2012-01-14 09:23:32 -0600 (Sat, 14 Jan 2012) | 3
+ lines Correct some 'set-but-not-used' variable warnings. ........
+ Merged revisions 350788-350789 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 350790 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-13 22:17 +0000 [r350738] Kinsey Moore <kmoore@digium.com>
+
+ * /, 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. ........ Merged revisions
+ 350736 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 350737 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-13 21:52 +0000 [r350735] Richard Mudgett <rmudgett@digium.com>
+
+ * /, configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample:
+ Correct eventtype names in cel_odbc and cel_pgsql sample files
+ ........ Merged revisions 350733 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 350734 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-13 21:42 +0000 [r350732] Kinsey Moore <kmoore@digium.com>
+
+ * /, configure.ac, bootstrap.sh, main/asterisk.c: 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 ........ Merged revisions
+ 350730 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 350731 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-13 20:32 +0000 [r350681] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/sip/config_parser.c: Set port to a default sane value
+ if a bogus one is provided when parsing hostnames. ........
+ Merged revisions 350679 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 350680 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-13 18:52 +0000 [r350605-350644] Richard Mudgett <rmudgett@digium.com>
+
+ * main/features.c: Remove some dead code in ast_bridge_call(). None
+ of the parameters to ast_bridge_call() can be NULL for the bridge
+ to work so no need to check for it.
+
+ * configs/cel_sqlite3_custom.conf.sample, cel/cel_odbc.c,
+ configs/cel.conf.sample, /, cel/cel_manager.c,
+ configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample,
+ main/cel.c, configs/cel_custom.conf.sample: Add missing CEL
+ logging fields to various CEL backends. Multiple revisions
+ 350555,350571 ........ r350555 | rmudgett | 2012-01-13 11:12:51
+ -0600 (Fri, 13 Jan 2012) | 12 lines 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
+ ........ r350571 | rmudgett | 2012-01-13 11:23:57 -0600 (Fri, 13
+ Jan 2012) | 8 lines 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) ........ Merged revisions 350555,350571
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 350585 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-13 17:00 +0000 [r350551-350554] Matthew Jordan <mjordan@digium.com>
+
+ * /, apps/app_queue.c: Realtime queues failed to load queue
+ information without queue member table Previously, realtime
+ queues could be loaded without defining the queue member table.
+ This allowed for queue members to be dynamic, while the realtime
+ queue definitions could exist in some backing storage. Revision
+ 342223 broke this when it changed the return value for
+ realtime_multientry to return NULL when no results are returned.
+ Previously, an empty ast_config object was expected. (closes
+ issue ASTERISK-19170) Reported by: Rene Mendoza Tested by: Rene
+ Mendoza Patches: rt_queue_member_patch.diff uploaded by Matt
+ Jordan (license 6283) ........ Merged revisions 350552 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 350553 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, bridges/bridge_builtin_features.c, channels/chan_bridge.c,
+ include/asterisk/bridging.h, apps/app_confbridge.c,
+ main/bridging.c: Fix crash from bridge channel hangup race
+ condition in ConfBridge This patch addresses two issues in
+ ConfBridge and the channel bridge layer: 1. It fixes a race
+ condition wherein the bridge channel could be hung up 2. It
+ removes the deadlock avoidance from the bridging layer and makes
+ the bridge_pvt an ao2 ref counted object Patch by David Vossel
+ (mjordan was merely the commit monkey) (issue ASTERISK-18988)
+ (closes issue ASTERISK-18885) Reported by: Dmitry Melekhov Tested
+ by: Matt Jordan Patches: chan_bridge_cleanup_v.diff uploaded by
+ David Vossel (license 5628) (closes issue ASTERISK-19100)
+ Reported by: Matt Jordan Tested by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1654/ ........ Merged
+ revisions 350550 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-12 16:10 +0000 [r350503] Jonathan Rose <jrose@digium.com>
+
+ * /, main/features.c: Adds peer to CEL report on CEL_BRIDGE_START
+ and CEL_BRIDGE_END (closes issue ASTERISK-17940) Reporter: Nic
+ Colledge Patches: features_18.patch uploaded by Nic Colledge
+ (license 6245) ........ Merged revisions 350501 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 350502 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-11 22:53 +0000 [r350416-350454] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/cel.c: Remove extraneous BRIDGEPEER AMI VarSet event on a
+ CEL dummy channel. (closes issue ASTERISK-19180) Reported by:
+ Corey Farrell Patches: asterisk_cel_noevent_varset.diff (license
+ #5909) patch uploaded by Corey Farrell ........ Merged revisions
+ 350452 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 350453 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * apps/app_dial.c, /, CHANGES, apps/app_followme.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. * Made 'N' option ignored if the call is
+ already answered. (closes issue ASTERISK-18969) Reported by:
+ rmudgett Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1656/ ........ Merged
+ revisions 350364 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 350415 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-11 19:19 +0000 [r350365] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c: Always treat arguments to get_by_name_cb as
+ strings Initially, support was left in for the old style of
+ searching, even though it wasn't actually used. In the case of
+ name_len != 0, the OBJ_KEY flag isn't passed because we aren't
+ matching on a full key and therefor can't use the hash function
+ to optimize. The code left in to support the old way of searching
+ unfortunately treated a prefix search like this as though an
+ ast_channel struct was passed as an arg and caused a crash. This
+ patch also adds needed parentheses around some matching
+ conditions. (closes issue ASTERISK-19182)
+
+2012-01-10 22:10 +0000 [r350273-350313] Richard Mudgett <rmudgett@digium.com>
+
+ * /, 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) ........ Merged revisions 350311 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 350312 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/channel.c: Fix compiler warnings reported by gcc v4.2.4.
+
+2012-01-09 22:15 +0000 [r350223] Terry Wilson <twilson@digium.com>
+
+ * main/udptl.c, apps/app_dahdibarge.c, addons/chan_ooh323.c,
+ channels/chan_local.c, main/rtp_engine.c, main/say.c,
+ apps/app_record.c, apps/app_test.c, channels/console_video.c,
+ apps/app_alarmreceiver.c, apps/app_chanisavail.c,
+ bridges/bridge_multiplexed.c, channels/chan_iax2.c,
+ main/indications.c, main/cli.c, channels/chan_dahdi.c,
+ channels/sig_analog.c, channels/chan_skinny.c, main/features.c,
+ apps/app_dumpchan.c, pbx/pbx_realtime.c, apps/app_amd.c,
+ channels/chan_alsa.c, apps/app_externalivr.c, main/bridging.c,
+ apps/app_milliwatt.c, channels/sig_ss7.c, apps/app_dial.c,
+ main/pbx.c, apps/app_page.c, apps/app_softhangup.c,
+ apps/app_fax.c, apps/app_dahdiras.c, channels/chan_agent.c,
+ apps/app_disa.c, include/asterisk/channel.h, main/aoc.c,
+ apps/app_talkdetect.c, main/cel.c, res/res_mutestream.c,
+ res/res_monitor.c, apps/app_playback.c, channels/chan_misdn.c,
+ funcs/func_channel.c, apps/app_macro.c, apps/app_mixmonitor.c,
+ apps/app_chanspy.c, apps/app_voicemail.c, res/res_calendar.c,
+ channels/chan_unistim.c, channels/chan_vpb.cc, main/ccss.c,
+ apps/app_meetme.c, apps/app_readexten.c, res/res_musiconhold.c,
+ main/autochan.c, channels/chan_gtalk.c, apps/app_followme.c,
+ res/res_jabber.c, main/cdr.c, main/channel.c, main/dial.c,
+ channels/chan_phone.c, main/manager.c, funcs/func_groupcount.c,
+ funcs/func_audiohookinherit.c, funcs/func_frame_trace.c,
+ res/res_agi.c, apps/app_minivm.c, main/app.c,
+ apps/app_confbridge.c, apps/app_rpt.c, addons/chan_mobile.c,
+ apps/app_parkandannounce.c, channels/chan_mgcp.c,
+ apps/app_jack.c, apps/app_adsiprog.c, channels/chan_sip.c,
+ res/res_fax.c, apps/app_waitforsilence.c, funcs/func_lock.c,
+ main/channel_internal_api.c (added), res/res_adsi.c,
+ pbx/pbx_lua.c, channels/chan_console.c, apps/app_getcpeid.c,
+ channels/sig_pri.c, apps/app_queue.c, channels/chan_oss.c,
+ funcs/func_global.c, channels/chan_usbradio.c,
+ channels/chan_jingle.c, apps/app_flash.c,
+ apps/app_directed_pickup.c, main/abstract_jb.c, main/file.c,
+ channels/chan_h323.c, res/snmp/agent.c, pbx/pbx_dundi.c,
+ apps/app_sms.c, channels/chan_nbs.c, apps/app_stack.c,
+ main/dsp.c: Replace direct access to channel name with accessor
+ functions There are many benefits to making the ast_channel an
+ opaque handle, from increasing maintainability to presenting ways
+ to kill masquerades. This patch kicks things off by taking things
+ a field at a time, renaming the field to
+ '__do_not_use_${fieldname}' and then writing setters/getters and
+ converting the existing code to using them. When all fields are
+ done, we can move ast_channel to a C file from channel.h and lop
+ off the '__do_not_use_'. This patch sets up
+ main/channel_interal_api.c to be the only file that actually
+ accesses the ast_channel's fields directly. The intent would be
+ for any API functions in channel.c to use the accessor functions.
+ No more monkeying around with channel internals. We should use
+ our own APIs. The interesting changes in this patch are the
+ addition of channel_internal_api.c, the moving of the AST_DATA
+ stuff from channel.c to channel_internal_api.c (note: the
+ AST_DATA stuff will have to be reworked to use accessor functions
+ when ast_channel is really opaque), and some re-working of the
+ way channel iterators/callbacks are handled so as to avoid
+ creating fake ast_channels on the stack to pass in matching data
+ by directly accessing fields (since "name" is a stringfield and
+ the fake channel doesn't init the stringfields, you can't use the
+ ast_channel_name_set() function). I went with
+ ast_channel_name(chan) for a getter, and
+ ast_channel_name_set(chan, name) for a setter. The majority of
+ the grunt-work for this change was done by writing a semantic
+ patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review:
+ https://reviewboard.asterisk.org/r/1655/
+
+2012-01-09 21:56 +0000 [r350222] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_iax2.c: Fix joinable thread terminating without
+ joiner memory leak in chan_iax.c. The iax2_process_thread() can
+ exit without anyone waiting to join the thread. If noone is
+ waiting to join the thread then a large memory leak occurs. *
+ Made iax2_process_thread() deatach itself if nobody is waiting to
+ join the thread. (closes issue ASTERISK-17339) Reported by:
+ Tzafrir Cohen Patches:
+ asterisk-1.8.4.4-chan_iax2-detach-thread-on-non-stop-exit.patch
+ (license #5617) patch uploaded by Alex Villacis Lasso (modified)
+ (closes issue ASTERISK-17825) Reported by: wangjin ........
+ Merged revisions 350220 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 350221 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-09 19:37 +0000 [r350181] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, main/db.c: Fix shutdown handling of sqlite3 astdb. If a
+ db_sync was scheduled just before shutdown, the atexit code
+ calling db_sync would have no effect, causing the astdb commit
+ thread to stay alive. This caused the SIP/realtime_sipregs test
+ to fail. (The fallback kill would run the atexit code again and
+ that would wreak havoc.) This fixes that the atexit kill
+ condition is picked up properly. (closes issue ASTERISK-18883)
+ Reviewed by: Terry Wilson Review:
+ https://reviewboard.asterisk.org/r/1659 ........ Merged revisions
+ 350180 from http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-09 18:58 +0000 [r350077-350130] Richard Mudgett <rmudgett@digium.com>
+
+ * /, contrib/scripts/valgrind_compare (added): Multiple revisions
+ 350127-350128 ........ r350127 | rmudgett | 2012-01-09 12:40:33
+ -0600 (Mon, 09 Jan 2012) | 12 lines 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 ........ r350128 |
+ rmudgett | 2012-01-09 12:54:56 -0600 (Mon, 09 Jan 2012) | 11
+ lines 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 ........
+ Merged revisions 350127-350128 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 350129 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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 ........
+ Merged revisions 350075 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 350076 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-09 15:40 +0000 [r350025] Kinsey Moore <kmoore@digium.com>
+
+ * /, apps/app_meetme.c: Prevent SLA settings from getting wiped out
+ on reload If SLA was reloaded without the config file being
+ changed, current settings got wiped out before the SLA reload
+ code decided it wasn't going to reload the file since nothing was
+ changed. Moving the settings reset later in the reload process
+ fixes this. (closes issue AST-744) ........ Merged revisions
+ 350023 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 350024 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-06 23:31 +0000 [r349978] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Don't leak CID in From header when
+ presentation=unavailable When someone does
+ Set(CALLERPRES()=unavailable) (or
+ Set(CALLERID(pres)=unavailable)) when sendrpid=no, the From
+ header shows "Anonymous" <anonymous@anonymous.invalid>. When
+ sendrpid=yes/pai, the From header will still display the callerid
+ info, even though we supply an rpid header with the anonymous
+ info. It seems like we shouldn't leak that info in any case.
+ Skimming http://tools.ietf.org/html/draft-ietf-sip-privacy-04
+ seems to indicate that one shouldn't send identifying info in the
+ From in this case. This patch anonymizes the From header as well
+ even when sendrpid=yes/pai. (closes issue ASTERISK-16538) Review:
+ https://reviewboard.asterisk.org/r/1649/ ........ Merged
+ revisions 349968 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 349977 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-06 21:26 +0000 [r349929] Kinsey Moore <kmoore@digium.com>
+
+ * /, pbx/pbx_lua.c: Fix lua goto detection to prevent unexpected
+ behavior with confbridge A bug in the pbx_lua goto detection was
+ causing the dialplan to hangup unexpectedly after confbridge
+ exited if it had called lua dialplan code during execution.
+ Patch-by: Timo Teras Acked-by: Matt Nicholson (closes issue
+ ASTERISK-18976) ........ Merged revisions 349928 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-06 16:50 +0000 [r349874] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_followme.c: Fix memory leaks in app_followme
+ find_realtime(). (closes issue ASTERISK-19055) Reported by: Matt
+ Jordan ........ Merged revisions 349872 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 349873 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-05 23:58 +0000 [r349823] Matthew Jordan <mjordan@digium.com>
+
+ * /, res/res_fax.c: Fix premature free'ing of the frame committed
+ in r349608 Even though we set the frame to the ast_null_frame and
+ return that, the caller of the frame hook may still need the
+ frame. This now is a bit more careful about when it frees the
+ frame, i.e., only under the same conditions that applied when we
+ duplicated it in the first place. ........ Merged revisions
+ 349822 from http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-05 23:47 +0000 [r349782-349821] Richard Mudgett <rmudgett@digium.com>
+
+ * /, 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 ........ Merged
+ revisions 349819 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 349820 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * UPGRADE.txt, pbx/pbx_config.c: Make pbx_config.c use Gosub
+ instead of Macro call for stdexten. Users created by users.conf
+ with hasvoicemail=yes have been documented as using a Gosub to
+ stdexten since v1.6.0. However, the code still generates dialplan
+ to access stdexten as a Macro as documented in v1.4; which does
+ not work with the newer extensions.conf.sample file. * Make
+ generated dialplan access the stdexten dialplan with the
+ documented Gosub instead of the older Macro style. (closes issue
+ ASTERISK-18809) Reported by: Jay Allen Patches:
+ gosub_patch-pbx_config.patch (license #6323) patch uploaded by
+ Jay Allen (modified) Tested by: rmudgett
+
+2012-01-05 22:11 +0000 [r349733] Kinsey Moore <kmoore@digium.com>
+
+ * /, main/file.c: Allow playback of formats that don't support
+ seeking ast_streamfile previously did unconditional seeking on
+ files that broke playback of formats that don't support that
+ functionality. This patch avoids the seek that was causing the
+ problem. This regression was introduced in r158062. (closes issue
+ ASTERISK-18994) Patch-by: Timo Teras ........ Merged revisions
+ 349731 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 349732 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-05 22:02 +0000 [r349674-349730] Jonathan Rose <jrose@digium.com>
+
+ * /, main/dsp.c: Fix an issue where dsp.c would interpret multiple
+ dtmf events from a single key press. When receiving calls from a
+ mobile phone into a DISA system on a connection with significant
+ interference, the reporter's Asterisk system would interpret DTMF
+ incorrectly and replicate digits received. This patch resolves
+ that by increasing the number of frames a mismatch has to be
+ detected before assuming the DTMF is over by 1 frame and adjusts
+ dtmf_detect function to reset hits and misses only when an edge
+ is detected. (closes issue ASTERISK-17493) Reported by: Alec
+ Davis Patches: bug18904-refactor.diff.txt uploaded by Alec Davis
+ (license 5546) Review: https://reviewboard.asterisk.org/r/1130/
+ ........ Merged revisions 349728 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 349729 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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) ........ Merged revisions 349672
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 349673 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-04 22:23 +0000 [r349609-349634] Matthew Jordan <mjordan@digium.com>
+
+ * /, apps/confbridge/conf_config_parser.c: Fix for ConfBridge
+ config parser unlocking channel mutex too many times When looking
+ up a ConfBridge profile, the config parser would, if it found a
+ channel datastore on the channel requesting the bridge profile,
+ unlock the channel mutex twice. Since that's a little aggressive,
+ it now only unlocks it once. (closes issue ASTERISK-19042)
+ Reported by: Matt Jordan Tested by: Matt Jordan Patches: 19042
+ uploaded by David Vossel (license 5628) ........ Merged revisions
+ 349619 from http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, res/res_fax.c: Free successfully translated frame in
+ fax_gateway_framehook A frame that is translated via
+ ast_translate is also duplicated via ast_frdup. This will
+ allocate a new frame on the heap, which needs to be free'd at the
+ appropriate time. This issue reporter used valgrind to find that
+ this occurred in res_fax's fax_gateway_framehook; a quick search
+ through the code showed that only place this was currently not
+ handling the translatted frame properly. (closes issue
+ ASTERISK-19133) Reported by: Sylvain Rochet ........ Merged
+ revisions 349608 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-04 20:55 +0000 [r349560] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Fix segfault in chan_dahdi for
+ CHANNEL(dahdi_span) evaluation on hangup. * Added NULL private
+ pointer checks in the following chan_dahdi channel callbacks:
+ dahdi_func_read(), dahdi_func_write(), dahdi_setoption(), and
+ dahdi_queryoption(). (closes issue ASTERISK-19142) Reported by:
+ Diego Aguirre Tested by: rmudgett ........ Merged revisions
+ 349558 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 349559 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-04 20:24 +0000 [r349506-349535] Kinsey Moore <kmoore@digium.com>
+
+ * contrib/init.d/rc.debian.asterisk, /: Make debian init script
+ conform to the LSB standard Previously, this init script would
+ return 1 if Asterisk was already running. This is incorrect
+ behavior according to the LSB standard and has been fixed by
+ returning 0 instead. (closes issue ASTERISK-17958) Reported-by:
+ johnc ........ Merged revisions 349529 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 349532 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, contrib/scripts/autosupport.8, contrib/scripts/autosupport:
+ 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) ........ Merged revisions 349504 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 349505 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-01-04 19:53 +0000 [r349452-349503] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: Adds Subscription-State header to notify
+ with call completion. per RFC3265 (Closes issue ASTERISK-17953)
+ Reported by: George Konopacki Patches: 19400.patch uploaded by
+ mmichelson (license 5049) ........ Merged revisions 349482 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 349502 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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 ........ Merged
+ revisions 349450 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 349451 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-31 15:48 +0000 [r349409-349410] Russell Bryant <russell@russellbryant.com>
+
+ * channels/chan_sip.c: Fix some minor formatting issues based on
+ coding guidelines.
+
+ * channels/sip/include/dialog.h, channels/chan_sip.c,
+ include/asterisk/astobj2.h, main/astobj2.c: Constify tag argument
+ in REF_DEBUG related code.
+
+2011-12-29 15:16 +0000 [r349341] Matthew Jordan <mjordan@digium.com>
+
+ * main/rtp_engine.c, /: Handle AST_CONTROL_UPDATE_RTP_PEER frames
+ in local bridge loop Failing to handle
+ AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop
+ causes the loop to exit prematurely. This causes a variety of
+ negative side effects, depending on when the loop exits. This
+ patch handles the frame by essentially swallowing the frame in
+ the local loop, as the current channel drivers expect the RTP
+ bridge to handle the frame, and, in the case of the local bridge
+ loop, no additional action is necessary. (issue ASTERISK-19040)
+ (issue ASTERISK-19128) (issue ASTERISK-17725) (issue
+ ASTERISK-18340) (closes issue ASTERISK-19095) Reported by: Stefan
+ Schmidt Tested by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1640/ ........ Merged
+ revisions 349339 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 349340 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-28 21:39 +0000 [r349291] Sean Bright <sean@malleable.com>
+
+ * /, main/audiohook.c: Use ast_audiohook_write_list_empty to
+ determine if our lists are empty instead of duplicating that
+ logic. ........ Merged revisions 349289 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 349290 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-28 19:00 +0000 [r349249-349251] Kevin P. Fleming <kpfleming@digium.com>
+
+ * utils, /: Tell Subversion to gnore the 'astdb2bdb' binary file if
+ it exists. ........ Merged revisions 349250 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, res/res_fax.c, include/asterisk/dsp.h,
+ include/asterisk/res_fax.h, res/res_fax_spandsp.c, main/dsp.c:
+ Improve T.38 gateway V.21 preamble detection. This commit removes
+ the V.21 preamble detection code previously added to the generic
+ DSP implementation in Asterisk, and instead enhances the res_fax
+ module to be able to utilize V.21 preamble detection
+ functionality made available by FAX technology modules. This
+ commit also adds such support to res_fax_spandsp, which uses the
+ Spandsp modem tone detection code to do the V.21 preamble
+ detection. There should be no functional change here, other than
+ much more reliable V.21 preamble detection (and thus T.38 gateway
+ initiation). ........ Merged revisions 349248 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-27 20:55 +0000 [r349196] Matthew Jordan <mjordan@digium.com>
+
+ * /, res/res_timing_pthread.c, include/asterisk/module.h,
+ res/res_timing_dahdi.c, res/res_timing_timerfd.c,
+ res/res_musiconhold.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/ ........ Merged
+ revisions 349194 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 349195 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-27 17:17 +0000 [r349146] Sean Bright <sean@malleable.com>
+
+ * /, main/audiohook.c: Once an audiohook is attached to a channel,
+ we continue to transcode all of the frames, even after all of the
+ hooks are detached. This patch short-cicuits us out before we
+ transcode unnecessarily. ........ Merged revisions 349144 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 349145 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-23 21:19 +0000 [r349106] Matthew Jordan <mjordan@digium.com>
+
+ * contrib/realtime/mysql/voicemail.sql,
+ configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
+ Allow overriding of IMAP server settings on a user by user basis
+ This patch allows the imapserver, imapport, and imapflags
+ settings to be overridden for any voicemail user. It also
+ documents the settings in the sample voicemail.conf file, and
+ updates the voicemail schema to allow storage of those columns.
+ (closes issue ASTERISK-16489) Reporter: Hubert Mickael Tested by:
+ Matt Jordan Review: https://reviewboard.asterisk.org/r/1614/
+
+2011-12-23 20:42 +0000 [r349097-349098] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c, main/features.c, configs/sip.conf.sample,
+ channels/sip/include/sip.h: INFO/Record request configurable to
+ use dynamic features Adds two new options to SIP peers allowing
+ them to specify features (dynamic or builtin) to use when sending
+ INFO/record requests. Recordonfeature activates whatever feature
+ is specified when recieving a record: on request while
+ recordofffeature activates whatever feature is specified when
+ receiving a record: off request. Both of these features can be
+ disabled by setting the feature to an empty string. (closes issue
+ ASTERISK-16507) Reported by: Jon Bright Review:
+ https://reviewboard.asterisk.org/r/1634/
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
+ channels/sip/include/sip.h: chan_sip autocreatepeer=persist
+ option for auto-created peers to survive reload This patch moves
+ destruction of sip peers to immediately after the general section
+ of sip.conf is read so that autocreatepeer setting can be read
+ before deletion of peers. If autocreatepeer=persist at reload,
+ then peers created by the autocreatepeer setting will be skipped
+ when purging the current SIP peer list. (closes ASTERISK-16508)
+ Reported by: Kirill Katsnelson Patches:
+ 017797-kkm-persist-autopeers-1.8.patch uploaded by Kirill
+ Katsnelson (license 5845)
+
+2011-12-23 17:36 +0000 [r349046] Sean Bright <sean@malleable.com>
+
+ * /, apps/app_chanspy.c: Merged revisions 349045 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r349045 | seanbright | 2011-12-23 12:32:33 -0500
+ (Fri, 23 Dec 2011) | 25 lines Merged revisions 349044 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r349044 | seanbright | 2011-12-23 12:25:01 -0500 (Fri, 23 Dec
+ 2011) | 18 lines 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 15:26 +0000 [r348994] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_dial.c, /: Fix missing doc tags found while fixing
+ ASTERISK-18689 Add missing <variable></variable> tags in app_dial
+ documentation. ........ Merged revisions 348992 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 348993 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-23 02:35 +0000 [r348953] Richard Mudgett <rmudgett@digium.com>
+
+ * main/pbx.c, /, channels/chan_sip.c, include/asterisk/pbx.h: 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/ ........ Merged
+ revisions 348940 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 348952 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-22 22:39 +0000 [r348890] Matthew Jordan <mjordan@digium.com>
+
+ * cel/cel_pgsql.c, /: Fix for memory leaks / cleanup in cel_pgsql
+ There were a number of issues in cel_pgsql's pgsql_log method: *
+ If either sql or sql2 could not be allocated, the method would
+ return while the pgsql_lock was still locked * If the execution
+ of the log statement succeeded, the sql and sql2 structs were
+ never free'd * Reconnection successes were logged as ERRORs. In
+ general, the severity of several logging statements was reduced
+ (closes issue ASTERISK-18879) Reported by: Niolas Bouliane Tested
+ by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1624/
+ ........ Merged revisions 348888 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 348889 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-22 21:12 +0000 [r348849] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Fix segfault on answer. Only
+ update/change RTP source if RTP has already been started and
+ connected to the subchannel.
+
+2011-12-22 20:44 +0000 [r348848] Matthew Jordan <mjordan@digium.com>
+
+ * /, main/say.c, main/file.c, main/app.c, apps/app_confbridge.c,
+ main/bridging.c: Add Asterisk TestSuite event hooks to support
+ ConfBridge testing This patch adds initial testsuite event hooks
+ so that ConfBridge tests can be executed in the Asterisk
+ TestSuite. (issue ASTERISK-19059) ........ Merged revisions
+ 348846 from http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-22 20:39 +0000 [r348847] Terry Wilson <twilson@digium.com>
+
+ * /, include/asterisk/format_pref.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/ ........ Merged
+ revisions 348833 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 348845 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-21 20:13 +0000 [r348737-348794] Richard Mudgett <rmudgett@digium.com>
+
+ * /, codecs/speex: Make codecs/speex ignore *.i files also.
+ ........ Merged revisions 348793 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, apps/confbridge: Make apps/confbridge ignore *.i files also.
+ ........ Merged revisions 348790 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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 ........ Merged revisions 348735 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 348736 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-20 20:06 +0000 [r348698] Matthew Nicholson <mnicholson@digium.com>
+
+ * contrib/scripts/safe_asterisk: This adds support for setting
+ several safe_asterisk parameters using environment variables and
+ also enables a custom run directory for asterisk (instead of
+ defaulting to /tmp). Patch by: Byron Clark (byronclark) (closes
+ ASTERISK-17810)
+
+2011-12-19 21:43 +0000 [r348649] Richard Mudgett <rmudgett@digium.com>
+
+ * /, configure, configure.ac: Fix crashes on other platforms caused
+ by interference from Darwin weak symbol support. Support weak
+ symbols on a platform specific basis. The Mac OS X (Darwin)
+ support must be isolated from the other platforms because it has
+ caused other platforms to crash. Several other platforms
+ including Linux have GCC versions that define the weak attribute.
+ However, this attribute is only setup for use in the code by
+ Darwin. (closes issue ASTERISK-18728) Reported by: Ben Klang
+ Review: https://reviewboard.asterisk.org/r/1617/ ........ Merged
+ revisions 348647 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 348648 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-19 19:55 +0000 [r348606] Leif Madsen <leif@leifmadsen.com>
+
+ * /, main/message.c: Update documentation for MESSAGE_SEND_STATUS
+ variable. (Closes issue ASTERISK-19056) Reported by: Yuri
+ Patches: 348360.diff uploaded by Yuri (license #5242) ........
+ Merged revisions 348605 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-19 01:36 +0000 [r348567] Terry Wilson <twilson@digium.com>
+
+ * /, res/res_srtp.c: Add a separate buffer for SRTCP packets The
+ function ast_srtp_protect used a common buffer for both SRTP and
+ SRTCP packets. Since this function can be called from multiple
+ threads for the same SRTP session (scheduler for SRTCP and
+ channel for SRTP) it was possible for the packets to become
+ corrupted as the buffer was used by both threads simultaneously.
+ This patch adds a separate buffer for SRTCP packets to avoid the
+ problem. (closes issue ASTERISK-18889, Reported/patch by Daniel
+ Collins) ........ Merged revisions 347995 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 347996 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-18 18:29 +0000 [r348518] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, configs/sip.conf.sample: Correct two flaws in sip.conf.sample
+ related to AST-2011-013. * The sample file listed *two* values
+ for the 'nat' option as being the default. Only 'force_rport' is
+ the default. * The warning about having differing 'nat' settings
+ confusingly referred to both peers and users. ........ Merged
+ revisions 348515 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
+ Merged revisions 348516 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 348517 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-16 23:58 +0000 [r348466] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /, main/features.c: Clean-up on isle five for
+ __ast_request_and_dial() and ast_call_forward(). * Add locking
+ when a channel inherits variables and datastores in
+ __ast_request_and_dial() and ast_call_forward(). Note: The
+ involved channels are not active so there was minimal potential
+ for problems. * Remove calls to ast_set_callerid() in
+ __ast_request_and_dial() and ast_call_forward() because the set
+ information is for the wrong direction. * Don't use C++ keywords
+ for variable names in ast_call_forward(). * Run the redirecting
+ interception macro if defined when forwarding a call in
+ ast_call_forward(). Note: Currently will never execute because
+ the only callers that supply a calling channel supply a hungup or
+ zombie channel. * Make feature_request_and_dial() put the
+ transferee into autoservice when it calls ast_call_forward() in
+ case a redirection interception macro is run. Note: Currently
+ will never happen because the caller channel (Party B) is always
+ hungup at this time. * Make feature_request_and_dial() ignore the
+ AST_CONTROL_PROCEEDING frame to silence a log message. ........
+ Merged revisions 348464 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 348465 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-16 22:00 +0000 [r348416] Jonathan Rose <jrose@digium.com>
+
+ * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
+ Voicemail with the saycid option will now play a caller's name
+ based on cid if available. In order to check the availability of
+ the caller's name, app_voicemail will check for an audio file in
+ <astspooldir>/recordings/callerids/ This change sets a precedent
+ for where to put recordings of names. Currently the idea is that
+ recordings here could also be used for applications like
+ confbridge and meetme to find recorded names in this folder from
+ callerid (when another recording isn't available) (closes issue
+ ASTERISK-18565) Reporter: Russell Brown Patches: r uploaded by
+ Russel Brown (license 6182)
+
+2011-12-16 21:30 +0000 [r348312-348408] Richard Mudgett <rmudgett@digium.com>
+
+ * main/channel.c, /: Fix cut and past error in ast_call_forward().
+ (issue ASTERISK-18836) ........ Merged revisions 348401 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 348405 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/channel.c, main/pbx.c, /, apps/app_authenticate.c,
+ funcs/func_cdr.c, main/features.c, include/asterisk/cdr.h,
+ apps/app_followme.c, apps/app_queue.c, res/res_monitor.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/ ........ Merged
+ revisions 348362 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 348363 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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. ........ Merged
+ revisions 348310 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 348311 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-14 22:36 +0000 [r348215-348266] Matthew Jordan <mjordan@digium.com>
+
+ * /, apps/app_originate.c: Added support for all slin formats to
+ app_originate Previously, app_originate could not originate a
+ call into a non-8kHz conference bridge as the formats for
+ non-8kHz slin codecs were not applied to the created channel.
+ This patch adds all of the formats by default, such that if a
+ created channel has a codec that supports a higher sampling rate,
+ a translation path can be built between it and other channels.
+ ........ Merged revisions 348265 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, apps/app_queue.c: Fixed Asterisk crash when function
+ QUEUE_MEMBER receives invalid input The function QUEUE_MEMBER has
+ two required parameters (queuename, option). It was only checking
+ for the presence of queuename. The patch checks for the existence
+ of the option parameter and provides better error logging when
+ invalid values are provided for the option parameter as well.
+ ........ Merged revisions 348211 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-14 22:05 +0000 [r348214] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, res/res_fax.c: Don't clear LOCALSTATIONID before sending or
+ receiving. The user may set that variable. ASTERISK-18921
+ ........ Merged revisions 348212 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 348213 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-14 21:08 +0000 [r348161] Jonathan Rose <jrose@digium.com>
+
+ * main/features.c, configs/features.conf.sample: Add and document
+ PARKEDCALL variable set during timeout PARKEDCALL variable tracks
+ which parking lot the call was last parked in. This can be used
+ afterwards for flow control when returntoorigin is set to off. I
+ went ahead and documented both this and the existing variable set
+ during timeout (PARKINGSLOT) in the sample features.conf since
+ there was no prior mention of variables being set during timeout.
+ (closes issue ASTERISK-16239) Reported By: Clod Patry Patches:
+ M17503.diff uploaded by Clod Patry (license 5138)
+
+2011-12-14 20:51 +0000 [r348160] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_confbridge.c: Improve error message in CONFBRIDGE_INFO
+ Provided a more descriptive error message when a value supplied
+ for the parameter type is not one of the acceptable values.
+ (closes issue ASTERISK-18717) Reported by: Paul Belanger Patches:
+ __20111103-better-confbridge_info-error-msg.txt (License #4999)
+
+2011-12-14 20:37 +0000 [r348156-348159] Jonathan Rose <jrose@digium.com>
+
+ * /, configs/features.conf.sample: Fix accidental use of tabs
+ instead of spaces from previous features.conf.sample change
+ ........ Merged revisions 348157 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 348158 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, configs/features.conf.sample: Document PARKINGSLOT variable in
+ features.conf.sample (issue ASTERISK-16239) ........ Merged
+ revisions 348154 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 348155 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-13 23:10 +0000 [r348103] Richard Mudgett <rmudgett@digium.com>
+
+ * /, bridges/bridge_builtin_features.c, apps/app_followme.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/ ........ Merged
+ revisions 348101 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 348102 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-13 15:22 +0000 [r348061] Stefan Schmidt <sst@sil.at>
+
+ * channels/chan_sip.c: Fix possible misshandling of an incoming SIP
+ response as a peer poke response. Also make sure peer has even
+ qualify enabled when handle a peer poke response. (closes issue
+ ASTERISK-18940) Reported by: Vitaliy Tested by: Vitaliy and
+ UnixDev Review: https://reviewboard.asterisk.org/r/1620 Reviewed
+ by: David Vossel ........ Merged revisions 348048 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 348056 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-12 19:35 +0000 [r347997] Matthew Jordan <mjordan@digium.com>
+
+ * include/asterisk/logger.h, utils/refcounter.c, main/logger.c,
+ utils/hashtest.c, UPGRADE.txt, utils/ael_main.c,
+ utils/hashtest2.c, CHANGES, main/asterisk.c, main/config.c,
+ configs/logger.conf.sample, main/loader.c, main/cli.c: Backed out
+ core changes from r346391 During testing, it was discovered that
+ there were a number of side effects introduced by r346391 and
+ subsequent check-ins related to it (r346429, r346617, and
+ r346655). This included the /main/stdtime/ test 'hanging', as
+ well as the remote console option failing to receive the
+ appropriate output after a period of time. I only backed out the
+ changes to main/ and utils/, as this was adequate to reverse the
+ behavior experienced. (issue ASTERISK-18974)
+
+2011-12-12 17:34 +0000 [r347954] Richard Mudgett <rmudgett@digium.com>
+
+ * configs/iax.conf.sample, configs/chan_dahdi.conf.sample, /,
+ configs/chan_ooh323.conf.sample, configs/vpb.conf.sample,
+ configs/extensions.lua.sample, configs/sip.conf.sample,
+ configs/extensions.conf.sample: Update sample configs to put
+ incoming calls into context public. * Add warning about the SIP
+ allowguest option in context public. (closes issue
+ ASTERISK-14122) Reported by: Alec Davis Review:
+ https://reviewboard.asterisk.org/r/719/ ........ Merged revisions
+ 347953 from http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-09 21:47 +0000 [r347866-347903] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_mixmonitor.c: Adds MixMonitor and StopMixMonitor AMI
+ commands to the manager These commands work much like the
+ dialplan applications that would otherwise invoke them. A nice
+ benefit of these is that they can be invoked on a call remotely
+ and at any time during a call. They work much like the Monitor
+ and StopMonitor ami commands. (closes issue ASTERISK-17726)
+ Reported by: Sergio González Martín Patches:
+ mixmonitor_actions.diff uploaded by Sergio González Martín
+ (license 5644) Review: https://reviewboard.asterisk.org/r/1193/
+
+ * include/asterisk/file.h, apps/app_sayunixtime.c, CHANGES: Remove
+ autojump extensions from SayUnixTime, make an option to perform
+ automatic jumps. When a caller sends DTMF while the SayUnixTime
+ application is saying the time, The call would jump to the next
+ extension much like it does during Background(). This patch adds
+ option 'j' to SayUnixTime which when used employs the old
+ behavior. Also, this patch allows arguments to sayunixtime to not
+ be used as empty strings in the case of something like
+ 'sayunixtime(,,,j)' or 'sayunixtime(,,pattern). (closes issue
+ ASTERISK-16675) Reported by: jlpedrosa Patches:
+ patch_SayUnixTime_noJump.patch uploaded by jlpedrosa (license
+ 5959) Review: https://reviewboard.asterisk.org/r/956/
+
+2011-12-09 01:33 +0000 [r347813] Richard Mudgett <rmudgett@digium.com>
+
+ * main/pbx.c, /: Fix some parsing issues in
+ add_exten_to_pattern_tree(). * Simplify compare_char() and avoid
+ potential sign extension issue. * Fix infinite loop in
+ add_exten_to_pattern_tree() handling of character set escape
+ handling. * Added buffer overflow checks in
+ add_exten_to_pattern_tree() character set collection. * Made
+ ignore empty character sets. * Added escape character handling to
+ end-of-range character in character sets. This has a slight
+ change in behavior if the end-of-range character is an escape
+ character. You must now escape it. * Fix potential sign extension
+ issue when expanding character set ranges. * Made remove
+ duplicated characters from character sets. The duplicate
+ characters lower extension matching priority and prevent
+ duplicate extension detection. * Fix escape character handling
+ when the escape character is trying to escape the end-of-string.
+ We could have continued processing characters after the end of
+ the exten string. We could have added the previous character to
+ the pattern matching tree incorrectly. (closes issue
+ ASTERISK-18909) Reported by: Luke-Jr ........ Merged revisions
+ 347811 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 347812 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-08 21:32 +0000 [r347735] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, channels/chan_sip.c: Fix regression when using tcpenable=no
+ and tlsenable=yes. The tlsenable settings are tucked away in
+ main/tcptls.c, so I missed them when resolving ASTERISK-18837.
+ This should resolve the test suite breakage of the sip tls tests.
+ Review: https://reviewboard.asterisk.org/r/1615 Reviewed by: Matt
+ Jordan ........ Merged revisions 347718 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 347727 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-08 20:55 +0000 [r347658] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_queue.c: Fix regressed behavior of queue set penalty
+ to work without specifying 'in <queuename>' r325483 caused a
+ regression in Asterisk 10+ that would make Asterisk segfault when
+ attempting to set penalty on an interface without specifying a
+ queue in the queue set penalty CLI command. In addition, no
+ attempt would be made whatsoever to perform the penalty setting
+ on all the queues in the core list with either the cli command or
+ the non-segfaulting ami equivalent. This patch fixes that and
+ also makes an attempt to document and rename some functions
+ required by this command to better represent what they actually
+ do. Oh yeah, and the use of this command without specifying a
+ specific queue actually works now. Review:
+ https://reviewboard.asterisk.org/r/1609/ ........ Merged
+ revisions 347656 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-08 17:55 +0000 [r347601] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Mark channel running the h exten with the
+ soft-hangup flag. When a bridge is broken, ast_bridge_call()
+ might execute the h exten on the calling channel. However, that
+ channel may not have been the channel that broke the bridge by
+ hanging up. The channel executing the h exten must be in a hung
+ up state so things like AGI run in the correct mode. * Make sure
+ ast_bridge_call() marks the channel it is executing the h exten
+ on as hung up. (The AST_SOFTHANGUP_APPUNLOAD flag is used so as
+ to match the pbx.c main dialplan execution loop when it executes
+ the h exten.) (closes issue ASTERISK-18811) Reported by: David
+ Hajek Patches: jira_asterisk_18811_v1.8.patch (license #5621)
+ patch uploaded by rmudgett Tested by: David Hajek, rmudgett
+ ........ Merged revisions 347595 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 347600 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-08 16:24 +0000 [r347533] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Don't crash on INFO automon request with
+ no channel AST-2011-014. When automon was enabled in
+ features.conf, it was possible to crash Asterisk by sending an
+ INFO request if no channel had been created yet. (closes issue
+ ASTERISK-18805) ........ Merged revisions 347530 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
+ Merged revisions 347531 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 347532 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-08 06:59 +0000 [r347490] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Fix segfault on answer. Fix a segfault if
+ an attempt to answer a call is made between when the inbound call
+ gives up (and the channel is removed) and when the device is
+ notified and removes the call from the device.
+
+2011-12-07 21:42 +0000 [r347440] Richard Mudgett <rmudgett@digium.com>
+
+ * main/manager.c, /: Update AMI Getvar and Setvar documentation
+ about supplying a channel name. (closes issue ASTERISK-18958)
+ Reported by: Red Patches: jira_asterisk_18958_v1.8.patch (license
+ #5621) patch uploaded by rmudgett ........ Merged revisions
+ 347438 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 347439 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-07 20:34 +0000 [r347395] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_meetme.c: Fix: Meetme recording variables from
+ realtime DB use null entries over channel variables Meetme would
+ attempt to substitute the realtime values of RECORDING_FILE and
+ RECORDING_FORMAT from the meetme db entry instead of using the
+ channel variable set for those variables in spite of those
+ database entries being NULL or even lacking a column to represent
+ them. (closes issue ASTERISK-18873) Reported by: Byron Clark
+ Patches: ASTERISK-18873-1.patch uploaded by Byron Clark (license
+ 6157) ........ Merged revisions 347369 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 347383 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-07 20:15 +0000 [r347345] Terry Wilson <twilson@digium.com>
+
+ * Makefile, include/asterisk/paths.h, /,
+ configs/asterisk.conf.sample, build_tools/make_defaults_h,
+ main/asterisk.c, main/db.c: Add ASTSBINDIR to the list of
+ configurable paths This patch also makes astdb2sqlite3 and
+ astcanary use the configured directory instead of relying on
+ $PATH. (closes issue ASTERISK-18959) Review:
+ https://reviewboard.asterisk.org/r/1613/ ........ Merged
+ revisions 347344 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-06 23:58 +0000 [r347294] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Make SIP INFO messages for dtmf-relay
+ signals case insensitive. (closes issue ASTERISK-18924) Reported
+ by: Kevin Taylor ........ Merged revisions 347292 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 347293 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-06 22:01 +0000 [r347241] Jonathan Rose <jrose@digium.com>
+
+ * main/pbx.c, /: Documents CHANNEL(musicclass) taking priority over
+ m([x]) in waitExten If waitExten specifies a music class to use
+ with its music on hold option, it will use CHANNEL(musicclass)
+ instead if that channel variable has been set on the initiating
+ channel. This documents that behavior in the waitExten app so
+ that this can be known without checking the documentation of the
+ code in function local_ast_moh_start. (closes issue
+ ASTERISK-18804) ........ Merged revisions 347239 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 347240 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-06 20:23 +0000 [r347157-347192] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * UPGRADE.txt, CHANGES, apps/app_voicemail.c: Add VM_INFO()
+ dialplan function to gather information about a mailbox.
+ Deprecates MAILBOX_EXISTS. Provides count, email, exists,
+ fullname, language, locale, pager, password, tz. (closes issue
+ ASTERISK-18634) Patch by: Kris Shaw Review:
+ https://reviewboard.asterisk.org/r/1568 Reviewed by: 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
+ ........ Merged revisions 347166 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 347167 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, apps/app_voicemail.c: Add regression tests for issue
+ ASTERISK-18838. Review: https://reviewboard.asterisk.org/r/1572
+ Reviewed by: Matt Jordan ........ Merged revisions 347131 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 347146 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, apps/app_voicemail.c: 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 ........ Merged revisions 347111 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 347124 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-06 19:09 +0000 [r347110] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/dlinkedlists.h, tests/test_linkedlists.c: Doubly
+ linked lists unit test and update to implementation. Update the
+ doubly linked list implementation. Now safe traversing can insert
+ before and after the current node when traversing in either
+ direction. Updated the linked lists unit test test_linkedlist to
+ also test doubly linked lists. The old test_dlinkedlist requires
+ a manual check of results and probably should be removed. Review:
+ https://reviewboard.asterisk.org/r/1569/
+
+2011-12-06 17:34 +0000 [r347069] Matthew Jordan <mjordan@digium.com>
+
+ * /, 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/ ........ Merged
+ revisions 347058 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 347068 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-05 17:44 +0000 [r347008] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/sig_analog.h: Restore call progress code for analog
+ ports. Extracting sig_analog from chan_dahdi lost call progress
+ detection functionality. * Fix analog ports from considering a
+ call answered immediately after dialing has completed if the
+ callprogress option is enabled. (closes issue ASTERISK-18841)
+ Reported by: Richard Miller Patches: chan_dahdi.diff (license
+ #5685) patch uploaded by Richard Miller (Modified by me)
+ sig_analog.c.diff (license #5685) patch uploaded by Richard
+ Miller (Modified by me) sig_analog.h.diff (license #5685) patch
+ uploaded by Richard Miller ........ Merged revisions 347006 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 347007 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-05 15:04 +0000 [r346956] Jonathan Rose <jrose@digium.com>
+
+ * main/pbx.c, /: Resolve duplicate label used in multiple
+ priorities for the same extension. Prior to this patch, if labels
+ with the same name were used for different priorities in the same
+ extension, the new label would be accepted, but it would be
+ unusable since attempts to reach that label would just go to the
+ first one. Now pbx.c detects this, generates a warning in logs,
+ and culls the label before adding it to the dialplan. (closes
+ issue ASTERISK-18807) Reported by: Kenneth Shumard Patches:
+ pbx.c.patch uploaded by Kenneth Shumard (License 5077) ........
+ Merged revisions 346954 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 346955 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-05 14:47 +0000 [r346953] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_jabber.exports.in, /: Fix chan_jingle/gtalk load
+ regression introduced in r346087 Add missing symbol exports for
+ ast_aji_client_destroy and ast_aji_buddy_destroy for usage
+ outside res_jabber. Testing of these changes focused on
+ res_jabber itself, so this problem was missed. Reported-by:
+ Michael Spiceland ........ Merged revisions 346951 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 346952 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-04 10:08 +0000 [r346901] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, channels/chan_sip.c: For SIP REGISTER fix domain-only URIs and
+ domain ACL bypass. The code that allowed admins to create users
+ with domain-only uri's had stopped to work in 1.8 because of the
+ reqresp parser rewrites. This is fixed now: if you have a
+ [mydomain.com] sip user, you can register with useraddr
+ sip:mydomain.com. Note that in that case -- if you're using
+ domain ACLs (a configured domain list) -- mydomain.com must be in
+ the allow list as well. Reviewboard r1606 shows a list of
+ registration combinations and which SIP response codes are
+ returned. Review: https://reviewboard.asterisk.org/r/1533/
+ Reviewed by: Terry Wilson (closes issue ASTERISK-18389) (closes
+ issue ASTERISK-18741) ........ Merged revisions 346899 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 346900 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-02 23:30 +0000 [r346857] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Update SIP MESSAGE To parsing to
+ correctly handle URI The previous patch (r346040) incorrectly
+ parsed the URI in the presence of a port, e.g.,
+ user@hostname:port would fail as the port would be double
+ appended to the SIP message. This patch uses the parse_uri
+ function to correctly parse the URI into its username and
+ hostname parts, and places them in the correct fields in the
+ sip_pvt structure. (issue ASTERISK-18903) Review:
+ https://reviewboard.asterisk.org/r/1597/ ........ Merged
+ revisions 346856 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-02 19:40 +0000 [r346777-346816] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c: implement nat option for rtp channels with
+ ooh323
+
+ * addons/chan_ooh323.c, /, channels/chan_h323.c: Merged revisions
+ 346763 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r346763 | may | 2011-12-02 20:42:32 +0400 (Fri,
+ 02 Dec 2011) | 14 lines Merged revisions 346762 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r346762 | may | 2011-12-02 20:19:19 +0400 (Fri, 02 Dec 2011) | 7
+ lines 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 21:19 +0000 [r346709] Richard Mudgett <rmudgett@digium.com>
+
+ * main/stun.c, /, res/res_stun_monitor.c,
+ configs/res_stun_monitor.conf.sample, include/asterisk/stun.h:
+ 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/
+ ........ Merged revisions 346700 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 346701 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-12-01 20:46 +0000 [r346699] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: Change 183 Ringing in sipfrag body to 180
+ ringing. 183 Ringing isn't even a thing. 183 is actually a
+ session progress message. (closes issue ASTERISK-18925) Reported
+ by: Sebastian Denz Tested by: jrose Patches:
+ asterisk18-use_180_instead_of_183_in_sipfrag.diff by Sebastian
+ Denz (License #6139) ........ Merged revisions 346697 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 346698 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-30 23:38 +0000 [r346617-346655] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * channels/chan_unistim.c, main/tcptls.c, channels/chan_sip.c,
+ main/config.c, main/loader.c: Remove the few places where we try
+ to ast_verbose() without a newline.
+
+ * main/asterisk.c: Fix edge case for overflow buffer.
+
+2011-11-30 22:03 +0000 [r346525-346566] Jonathan Rose <jrose@digium.com>
+
+ * main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h:
+ 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/ ........ Merged
+ revisions 346564 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 346565 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/tcptls.c, channels/chan_sip.c, include/asterisk/tcptls.h:
+ Reverting 346525 due to accidental patch against trunk instead of
+ 1.8
+
+ * main/tcptls.c, channels/chan_sip.c, include/asterisk/tcptls.h:
+ 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 19:37 +0000 [r346474] Leif Madsen <leif@leifmadsen.com>
+
+ * configs/queues.conf.sample: Update queues.conf.sample
+ documentation. Update the documentation surrounding the use of
+ MONITOR_EXEC to make it more clear that it can be used for both
+ Monitor() and MixMonitor() usage. (closes issue ASTERISK-17413)
+ Reported by: David Woolley Patches:
+ issue18817_mixmonitor_queues_doc.diff by Michael L. Young
+ (License #5026) ........ Merged revisions 346472 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 346473 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-29 20:32 +0000 [r346391-346429] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * utils/refcounter.c, utils/hashtest.c, utils/ael_main.c,
+ utils/hashtest2.c: Fix compilation of utilities (caught by
+ Bamboo).
+
+ * addons/chan_ooh323.c, channels/chan_sip.c, main/say.c,
+ res/res_fax.c, UPGRADE.txt, res/res_musiconhold.c,
+ res/res_jabber.c, CHANGES, configs/logger.conf.sample,
+ main/cli.c, channels/chan_usbradio.c, include/asterisk/logger.h,
+ main/dial.c, channels/chan_skinny.c, main/logger.c,
+ codecs/codec_dahdi.c, apps/app_rpt.c, apps/app_verbose.c,
+ main/asterisk.c, main/bridging.c, res/res_clialiases.c,
+ addons/res_config_mysql.c, apps/app_voicemail.c: Allow each
+ logging destination and console to have its own notion of the
+ verbosity level. Review: https://reviewboard.asterisk.org/r/1599
+
+2011-11-29 00:03 +0000 [r346350] David Vossel <dvossel@digium.com>
+
+ * /, include/asterisk/message.h, main/message.c: Merged revisions
+ 346349 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r346349 | dvossel | 2011-11-28 18:00:11 -0600 (Mon, 28 Nov 2011)
+ | 10 lines Fixes memory leak in message API. The ast_msg_get_var
+ function did not properly decrement the ref count of the var it
+ retrieves. The way this is implemented is a bit tricky, as we
+ must decrement the var and then return the var's value. As long
+ as the documentation for the function is followed, this will not
+ result in a dangling pointer as the ast_msg structure owns its
+ own reference to the var while it exists in the var container.
+ ........
+
+2011-11-28 14:34 +0000 [r346294] Stefan Schmidt <sst@sil.at>
+
+ * res/res_rtp_asterisk.c, /: Fix regression that 'rtp/rtcp set
+ debup ip' only works when also a port was specified. (closes
+ issue ASTERISK-18693) Reported by: Davide Dal Fra Review:
+ https://reviewboard.asterisk.org/r/1600/ Reviewed by: Walter
+ Doekes ........ Merged revisions 346292 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 346293 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-23 23:03 +0000 [r346241] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/acl.h, /, channels/chan_skinny.c,
+ channels/chan_h323.c, main/acl.c, channels/chan_iax2.c: Fix calls
+ to ast_get_ip() not initializing the address family. ........
+ Merged revisions 346239 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 346240 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-23 20:48 +0000 [r346146-346199] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, channels/chan_sip.c: Minor cleanup in chan_sip get_msg_text()
+ function. In r116240, get_msg_text() got an extra parameter to
+ fix the unwanted addition of trailing newlines to SIP MESSAGE
+ bodies. This caused all linefeeds to be trimmed, which isn't
+ right either. This is a stop-gap; the right fix is to return the
+ original SIP request body. Review:
+ https://reviewboard.asterisk.org/r/1586 Reviewed by: Matt Jordan
+ ........ Merged revisions 346147 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 346198 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, include/asterisk/strings.h: Fix ast_str_truncate signedness
+ warning and documentation. Review:
+ https://reviewboard.asterisk.org/r/1594 ........ Merged revisions
+ 346144 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 346145 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-23 17:16 +0000 [r346088] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_jingle.c, /, include/asterisk/jabber.h,
+ channels/chan_gtalk.c, res/res_jabber.c: 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 ........ Merged revisions
+ 346086 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 346087 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-23 16:23 +0000 [r346053] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Fixed SendMessage stripping extension
+ from To: header in SIP MESSAGE When using the MessageSend
+ application to send a SIP MESSAGE to a non-peer, chan_sip
+ attempted to validate the hostname or IP Address. In the process,
+ it stripped off the extension and failed to add it back to the
+ sip_pvt structure before transmitting. This patch adds the full
+ URI passed in from the message core to the sip_pvt structure.
+ (closes issue ASTERISK-18903) Reported by: Shaun Clark Tested by:
+ Matt Jordan Review: https://reviewboard.asterisk.org/r/1597/
+ ........ Merged revisions 346040 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-23 16:12 +0000 [r346033] Terry Wilson <twilson@digium.com>
+
+ * /, res/res_musiconhold.c: Resume playing existing hold music for
+ cached realtime MOH As a result of the fix for ASTERISK-18039,
+ realtime caching MOH no longer properly resumes playing back a
+ file between different holds in the same call. This is because
+ scanning for new files causes the existing file array to be
+ emptied and we were just comparing that the saved pointer to the
+ filename matched the pointer to the filename in a particular
+ position in the array. An easy fix is to save the filename
+ instead of a pointer to it and then do a strcmp instead of
+ comparing the addresses. (closes issue ASTERISK-18912) Review:
+ https://reviewboard.asterisk.org/r/1596/ ........ Merged
+ revisions 346030 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 346031 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-23 16:10 +0000 [r346032] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, res/res_format_attr_silk.c, res/res_format_attr_celt.c: Added
+ support level for new modules ........ Merged revisions 346029
+ from http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-22 23:06 +0000 [r345978] Richard Mudgett <rmudgett@digium.com>
+
+ * main/dnsmgr.c, /, include/asterisk/dnsmgr.h: 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.
+ ........ Merged revisions 345976 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 345977 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-22 20:32 +0000 [r345925] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * include/asterisk/logger.h, /: Clarify why the AST_LOG_* macros
+ exist next to the LOG_* macros. (issue ASTERISK-17973) ........
+ Merged revisions 345923 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 345924 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-22 16:41 +0000 [r345883] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, apps/confbridge/conf_config_parser.c: Add missing
+ sound_only_one config variable (closes issue ASTERISK-18895)
+ Reported by: zvision Patches: conf_config_parser.diff (license
+ #5755) patch uploaded by zvision ........ Merged revisions 345882
+ from http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-21 21:09 +0000 [r345831] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: 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 ........
+ Merged revisions 345828 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 345830 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-21 16:40 +0000 [r345735] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * CHANGES, main/config.c: Add #tryinclude statement This provides
+ the same functionality as #include however an asterisk module
+ will still load if the filename does not exist. Review:
+ https://reviewboard.asterisk.org/r/1476/
+
+2011-11-19 15:11 +0000 [r345643-345684] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, main/db.c: Update the documentation to better clarify how the
+ existing commands work. Review:
+ https://reviewboard.asterisk.org/r/1593/ ........ Merged
+ revisions 345682 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 345683 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, main/db.c: Fix a change in behavior in 'database show' from
+ 1.8. In 1.8 and previous versions, one could use any fullword
+ portion of the key name, including the full key, to obtain the
+ record. Until this patch, this did not work for the full key.
+ Closes issue ASTERISK-18886 Patch: by tilghman Review: by twilson
+ (http://pastebin.com/7rtu6bpk) on #asterisk-dev ........ Merged
+ revisions 345640 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-17 19:47 +0000 [r345560-345601] Matthew Jordan <mjordan@digium.com>
+
+ * contrib/realtime/mysql/sipfriends.sql (removed): Accidentally
+ readded sipfriends.sql in r345560. This was removed in r342871
+
+ * configs/confbridge.conf.sample,
+ apps/confbridge/include/confbridge.h, apps/app_confbridge.c,
+ CHANGES, contrib/realtime/mysql/sipfriends.sql (added),
+ apps/confbridge/conf_config_parser.c: Add admin toggle mute all
+ and participant count menu options to app_confbridge This patch
+ adds two new menu features to app_confbridge, admin_toggle_menu_
+ participants and participant_count. The admin action will
+ globally mute / unmute all non-admin participants on a
+ converence, while the participant count simply exposes the
+ existing participant count function to the conference bridge
+ menu. This also adds configuration options to change the sound
+ played when the conference is globally muted / unmuted, as well
+ as the necessary config hooks to place these functions in the
+ DTMF menus. (closes issue ASTERISK-18204) Reported by: Kevin
+ Reeves Tested by: Matt Jordan Patches:
+ app_confbridge.c.patch.txt, conf_config_parser.c.patch.txt,
+ confbridge.h.patch.txt uploaded by Kevin Reeves (license 6281)
+ Review: https://reviewboard.asterisk.org/r/1518/
+
+2011-11-17 17:31 +0000 [r345559] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Remove dead code since pri_grab() can
+ never fail. Dead code makes programmers sick. I am sick of
+ looking at it. ........ Merged revisions 345546 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 345558 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-16 14:56 +0000 [r345489] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_voicemail.c: Guarantee messages go into the right
+ folders with multiple recipients Before, using the U flag in
+ Voicemail with multiple recipients would put urgent messages in
+ the INBOX folder for all users past the first thanks to a bug
+ with the message copying function. This would also cause messages
+ to fail to be sent if the INBOX directory hadn't been created for
+ that mailbox yet. (closes issue ASTERISK-18245) Reported by: Matt
+ Jordan (closes issue ASTERISK-18246) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/1589/ ........ Merged
+ revisions 345487 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 345488 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-15 20:11 +0000 [r345221-345433] Richard Mudgett <rmudgett@digium.com>
+
+ * /, res/res_agi.c: Make FastAGI HANGUP show up in AGI debug
+ output. * Change from using send() to ast_agi_send() so the
+ HANGUP shows up in the AGI debug output. (closes issue
+ ASTERISK-18723) Reported by: James Van Vleet Patches:
+ jira_asterisk_18723_v1.8.patch (license #5621) patch uploaded by
+ rmudgett ........ Merged revisions 345431 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 345432 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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 ........ Merged revisions 345370 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 345371 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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/ ........ Merged
+ revisions 345285 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 345290 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample, UPGRADE-1.8.txt,
+ channels/sip/include/sip.h: 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/ ........ Merged
+ revisions 345273 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 345275 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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
+ ........ Merged revisions 345219 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 345220 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-14 19:12 +0000 [r345165] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, /: Don't read past end of input when calling
+ write() int blah = 1; ... write(chan->alertpipe[1], &blah,
+ new_frames * sizeof(blah)) != (new_frames * sizeof(blah))) is
+ only valid when new_frames == 1. Otherwise we start reading into
+ adjacent variables declared on the stack. The read end discards
+ what is read, so the values don't matter but it's not a good idea
+ to read past where we want even though new_frames is almost
+ always 1 and should never be large. This patch is basically taken
+ out of kpfleming's eventfd branch, as he mentioned that he
+ remembered fixing it there when I talked to him about this issue.
+ Review: https://reviewboard.asterisk.org/r/1583/ ........ Merged
+ revisions 345163 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 345164 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-14 19:03 +0000 [r345162] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, channels/sip/include/reqresp_parser.h: Update reqresp_parser
+ parse_uri doxygen comments. The issue mentioned in the bug report
+ had been fixed recently by twilson. The reporter included this
+ documentation fix. (closes issue ASTERISK-18572) Reported by:
+ Richard Miller Patch by: Richard Miller (modified) ........
+ Merged revisions 345160 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 345161 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-14 16:21 +0000 [r345120] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_voicemail.c: Moves voicemail setup password entry to
+ the end of the setup process. This change was made because
+ forcegreeting and forcename settings in voicemail could be
+ circumvented by hanging up after entering a password, because the
+ only way voicemail currently observes whether a mailbox is new or
+ not is by checking to see if the password is the same as the
+ mailbox number or not. (closes issue ASTERISK-18282) Reported by:
+ Matt Jordan Review: https://reviewboard.asterisk.org/r/1581/
+ ........ Merged revisions 345062 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 345117 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-14 15:11 +0000 [r345065] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Ensure that a null vmexten does not cause
+ a segfault When sip_send_mwi_to_peer was modified recently to
+ avoid deadlocks, vmexten was not expected to be null. This change
+ handles that situation to avoid a segfault. ........ Merged
+ revisions 345063 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 345064 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-14 01:25 +0000 [r345023] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c: Increased max number of destinations.
+
+2011-11-12 16:32 +0000 [r344979] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * channels/chan_misdn.c, /: mISDN Round Robin break when no channel
+ is available Prevent channels been parsed repetitively. ........
+ Merged revisions 344965 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 344966 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-12 00:36 +0000 [r344901] Terry Wilson <twilson@digium.com>
+
+ * /, res/res_musiconhold.c: Don't forget to rescan MOH files for
+ cached realtime classes Realtime MOH class caching was
+ implemented because without it, you would build a completely new
+ MOH class and would start the music over at the beginning each
+ time hold was pressed in a conversation. Unfortunately, this
+ broke re-scanning for file changes for realtime MOH classes. This
+ patch corrects that issue. (closes issue ASTERISK-18039) Review:
+ https://reviewboard.asterisk.org/r/1579/ ........ Merged
+ revisions 344899 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 344900 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-11 22:00 +0000 [r344846] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * include/asterisk/utils.h, /, main/utils.c,
+ include/asterisk/stringfields.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
+ ........ Merged revisions 344843 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 344845 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-11 21:57 +0000 [r344844] Matthew Jordan <mjordan@digium.com>
+
+ * /, main/file.c: Video format was treated as audio when removed
+ from the file playback scheduler This patch fixes the format type
+ check in ast_closestream and filestream_destructor. Previously a
+ comparison operator was used, but since audio formats are no
+ longer contiguous (and AST_FORMAT_AUDIO_MASK includes formats
+ that have a value greater than the video formats), a bitwise AND
+ operation is used instead. Duplicated code was also moved to
+ filestream_close. (closes issue ASTERISK-18682) Reported by: Aldo
+ Bedrij Tested by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1580/ ........ Merged
+ revisions 344823 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 344842 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-11 21:37 +0000 [r344838-344840] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, 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. ........
+ Merged revisions 344837 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 344839 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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 ........ Merged revisions
+ 344835 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 344836 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-11 20:15 +0000 [r344771] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Fix regression introduced by SDP fixups
+ If capability is adjusted when switching to UDPTL during fax
+ transmission, fax teardown fails. Make sure capability is only
+ touched if RTP is active. This regression was introduced in
+ R344385. ........ Merged revisions 344769 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 344770 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-11 18:37 +0000 [r344663-344717] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Check sip.conf maxforwards parameter for
+ range 1 <= x <= 255. JIRA AST-710 ........ Merged revisions
+ 344715 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 344716 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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 ........ Merged revisions 344661 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 344662 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-11 15:47 +0000 [r344610] Jonathan Rose <jrose@digium.com>
+
+ * main/pbx.c, /: Fix a segmentation fault when using an extension
+ with CID matching and no CID. Attempting to call an extension
+ which used Caller ID matching with a channel that has an empty
+ caller id string would result in a segmentation fault. (closes
+ issue ASTERISK-18392 Reported By: Ales Zelenik ........ Merged
+ revisions 344608 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 344609 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-10 23:21 +0000 [r344538-344560] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_macro.c: Fix app_macro.c MODULEINFO section
+ termination. (closes issue ASTERISK-18848) Reported by: Tony
+ Mountifield ........ Merged revisions 344557 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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. ........ Merged revisions
+ 344539 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 344540 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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 ........ Merged revisions 344536 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 344537 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-10 21:56 +0000 [r344494] David Vossel <dvossel@digium.com>
+
+ * /, main/bridging.c: Merged revisions 344493 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r344493 | dvossel | 2011-11-10 15:54:42 -0600 (Thu, 10 Nov 2011)
+ | 12 lines Fixes issue with ConfBridge participants hanging up
+ during DTMF feature menu usage getting stuck in conference
+ forever. When a conference user enters the DTMF menu they are
+ suspended from the bridge while the channel is handed off to the
+ DTMF feature code. If a user entered this state and hungup, there
+ existed a race condition where the channel could not exit the
+ conference because it was waiting on a signal that would never
+ arrive. This patch fixes that, because it would stupid for me to
+ talk about the problem and commit a patch for something else.
+ (closes issue ASTERISK-18829) Reported by: zvision ........
+
+2011-11-10 21:15 +0000 [r344387-344441] Kinsey Moore <kmoore@digium.com>
+
+ * /, apps/app_meetme.c: Fix another incorrect case with meetme's
+ PIN logic and add documentation This fixes an issue where a user
+ of a dynamic conference was asked for a PIN twice. This also adds
+ documentation to assist in future modifications to the piece of
+ code responsible for PIN checking. (closes issue AST-670)
+ ........ Merged revisions 344439 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 344440 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: 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/ ........ Merged
+ revisions 344385 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 344386 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-10 16:29 +0000 [r344335] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_rtp_asterisk.c, /: only attempt to do stun handling on
+ ipv4 or ipv4 mapped to ipv6 addresses Patch by: jkonieczny
+ (modified) ASTERISK-18490 ........ Merged revisions 344330 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 344334 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-09 20:55 +0000 [r344272] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Fix deadlock during dialplan reload.
+ Another deadlock between the conlock/hints and channels/channel
+ locking orders. * Don't hold the channel and private lock in
+ sip_new() when calling ast_exists_extension(). (closes issue
+ ASTERISK-18740) Reported by: Byron Clark Patches:
+ sip_exists_exten_dlock_3.diff (license #5041) patch uploaded by
+ Gregory Hinton Nietsky ASTERISK-18740.patch (license #6157) patch
+ uploaded by Byron Clark Tested by: Byron Clark ........ Merged
+ revisions 344268 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 344271 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-09 20:10 +0000 [r344214-344217] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/reqresp_parser.c,
+ channels/sip/include/sip.h,
+ channels/sip/include/reqresp_parser.h: 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/ ........ Merged
+ revisions 344215 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 344216 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, tests/test_netsock2.c: Add a unit test for
+ ast_sockaddr_split_hostport Review:
+ https://reviewboard.asterisk.org/r/1575/ ........ Merged
+ revisions 344157 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 344175 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-09 19:08 +0000 [r344161] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh323.c, /, addons/ooh323c/src/ooh245.c,
+ addons/ooh323c/src/ooq931.h, addons/ooh323c/src/ootypes.h,
+ addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooq931.c:
+ Generate response to Status Enquiry message with Status q.931
+ message. Some PBXes require this for call status checking (closes
+ issue ASTERISK-18748) Reported by: Fabrizio Lazzaretti Patches:
+ ASTERISK-18748-5.patch (License #5415) patch uploaded by may213
+ Tested by: Fabrizio Lazzaretti ........ Merged revisions 344158
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 344159 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-09 17:15 +0000 [r344104] Kinsey Moore <kmoore@digium.com>
+
+ * /, apps/app_meetme.c: Fix pin parameter behavior regression in
+ MeetMe The last time this code was touched (by me), a subtlety
+ was missed based on the difference between needing to check a
+ pin's validity and the need to prompt for a pin. (closes issue
+ ASTERISK-18488) ........ Merged revisions 344102 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 344103 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-09 15:28 +0000 [r344050] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, formats/format_wav.c: don't call ltohl() twice on the same
+ value ASTERISK-18739 Patch by: pawel (modified) ........ Merged
+ revisions 344048 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 344049 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-08 22:14 +0000 [r344005] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Residual changes for Asterisk v10 branch
+ from ASTERISK-18747. Residual changes for Asterisk v10 branch
+ from ASTERISK-18747 after
+ https://reviewboard.asterisk.org/r/1564/ commit and associated
+ dialogs callid hash key change fix. * Make check_rtp_timeout()
+ return CMP_MATCH if need to delete dialog from dialogs_rtpcheck.
+ This is an optimization to avoid an unneeded lock/unlock and
+ object search when using ao2_unlink. * Prevent crash in
+ check_rtp_timeout() if dialog->rtp is NULL. Review:
+ https://reviewboard.asterisk.org/r/1557/ ........ Merged
+ revisions 344004 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-08 19:29 +0000 [r343951] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, pbx/pbx_config.c: Fix crash when dialplan remove include is
+ called with too few arguments. "dialplan remove include x from y"
+ crashed when the amount of arguments was less than 6. (closes
+ issue ASTERISK-18762) Reported by: Andrey Solovyev Tested by:
+ Andrey Solovyev ........ Merged revisions 343936 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 343944 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-08 18:35 +0000 [r343905] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 343900 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r343900 | dvossel | 2011-11-08 12:29:33 -0600 (Tue, 08 Nov 2011)
+ | 11 lines Fixes regression caused by r343635 There was a missing
+ unlock for a function return that is only present in Asterisk 10
+ and Asterisk Trunk. (closes issue ASTERISK-18839) Reported by:
+ Michael L. Young Patches:
+ asterisk-18839-missing-lock-trunk-v2.diff (License #5026) patch
+ uploaded by Michael L. Young ........
+
+2011-11-08 18:02 +0000 [r343853] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c, main/acl.c: Fixed reference to incorrect
+ variable if unknown host configured crash. * Fixed a LOG_ERROR
+ message referencing the config variable list v that had
+ previously been processed and became NULL. * Added error return
+ value set that was missing in an ast_append_ha() error return
+ path. (closes issue ASTERISK-18743) Reported by: Michele Patches:
+ issueA18743-fix_dynamic_exclude_static_bad_host_log.patch
+ (license #5674) patch uploaded by Walter Doekes Tested by:
+ Michele ........ Merged revisions 343851 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 343852 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-08 13:23 +0000 [r343790] Leif Madsen <leif@leifmadsen.com>
+
+ * /, build_tools/prep_tarball: Fix boo-boo in prep_tarball script.
+ A hardcoded a branch number was in the prep_tarball which could
+ not work. Changed it to the variable. ........ Merged revisions
+ 343789 from http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-07 22:37 +0000 [r343744] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Make "sip show settings" CLI command get
+ RPID flags from the right global page The "Trust RPID" and "Send
+ RPID" entries in the "sip show settings" CLI command pulled the
+ flags from the incorrect global flags page. These are now read
+ from sip global flags page 0. (closes issue AST-711) ........
+ Merged revisions 343743 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-07 21:58 +0000 [r343693] Leif Madsen <leif@leifmadsen.com>
+
+ * configs/dundi.conf.sample, pbx/pbx_dundi.c, CHANGES: Allow built
+ in variables to be used with dynamic weights. You can now use the
+ built in variables , , and within a dynamic weight. For example,
+ this could be useful when you want to pass requested lookup
+ number to the SHELL() function which could be used to execute a
+ script to dynamically set the weight of the result. (Closes issue
+ ASTERISK-13657) Reported by: Joel Vandal Tested by: Leif Madsen,
+ Russell Bryant Patches: asterisk-1.6-dundi-varhead.patch uploaded
+ by Joel Vandal (License #5374)
+
+2011-11-07 21:44 +0000 [r343692] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: respect case changes in peer names on sip
+ reload ASTERISK-18669 ........ Merged revisions 343690 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 343691 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-07 21:29 +0000 [r343684] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Fix __sip_subscribe_mwi_do() incorectly
+ changing dialogs hash key callid. Changing an object value used
+ as a container key requires removing the object from the
+ container and reinserting it. * Created change_callid_pvt() to
+ call instead of build_callid_pvt(). The change_callid_pvt() will
+ correctly change the dialog callid so the ao2 conainter can
+ explicitly unlink it. ........ Merged revisions 343637 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 343677 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-07 20:35 +0000 [r343636] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Prevent BLF subscriptions from causing
+ deadlocks Fix a locking inversion in sip_send_mwi_to_peer that
+ was causing deadlocks. This function now requires that both the
+ peer and associated pvt be unlocked before it is called for cases
+ where peer and peer->mwipvt form a circular reference. (closes
+ issue ASTERISK-18663) Review:
+ https://reviewboard.asterisk.org/r/1563/ ........ Merged
+ revisions 343621 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 343635 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-07 19:58 +0000 [r343581] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * main/udptl.c, /, UPGRADE.txt: Correct the default udptl port
+ range. The udptl port range was defined as 4000-4999 in the
+ udptl.conf.sample, as 4500-4599 if you didn't have a config and
+ 4500-4999 if your config was broken. Default is now 4000-4999.
+ (closes issue ASTERISK-16250) Reviewed by: Tilghman Lesher
+ Review: https://reviewboard.asterisk.org/r/1565 ........ Merged
+ revisions 343580 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-07 19:54 +0000 [r343579] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Fix deadlock if peer is destroyed while
+ sending MWI notice. A dialog cannot be destroyed by the
+ ao2_callback dialog_needdestroy because of a deadlock between the
+ dialogs container lock and the RWLOCK of the events subscription
+ list. * Create dialogs_to_destroy container to hold dialogs that
+ will be destroyed. * Ensure that the event subscription callback
+ will never happen with an invalid peer pointer by making the
+ event callback removal the first thing in the peer destructor
+ callback. NOTE: This particular deadlock will not happen with
+ Asterisk 10, but some of the changes still apply. (closes issue
+ ASTERISK-18747) Reported by: Gregory Hinton Nietsky Review:
+ https://reviewboard.asterisk.org/r/1564/ ........ Merged
+ revisions 343577 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 343578 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-07 18:42 +0000 [r343534] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/format.c, /: list all of the codecs associated with a
+ particular format id for CLI command "core show codec" AST-699
+ ........ Merged revisions 343533 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-06 09:51 +0000 [r343492] Olle Johansson <oej@edvina.net>
+
+ * main/tcptls.c, include/asterisk/tcptls.h: Formatting and doxygen
+ improvements
+
+2011-11-04 19:50 +0000 [r343448] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooGkClient.c, addons/ooh323c/src/ooTimer.c,
+ addons/ooh323c/src/dlist.c, /, addons/ooh323c/src/dlist.h,
+ addons/ooh323c/src/printHandler.c, addons/ooh323c/src/ooq931.c:
+ 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. ........ Merged revisions
+ 343281 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 343445 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-03 20:37 +0000 [r343394] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, res/res_config_sqlite.c: Fix sqlite config driver segfault and
+ broken queries The sqlite realtime handler assumed you had a
+ static config configured as well. The realtime multientry handler
+ assumed that you weren't using dynamic realtime. (closes issue
+ ASTERISK-18354) (closes issue ASTERISK-18355) Review:
+ https://reviewboard.asterisk.org/r/1561 ........ Merged revisions
+ 343375 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 343393 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-03 19:57 +0000 [r343338] Richard Mudgett <rmudgett@digium.com>
+
+ * /, funcs/func_dialgroup.c: Remove invalid flag given to iterator
+ in func_dialgroup.c ........ Merged revisions 343336 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 343337 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-03 15:40 +0000 [r343222-343278] Terry Wilson <twilson@digium.com>
+
+ * /, channels/sip/include/sip.h: Make room for the fax detect flags
+ The original REGISTERTRYING flag, in addition to being impossible
+ to check, also encroached on the space for the flag above it.
+ This patch moves the flags that were below REGISTERTRYING back to
+ where they were as though we had just removed the REGISTERTRYING
+ option. ........ Merged revisions 343276 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 343277 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * contrib/realtime/mysql/sippeers.sql, /, channels/chan_sip.c,
+ channels/sip/include/sip.h: 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/ ........ Merged
+ revisions 343220 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 343221 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-02 22:46 +0000 [r343163-343219] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, channels/chan_sip.c: Fix improper warning introduced by
+ r342927 and more tweaks Changeset r342927 introduced a warning
+ which was only supposed to be emitted when a found realtime peer
+ had an empty (or no) name. It turned out that there were some
+ inconsistencies left. Now found peers with an empty name are
+ explicitly ignored like before r342927 but better. Reviewed by:
+ Stefan Schmidts, Terry Wilson Review:
+ https://reviewboard.asterisk.org/r/1560 ........ Merged revisions
+ 343181 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 343192 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * include/asterisk/utils.h, /, main/utils.c,
+ include/asterisk/stringfields.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 ........ Merged revisions
+ 343157 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 343158 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-02 19:33 +0000 [r343049-343104] Leif Madsen <leif@leifmadsen.com>
+
+ * apps/app_authenticate.c: Add note about how Authenticate()
+ application with option 'd' works. (closes issue ASTERISK-17422)
+ Reported by: Leif Madsen ........ Merged revisions 343102 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 343103 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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) ........ Merged revisions 343047 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 343048 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-02 13:46 +0000 [r342992] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_meetme.c: Modify comments in MeetMe application
+ documentation about DAHDI. The MeetMe application documentation
+ has some comments about usage of DAHDI, and they were a bit
+ outdated relative to modern DAHDI releases. This patch changes
+ the comment to just tell the user that a functional DAHDI timing
+ source is required, and no longer mention 'dahdi_dummy', since
+ that module does not exist in current DAHDI releases. ........
+ Merged revisions 342990 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 342991 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-11-01 21:02 +0000 [r342871-342930] Walter Doekes <walter+asterisk@wjd.nu>
+
+ * /, channels/chan_sip.c, configs/extconfig.conf.sample,
+ include/asterisk/config.h, main/config.c: 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 ........
+ Merged revisions 342927 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 342929 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * contrib/realtime/mysql/sippeers.sql (added),
+ configs/res_config_mysql.conf.sample, /,
+ configs/extconfig.conf.sample, configs/res_ldap.conf.sample,
+ res/res_realtime.c, UPGRADE-1.8.txt, configs/dbsep.conf.sample,
+ main/config.c, contrib/realtime/mysql/sipfriends.sql (removed):
+ 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 ........ Merged revisions
+ 342869 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 342870 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-31 17:51 +0000 [r342825] Richard Mudgett <rmudgett@digium.com>
+
+ * main/format.c, /, main/format_cap.c: Misc format capability
+ fixes. * Fixed typo in format_cap.c:joint_copy_helper() using the
+ wrong variable. * Fix potential race between checking if an
+ interface exists and adding it to the container in
+ format.c:ast_format_attr_reg_interface(). * Fixed double rwlock
+ destroy in format.c:ast_format_attr_init() error exit path. *
+ Simplified format.c:find_interface() and
+ format.c:has_interface(). ........ Merged revisions 342824 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-31 16:10 +0000 [r342771] Matthew Jordan <mjordan@digium.com>
+
+ * main/pbx.c, /, channels/chan_iax2.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 ........ Merged
+ revisions 342769 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 342770 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-30 02:31 +0000 [r342716] Terry Wilson <twilson@digium.com>
+
+ * /, res/res_calendar.c: Don't crash on empty notify channel
+ ........ Merged revisions 342715 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-29 04:41 +0000 [r342663-342664] Richard Mudgett <rmudgett@digium.com>
+
+ * include/asterisk/linkedlists.h: Whitespace and some better macro
+ variable names. * Renamed AST_LIST_TRAVERSE_SAFE_BEGIN __new_prev
+ to __list_current. * Renamed AST_LIST_MOVE_CURRENT __list_cur to
+ __extracted.
+
+ * /, include/asterisk/linkedlists.h, tests/test_linkedlists.c: 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(). ........ Merged revisions 342661
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 342662 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-27 20:11 +0000 [r342606] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/dsp.c: tweak the v21 detector to detect an additional
+ pattern of hits and misses ........ Merged revisions 342605 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-27 19:48 +0000 [r342557-342604] Jonathan Rose <jrose@digium.com>
+
+ * res/res_rtp_multicast.c, /: Fix sequence number overflow over 16
+ bits causing codec change in RTP packets. Sequence number was
+ handled as an unsigned integer (usually 32 bits I think, more
+ depending on the architecture) and was put into the rtp packet
+ which is basically just a bunch of bits using an or operation.
+ Sequence number only has 16 bits allocated to it in an RTP packet
+ anyway, so it would add to the next field which just happened to
+ be the codec. This makes sure the sequence number is set to be a
+ 16 bit integer regardless of architecture (hopefully) and also
+ makes it so the incrementing of the sequence number does bitwise
+ or at the peak of a 16 bit number so that the value will be set
+ back to 0 when going beyond 65535 anyway. (closes issue
+ ASTERISK-18291) Reported by: Will Schick Review:
+ https://reviewboard.asterisk.org/r/1542/ ........ Merged
+ revisions 342602 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 342603 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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/ ........ Merged
+ revisions 342545 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 342546 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-25 22:06 +0000 [r342486-342489] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/astobj2.c: Check fopen return value for ao2 reference
+ debug output. Reported by: wdoekes Patched by: wdoekes Review:
+ https://reviewboard.asterisk.org/r/1539/ ........ Merged
+ revisions 342487 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 342488 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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!". ........
+ Merged revisions 342484 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 342485 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-25 21:11 +0000 [r342382-342437] Terry Wilson <twilson@digium.com>
+
+ * /, apps/app_queue.c: Use int for storing ao2_container_count
+ instad of size_t AST-676 ........ Merged revisions 342435 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 342436 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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/ ........ Merged
+ revisions 342383 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 342384 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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/ ........ Merged
+ revisions 342380 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 342381 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-25 19:09 +0000 [r342278-342330] Kinsey Moore <kmoore@digium.com>
+
+ * pbx/pbx_spool.c, /: Fix compilation on Snow Leopard/FreeBSD for
+ pbx_spool.c One of the changes in the recent spool handling of
+ hardlinks patch was just outside a HAVE_INOTIFY block and caused
+ compilation to fail in some build environments. This has been
+ corrected. ........ Merged revisions 342328 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 342329 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * pbx/pbx_spool.c, /: Merged revisions 342277 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r342277 | kmoore | 2011-10-25 11:08:04 -0500
+ (Tue, 25 Oct 2011) | 25 lines Merged revisions 342276 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r342276 | kmoore | 2011-10-25 11:06:57 -0500 (Tue, 25 Oct 2011) |
+ 18 lines 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 01:29 +0000 [r342225] Terry Wilson <twilson@digium.com>
+
+ * /, include/asterisk/config.h, main/config.c: 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/
+ ........ Merged revisions 342223 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 342224 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-24 22:37 +0000 [r342184] Richard Mudgett <rmudgett@digium.com>
+
+ * /, include/asterisk/astobj2.h: Fix ao2obj.h comment typos and add
+ missing link/unlink nolock debug defines. ........ Merged
+ revisions 342183 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-24 22:09 +0000 [r342148] Jonathan Rose <jrose@digium.com>
+
+ * main/features.c: Fixes a segfault caused by referencing null
+ frames introduced in r338623
+
+2011-10-24 21:01 +0000 [r342112] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_queue.c: Fix use of OBJ_KEY in Queue application. To use
+ the new OBJ_KEY flag, the container hash and compare callback
+ functions must be updated to support OBJ_KEY. Otherwise, bad
+ things happen. (issue ASTERISK-14769)
+
+2011-10-24 20:01 +0000 [r342063] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: Outbound SIP OPTIONS messages will now
+ include fromuser of related peer. This behavior matches up more
+ closely with the way invite/register/etc are handled. This patch
+ also modifies some adjacent code for code style compliance.
+ Pretty minor. (closes issue ASTERISK-17616) Reported by: Jeremy
+ Kister Patches: chan_sip.c-options-fromuser-fix-v1.patch uploaded
+ by Jeremy Kister (license #6232) ........ Merged revisions 342061
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 342062 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-24 07:40 +0000 [r341923-342018] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * /, apps/app_queue.c: queues container needs locking when using
+ the OBJ_NOLOCK flag ........ Merged revisions 342017 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, apps/app_queue.c: Remove some ref leaks and a return without
+ unlock. There some resource leaks introduced in asterisk 10 make
+ sure that locks are not held on return and we release ref's held.
+ ........ Merged revisions 341972 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * apps/app_queue.c: Whitespace Fixups / Add Braces This janitorial
+ patch is related to work on RB1538
+
+2011-10-22 12:03 +0000 [r341869] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/chan_ooh323.c, /: Merged revisions 341313 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r341313 | may | 2011-10-19 03:33:49 +0400 (Wed,
+ 19 Oct 2011) | 10 lines Merged revisions 341312 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r341312 | may | 2011-10-19 03:20:53 +0400 (Wed, 19 Oct 2011) | 3
+ lines fix issue on channel numbering (calls could have same
+ channel number on heavy loaded system) ........ ................
+
+2011-10-21 16:42 +0000 [r341808-341811] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, pbx/pbx_lua.c: only process args that exist ASTERISK-18395
+ ........ Merged revisions 341809 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 341810 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, pbx/pbx_lua.c: don't limit the length of app and function
+ arguments ASTERISK-18395 ........ Merged revisions 341806 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 341807 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-21 09:16 +0000 [r341769] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * res/res_fax.c: White space fixes in res_fax
+
+2011-10-20 22:03 +0000 [r341719] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c, res/res_agi.c, include/asterisk/features.h:
+ 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) ........ Merged revisions 341717 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 341718 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-20 21:28 +0000 [r341666-341713] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, funcs/func_callerid.c: Fixed typo from previous commit
+ ........ Merged revisions 341704 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 341707 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, funcs/func_callerid.c: Updated documentation for the optional
+ CID parameter with CALLERID ........ Merged revisions 341664 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 341665 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-20 18:27 +0000 [r341583-341624] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * /, configs/queues.conf.sample: Merged revisions 341599 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ........ r341599 | irroot | 2011-10-20 20:20:08 +0200 (Thu, 20
+ Oct 2011) | 8 lines add documentation for check_state_unknown in
+ configs/queues.conf.sample app_queue allows calls to members in a
+ "Unknown" state to be treated as available setting
+ check_state_unknown = yes will cause app_queue to query the
+ channel driver to better determine the state this only applies to
+ queues with ringinuse or ignorebusy set appropriately. ........
+
+ * /, CHANGES, apps/app_queue.c: Merged revisions 341580 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ........ r341580 | irroot | 2011-10-20 19:13:23 +0200 (Thu, 20
+ Oct 2011) | 15 lines Add option to check state when state is
+ unknown r341486 reverts r325483 this is a rework of the patch.
+ optimize to minimize load. add option check_state_unknown to
+ control whether a member with unknown device state is checked
+ there is a small % chance that calls will be sent to the member
+ when they on a call. app_queue will see a device with unknown
+ state as available and does not try verify the state without this
+ option enabled. Review: https://reviewboard.asterisk.org/r/1535/
+ ........
+
+2011-10-20 15:17 +0000 [r341533] Terry Wilson <twilson@digium.com>
+
+ * /, include/asterisk/strings.h: Clean up ast_check_digits The code
+ was originally copied from the is_int() function in the AEL code.
+ wdoekes pointed out that the function should take a const char*
+ and that their was an unneeded variable. This is now fixed.
+ ........ Merged revisions 341529 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 341530 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-19 21:24 +0000 [r341487] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 341486 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r341486 | mnicholson | 2011-10-19 16:23:17 -0500 (Wed, 19 Oct
+ 2011) | 18 lines Fix a performance regression introduced in
+ r325483. The regression was caused by a call to
+ ast_parse_device_state() in app_queue's ring_entry() function.
+ The ast_parse_device_state() function eventually calls
+ ast_channel_get_full() with a channel name prefix which causes it
+ to walk the channel list causing massive lock contention and slow
+ downs. This patch fixes the regression by removing the call to
+ ast_parase_device_state() which should be unnecessary. Queue
+ member device state should be maintained by device state events.
+ Some users have seen instances where busy agents were called when
+ they shouldn't have, which is the reason the call to
+ ast_parse_device_state() was added. That change appears to have
+ resolved that issue but also causes this performance regression.
+ There may still be issues with queue member status, and if so,
+ alternative methods should be investigated to resolve them.
+ AST-695 ........
+
+2011-10-19 19:02 +0000 [r341437] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, channels/chan_gtalk.c: Outgoing calls with Google Voice Google
+ has recently make some changes (again) to their protocol. Rather
+ then patching asterisk to flip between the two different methods,
+ we now allow both. Lets hope this keeps Google Voice happy for a
+ while. (closes issue ASTERISK-18714) Reported by: Iordan Iordanov
+ Patches: chan_gtalk.patch uploaded by Iordan Iordanov (licenses
+ 6311) ........ Merged revisions 341435 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 341436 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-19 07:45 +0000 [r341381] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c, include/asterisk/strings.h: 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 ........ Merged revisions 341379 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 341380 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-19 07:27 +0000 [r341378] Stefan Schmidt <sst@sil.at>
+
+ * /, channels/chan_sip.c: Don't sent in-dialog requests like UPDATE
+ when Asterisk has not yet received a Contact URI from a UAS
+ ........ Merged revisions 341366 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 341377 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-18 23:45 +0000 [r341316] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Don't resolve numeric hosts or contact
+ unresolved hosts If a SIP dial string contains a numeric hostname
+ that is not a peer name, don't try to resolve it as it is
+ unlikely that someone really means Dial(SIP/0.0.4.26) when
+ Dial(SIP/1050) is called. Also, make sure that create_addr
+ returns -1 if an address isn't resolved so that we don't attempt
+ to send SIP requests to an address that doesn't resolve. (closes
+ issue ASTERISK-17146, ASTERISK-17716) Review:
+ https://reviewboard.asterisk.org/r/1532/ ........ Merged
+ revisions 341314 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 341315 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-18 21:15 +0000 [r341256] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/chan_sip.c, main/features.c, channels/chan_iax2.c,
+ channels/sip/include/sip.h, channels/chan_mgcp.c,
+ include/asterisk/features.h: 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. ........
+ Merged revisions 341254 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 341255 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-17 17:58 +0000 [r341198] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * /, pbx/pbx_realtime.c: Remove an unused include of md5.h Unused
+ include of asterisk/md5.h in pbx_realtime.c . A commit needed to
+ test the commit message. Merged-From:
+ http://svn.asterisk.org/svn/asterisk/branches/1.8@341074
+ Merged-From:
+ http://svn.asterisk.org/svn/asterisk/branches/10@341148
+
+2011-10-17 17:38 +0000 [r341191] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Initialize variables before calling
+ parse_uri If parse_uri was called with an empty URI, some
+ pointers would be modified and an invalid read could result. This
+ patch avoids calling parse_uri with an empty contact uri when
+ parsing REGISTER requests. AST-2011-012 (closes issue
+ ASTERISK-18668) ........ Merged revisions 341189 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 341190 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-17 16:39 +0000 [r341126-341147] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, tests/test_format_api.c: Set 'core' support level for
+ test_format_api.c ........ Merged revisions 341146 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, apps/app_voicemail.c: Multiple revisions 341108,341112
+ ........ r341108 | pabelanger | 2011-10-17 12:22:19 -0400 (Mon,
+ 17 Oct 2011) | 2 lines Voicemail compiler flags are 'core'
+ support ........ r341112 | pabelanger | 2011-10-17 12:23:33 -0400
+ (Mon, 17 Oct 2011) | 2 lines Fix previous commit ........ Merged
+ revisions 341108,341112 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 341122 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-17 16:18 +0000 [r341096] Jason Parker <jparker@digium.com>
+
+ * /, CHANGES: Add information about limitations of new codec
+ support in channel drivers. (issue ASTERISK-18680) ........
+ Merged revisions 341094 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-17 15:45 +0000 [r341090] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Don't try to remove peers without IPs
+ from peers_by_ip (closes issue ASTERISK-18696) ........ Merged
+ revisions 341088 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 341089 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-14 21:37 +0000 [r341024] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, build_tools/embed_modules.xml, Makefile.moddir_rules: Change
+ the internal name of the menuselect options that are used to
+ control whether modules are embedded or not; using just the bare
+ category name led to accidentally enabling these options when
+ users used the wrong "--enable" operation on the menuselect
+ command line. Now the internal option names are prefixed with
+ "EMBED_", so they won't be the same as the name of the category
+ containing the modules they control the embedding of. ........
+ Merged revisions 341022 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 341023 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-14 21:15 +0000 [r340973] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Fix simple switch to not progress a call
+ when call already progressed. If a simple switch was started on a
+ device and then a specific call made (such as redial or speed
+ dial), on timeout of the simple switch the call would be
+ attempted again. This patch only allows the simple switch to make
+ a call if the substate is still in the collecting digits mode.
+ Also added small debug message to dialAndAactivate sub. Tested by
+ snuff and myself.
+
+2011-10-14 20:51 +0000 [r340972] Kinsey Moore <kmoore@digium.com>
+
+ * res/res_rtp_asterisk.c, /, channels/chan_sip.c: Merged revisions
+ 340971 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r340971 | kmoore | 2011-10-14 15:50:37 -0500
+ (Fri, 14 Oct 2011) | 15 lines Merged revisions 340970 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r340970 | kmoore | 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) |
+ 8 lines 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 18:38 +0000 [r340932] Jonathan Rose <jrose@digium.com>
+
+ * utils/utils.xml, /, funcs/func_jitterbuffer.c: Some additional
+ module documentation changes for 10 for the menuselect change.
+ (issue ASTERISK-18268) ........ Merged revisions 340931 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-14 16:45 +0000 [r340880] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, /: Avoid unnecessary WARNING message Add
+ AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid
+ displaying a WARNING message. (closes issue ASTERISK-18610) Patch
+ by: Kristijan_Vrban ........ Merged revisions 340878 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 340879 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-13 23:08 +0000 [r340811-340813] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Fix DTMF blind transfer continuing to execute
+ dialplan after transfer. Party A calls Party B. Party A DTMF
+ blind transfers Party B to Party C. Party A channel continues to
+ execute dialplan. * Fixed the return value of
+ builtin_blindtransfer() to return the correct value after a
+ transfer so the dialplan will not keep executing. * Removed
+ unnecessary connected line update that did not really do
+ anything. * Made access to GOTO_ON_BLINDXFR thread safe in
+ check_goto_on_transfer(). * Fixed leak of xferchan for failure
+ cases in check_goto_on_transfer(). * Updated debug messages in
+ builtin_blindtransfer() and check_goto_on_transfer(). (closes
+ issue ASTERISK-18275) Reported by: rmudgett Tested by: rmudgett
+ ........ Merged revisions 340809 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 340810 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /: Update 10 merged property.
+
+ * /: Restore branch 10 merge properties.
+
+2011-10-13 08:53 +0000 [r340771] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * /: Merged revisions 339463 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r339463 | irroot | 2011-10-05 08:28:46 +0200 (Wed, 05 Oct 2011) |
+ 9 lines Only change the capabilities on the gateway when the
+ session is been destroyed there is still a race condition that
+ ends in a segfault. if the caps are changed the logic in
+ res_fax_spandsp will run T30 code not gateway code to end the
+ session. this has been experienced on a "slower" under spec
+ system. ........
+
+2011-10-13 07:05 +0000 [r340720] Stefan Schmidt <sst@sil.at>
+
+ * channels/chan_sip.c: Merged revisions 340718 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r340718 | schmidts | 2011-10-13 06:59:50 +0000
+ (Thu, 13 Oct 2011) | 9 lines Merged revisions 340717 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r340717 | schmidts | 2011-10-13 06:58:00 +0000 (Thu, 13
+ Oct 2011) | 3 lines storing the route-set also on a 181 response
+ not only on 180,182 or 183. ........ ................
+
+2011-10-13 07:02 +0000 [r340665-340719] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Initialize ast_sockaddr before calling
+ ast_sockaddr_resolve Avoid possible jump based on unitialized
+ value ........ Merged revisions 340715 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 340716 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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. ........ Merged revisions 340662 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 340663 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-12 21:28 +0000 [r340626] Stefan Schmidt <sst@sil.at>
+
+ * channels/chan_sip.c: Merged revisions 340577 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r340577 | schmidts | 2011-10-12 20:33:37 +0000
+ (Mit, 12 Okt 2011) | 9 lines Merged revisions 340576 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r340576 | schmidts | 2011-10-12 20:30:37 +0000 (Mit, 12
+ Okt 2011) | 3 lines Store route-set from provisional SIP
+ responses so early-dialog requests can be routed properly
+ ........ ................
+
+2011-10-12 21:02 +0000 [r340579] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 340578 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r340578 | twilson | 2011-10-12 13:57:19 -0700
+ (Wed, 12 Oct 2011) | 16 lines Merged revisions 340534 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r340534 | twilson | 2011-10-12 13:19:36 -0700 (Wed, 12 Oct 2011)
+ | 9 lines 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 20:09 +0000 [r340472-340524] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Initialize the PRI channel alarms
+ properly on startup. The PRI channel alarms were initialized with
+ an inverted sense. (closes issue ASTERISK-18710) Reported by:
+ Tzafrir Cohen ........ Merged revisions 340522 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 340523 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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 ........ Merged revisions 340470 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 340471 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-12 16:29 +0000 [r340420] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, channels/chan_sip.c: Fix verbose messages when IPv6 logic was
+ added (closes issue ASTERISK-18612) Reported by: Tim Osman
+ ........ Merged revisions 340418 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 340419 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-11 21:06 +0000 [r340318-340367] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_ss7.h, /, channels/sig_ss7.c:
+ 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 ........ Merged revisions 340365 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 340366 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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. ........ Merged revisions 340284 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 340310 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-11 19:06 +0000 [r340283] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * main/channel.c, /, main/sha1.c, include/asterisk/sha1.h: 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/
+ Merge-From:
+ http://svn.asterisk.org/svn/asterisk/branches/1.8@340263
+ Merge-From:
+ http://svn.asterisk.org/svn/asterisk/branches/10@340280
+
+2011-10-11 18:57 +0000 [r340282] Richard Mudgett <rmudgett@digium.com>
+
+ * main/manager.c, /, include/asterisk/manager.h: 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 ........ Merged revisions 340279 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 340281 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-10-10 23:10 +0000 [r340221-340224] Terry Wilson <twilson@digium.com>
+
+ * UPGRADE.txt, main/db.c: Return error when no rows are deleted for
+ AMI DBDelTree (closes issue AST-654)
+
+ * /, main/db.c: Merged revisions 340222 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r340222 | twilson | 2011-10-10 15:55:39 -0700 (Mon, 10 Oct 2011)
+ | 8 lines On astdb conversion, also warn about permissions
+ requirements The user running Asterisk must have permission to
+ the directory the Asterisk database resides in since SQLite 3
+ needs to be able to create a journal file. (closes issue
+ ASTERISK-18174) ........
+
+ * utils/Makefile, utils/utils.xml, /, UPGRADE.txt,
+ utils/astdb2bdb.c (added): Merged revisions 340219-340220 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ........ r340219 | twilson | 2011-10-10 15:38:06 -0700 (Mon, 10
+ Oct 2011) | 8 lines Add astdb conversion utility for Berkeley to
+ SQLite 3 If someone wants to backtrack from Asterisk 1.8 to 10
+ they can use the astdb2bdb utility to convert the database back
+ to the Berkeley format that Asterisk 1.8 uses. Review:
+ https://reviewboard.asterisk.org/r/1502/ ........ r340220 |
+ twilson | 2011-10-10 15:39:41 -0700 (Mon, 10 Oct 2011) | 2 lines
+ Add a missing file for the astdb2bdb conversion utility ........
+
+2011-10-10 20:39 +0000 [r340166] Matthew Jordan <mjordan@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 340165 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r340165 | mjordan | 2011-10-10 15:30:18 -0500
+ (Mon, 10 Oct 2011) | 20 lines Merged revisions 340164 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r340164 | mjordan | 2011-10-10 15:23:48 -0500 (Mon, 10 Oct 2011)
+ | 13 lines 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 14:16 +0000 [r340110] Matthew Nicholson <mnicholson@digium.com>
+
+ * 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, doc/appdocsxml.dtd, main/loader.c,
+ main/xmldoc.c: Merged revisions 340109 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r340109 | mnicholson | 2011-10-10 09:15:41 -0500
+ (Mon, 10 Oct 2011) | 18 lines Merged revisions 340108 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r340108 | mnicholson | 2011-10-10 09:14:48 -0500 (Mon, 10 Oct
+ 2011) | 11 lines 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/ ........
+ ................
+
+2011-10-10 00:57 +0000 [r339993-340071] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Add skinny version 17 protocol support.
+ Added some data to skinny packet structures to make compatible
+ with v17. Added protocolversion to device, set on registration
+ based on the version provided by device. v17 includes some
+ increased ip space for ip6. This patch increases ip space in the
+ packets but still only uses ip4. Some packet structures
+ duplicated (ip4 and ip6 types). ip4 type used unless version is
+ greater or equal to 17. Tested by snuff and myself on 7961 with
+ recent 8.5 firmware. Also tested compatible with old 7960 and
+ older 30VIPs.
+
+ * channels/chan_skinny.c: Increase SKINNY_MAX_PACKET and add some
+ logging. Increase SKINNY_MAX_PACKET to 2000 bytes to handle some
+ messages in v17 that are greater than the old 1000 bytes. Also
+ add some useful logging regarding packet and session handling. A
+ device (with protocol v17) was sending a packet with length
+ greater than 1000 which resulted in the TCP session being
+ destroyed and registration being retryed.
+
+ * /, channels/chan_skinny.c: Merged revisions 340031 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r340031 | wedhorn | 2011-10-10 09:18:27 +1100 (Mon, 10 Oct 2011)
+ | 8 lines Return -1 to skinny_session if register rejected. If
+ device registration is rejected, return -1 so that the session is
+ destroyed immediately. Previously, a segfault would occur on a
+ graceful shutdown if a register is rejected and the
+ skinny_session has not yet timed out. ........
+
+ * /, channels/chan_skinny.c: Merged revisions 339992 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r339992 | wedhorn | 2011-10-10 08:09:12 +1100 (Mon, 10 Oct 2011)
+ | 9 lines Remove log message on traverse session list. On
+ destroying a session, a list of sessions is traversed to find the
+ matching session. For each session not matching, skinny
+ erroneously logged that the session was not matched. While
+ technically correct the message was misleading, and tended to
+ indicate errors that were not there. ........
+
+2011-10-09 01:19 +0000 [r339832-339947] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
+
+ * channels/chan_unistim.c, /: Merged revisions 339942 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r339942 | igorg | 2011-10-09 08:18:02 +0700
+ (Вск, 09 Окт 2011) | 12 lines Merged revisions 339938 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339938 | igorg | 2011-10-09 08:16:09 +0700 (Вск, 09 Окт 2011) |
+ 6 lines Fix compilation issue, caused by missed session structure
+ (closes issue ASTERISK-18694) Reported by: alex70 ........
+ ................
+
+ * channels/chan_unistim.c, /: Merged revisions 339885 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r339885 | igorg | 2011-10-08 22:46:27 +0700
+ (Сбт, 08 Окт 2011) | 13 lines Merged revisions 339884 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339884 | igorg | 2011-10-08 22:45:20 +0700 (Сбт, 08 Окт 2011) |
+ 7 lines Fix segfault in Unistim channel (closes issue
+ ASTERISK-18638) Reported by: jonnt ........ ................
+
+ * channels/chan_unistim.c, /: Merged revisions 339831 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r339831 | igorg | 2011-10-08 22:01:35 +0700
+ (Сбт, 08 Окт 2011) | 14 lines Merged revisions 339830 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339830 | igorg | 2011-10-08 21:56:35 +0700 (Сбт, 08 Окт 2011) |
+ 8 lines Fix char array cast as short array in send_client()
+ function (for ARM platform) (closes issue ASTERISK-17314)
+ Reported by: jjoshua ........ ................
+
+2011-10-07 19:37 +0000 [r339721-339778] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_url.c: Merged revisions 339777 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r339777 | rmudgett | 2011-10-07 14:36:24 -0500
+ (Fri, 07 Oct 2011) | 12 lines Merged revisions 339776 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339776 | rmudgett | 2011-10-07 14:34:55 -0500 (Fri, 07 Oct 2011)
+ | 5 lines Initialize option flags for SendURL application.
+ (closes issue ASTERISK-18574) Reported by: marcelloceschia
+ ........ ................
+
+ * /: Recorded merge of revisions 339681 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r339681 | wedhorn | 2011-10-06 15:47:08 -0500 (Thu, 06 Oct 2011)
+ | 10 lines Fixed segfault on core stop gracefully. There was an
+ issue that the cap and confcap pointers for each line and device
+ were being memcpy'd so they all pointed to the same
+ ast_format_cap. On destroying, a segfault occured on the second
+ call to the same struct. skinny reload now works again as well.
+ Tested by snuff (in trunk) and myself. ........
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ autoconf/ast_ext_lib.m4: Merged revisions 339720 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r339720 | rmudgett | 2011-10-06 17:58:40 -0500
+ (Thu, 06 Oct 2011) | 27 lines Merged revisions 339719 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ 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 ........
+ ................
+
+2011-10-06 20:18 +0000 [r339680] Damien Wedhorn <voip@facts.com.au>
+
+ * channels/chan_skinny.c: Fixed segfault on core stop gracefully.
+ There was an issue that the cap and confcap pointers for each
+ line and device were being memcpy'd so they all pointed to the
+ same ast_format_cap. On destroying, a segfault occured on the
+ second call to the same struct. skinny reload now works again as
+ well. Tested by snuff and myself.
+
+2011-10-06 17:54 +0000 [r339627] Richard Mudgett <rmudgett@digium.com>
+
+ * main/udptl.c, /, channels/chan_sip.c: Merged revisions 339626 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r339626 | rmudgett | 2011-10-06 12:53:00 -0500
+ (Thu, 06 Oct 2011) | 25 lines Merged revisions 339625 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339625 | rmudgett | 2011-10-06 12:49:38 -0500 (Thu, 06 Oct 2011)
+ | 18 lines 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 ........ ................
+
+2011-10-06 13:43 +0000 [r339587] Leif Madsen <leif@leifmadsen.com>
+
+ * build_tools/prep_tarball: Merged revisions 339586 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r339586 | lmadsen | 2011-10-06 08:43:21 -0500
+ (Thu, 06 Oct 2011) | 16 lines Merged revisions 339566 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339566 | lmadsen | 2011-10-05 16:30:11 -0500 (Wed, 05 Oct 2011)
+ | 8 lines 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 17:02 +0000 [r339510-339513] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 339512 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r339512 | rmudgett | 2011-10-05 12:01:46 -0500
+ (Wed, 05 Oct 2011) | 9 lines Merged revisions 339511 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r339511 | rmudgett | 2011-10-05 12:01:01 -0500 (Wed, 05
+ Oct 2011) | 1 line Fix Dial F option notes formatting. ........
+ ................
+
+ * main/manager.c, /: Merged revisions 339508 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r339508 | rmudgett | 2011-10-05 11:35:02 -0500
+ (Wed, 05 Oct 2011) | 18 lines Merged revisions 339504,339506 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339504 | rmudgett | 2011-10-05 11:26:45 -0500 (Wed, 05 Oct 2011)
+ | 7 lines 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 ........ r339506 | rmudgett | 2011-10-05
+ 11:32:03 -0500 (Wed, 05 Oct 2011) | 1 line Fix XML error in AMI
+ action Challenge. ........ ................
+
+2011-10-05 16:35 +0000 [r339509] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, res/res_fax.c: Merged revisions 339507 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r339507 | mnicholson | 2011-10-05 11:32:59 -0500
+ (Wed, 05 Oct 2011) | 10 lines Merged revisions 339505 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339505 | mnicholson | 2011-10-05 11:31:21 -0500 (Wed, 05 Oct
+ 2011) | 3 lines The app name in the documentation must match what
+ we register the application as. ........ ................
+
+2011-10-05 06:50 +0000 [r339464-339465] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * res/res_fax.c, include/asterisk/res_fax.h, CHANGES: Add generic
+ faxdetect framehook to res_fax Added func
+ FAXOPT(faxdetect)=yes,cng,t38[,timeout]/no to enable dialplan
+ faxdetect allowing more flexibility. as soon as a fax tone is
+ detected the framehook is removed. there is a penalty involved in
+ running this framehook on non G711 channels as they will be
+ transcoded. CNG tone is suppresed using the SQUELCH flag to allow
+ WaitForNoise to be run on the channel to detect Voice. (Closes
+ issue ASTERISK-18569) Reported by: Myself Reviewed by: Matthew
+ Nicholson, Kevin Fleming Review:
+ https://reviewboard.asterisk.org/r/1116/
+
+ * /, res/res_fax.c: Merged revisions 339463 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r339463 | irroot | 2011-10-05 08:28:46 +0200 (Wed, 05 Oct 2011) |
+ 9 lines Only change the capabilities on the gateway when the
+ session is been destroyed there is still a race condition that
+ ends in a segfault. if the caps are changed the logic in
+ res_fax_spandsp will run T30 code not gateway code to end the
+ session. this has been experienced on a "slower" under spec
+ system. ........
+
+2011-10-04 22:59 +0000 [r339408] Richard Mudgett <rmudgett@digium.com>
+
+ * Makefile, /: Merged revisions 339407 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r339407 | rmudgett | 2011-10-04 17:56:25 -0500
+ (Tue, 04 Oct 2011) | 15 lines Merged revisions 339406 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339406 | rmudgett | 2011-10-04 17:54:15 -0500 (Tue, 04 Oct 2011)
+ | 8 lines 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 19:51 +0000 [r339315-339354] Jonathan Rose <jrose@digium.com>
+
+ * /, main/say.c: Merged revisions 339353 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r339353 | jrose | 2011-10-04 14:44:02 -0500
+ (Tue, 04 Oct 2011) | 18 lines Merged revisions 339352 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339352 | jrose | 2011-10-04 14:33:12 -0500 (Tue, 04 Oct 2011) |
+ 12 lines 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: Merged revisions 339298 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r339298 | jrose | 2011-10-04 09:09:50 -0500
+ (Tue, 04 Oct 2011) | 19 lines Merged revisions 339297 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339297 | jrose | 2011-10-04 09:01:05 -0500 (Tue, 04 Oct 2011) |
+ 13 lines 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 12:27 +0000 [r339262] Alexandr Anikin <may@telecom-service.ru>
+
+ * /, addons/ooh323c/src/memheap.c: Merged revisions 339245 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r339245 | may | 2011-10-04 15:49:49 +0400 (Tue,
+ 04 Oct 2011) | 9 lines Merged revisions 339244 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339244 | may | 2011-10-04 15:44:55 +0400 (Tue, 04 Oct 2011) | 2
+ lines fix forget declaration in previous change ........
+ ................
+
+2011-10-04 09:43 +0000 [r339206] Olle Johansson <oej@edvina.net>
+
+ * main/manager.c, CHANGES: Generate error message when AMI action
+ originate extension doesn't exist Review:
+ https://reviewboard.asterisk.org/r/1445/ Is this a bug or a new
+ feature? No responses on Asterisk-dev so I'm committing to trunk
+ only.
+
+2011-10-03 20:13 +0000 [r339146-339149] Leif Madsen <leif@leifmadsen.com>
+
+ * channels/chan_sip.c: Merged revisions 339148 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r339148 | lmadsen | 2011-10-03 15:13:16 -0500
+ (Mon, 03 Oct 2011) | 14 lines Merged revisions 339147 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339147 | lmadsen | 2011-10-03 15:12:43 -0500 (Mon, 03 Oct 2011)
+ | 6 lines 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: Merged revisions 339145 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r339145 | lmadsen | 2011-10-03 14:55:15 -0500
+ (Mon, 03 Oct 2011) | 13 lines Merged revisions 339144 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339144 | lmadsen | 2011-10-03 14:54:52 -0500 (Mon, 03 Oct 2011)
+ | 6 lines 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 19:16 +0000 [r339091] Alexandr Anikin <may@telecom-service.ru>
+
+ * /, addons/ooh323c/src/memheap.c: Merged revisions 339089 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r339089 | may | 2011-10-03 22:52:55 +0400 (Mon,
+ 03 Oct 2011) | 10 lines Merged revisions 339087 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339087 | may | 2011-10-03 22:42:49 +0400 (Mon, 03 Oct 2011) | 4
+ lines destroy memheap mutex properly before memheap deleted (fix
+ memory leak occured after r304950 changes with DEBUG_THREAD
+ compile option) ........ ................
+
+2011-10-03 18:58 +0000 [r339090] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c, main/file.c: Merged revisions 339088 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r339088 | twilson | 2011-10-03 11:44:27 -0700
+ (Mon, 03 Oct 2011) | 17 lines Merged revisions 339086 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r339086 | twilson | 2011-10-03 11:40:52 -0700 (Mon, 03 Oct 2011)
+ | 10 lines 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 ........ ................
+
+2011-10-03 15:55 +0000 [r339021-339046] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, res/res_fax.c: Merged revisions 339045 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r339045 | mnicholson | 2011-10-03 10:54:55 -0500 (Mon, 03 Oct
+ 2011) | 4 lines Ported ast_fax_caps_to_str() to 10, not sure why
+ it wasn't already here. This function prints a list of caps
+ instead of a hex bitfield. ........
+
+ * /, res/res_fax.c: Merged revisions 339043 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r339043 | mnicholson | 2011-10-03 10:41:36 -0500 (Mon, 03 Oct
+ 2011) | 2 lines Don't clear the AST_FAX_TECH_MULTI_DOC flag right
+ after we set it. ........
+
+ * /, res/res_fax.c: Merged revisions 339011 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r339011 | mnicholson | 2011-10-03 10:19:44 -0500 (Mon, 03 Oct
+ 2011) | 2 lines properly remove the AST_FAX_TECH_GATEWAY flag
+ (instead of setting all of the other flags) ........
+
+2011-10-03 14:40 +0000 [r338905-338998] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * /, CHANGES: Merged revisions 338997 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r338997 | irroot | 2011-10-03 16:38:25 +0200 (Mon, 03 Oct 2011) |
+ 1 line Documentation noting the extension of CHANNEL() for
+ chan_ooh323 ........
+
+ * addons/chan_ooh323.c, /, funcs/func_channel.c: Merged revisions
+ 338995 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r338995 | irroot | 2011-10-03 16:21:40 +0200 (Mon, 03 Oct 2011) |
+ 6 lines Remove the channel function OOH323() and place its
+ options into CHANNEL() channel drivers should not have there own
+ dialplan functions. ........
+
+ * /, res/res_fax.c: Merged revisions 338950 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r338950 | irroot | 2011-10-03 11:37:59 +0200 (Mon, 03 Oct 2011) |
+ 14 lines Fixup a race condition in res_fax.c where
+ FAXOPT(gateway)=no will turn off the gateway but the framehook is
+ not destroyed. this problem happens when a gateway is attempted
+ in the dialplan and the device is not available i may want to do
+ fax to mail in the server it will not be allowed. instead of
+ checking only AST_FAX_TECH_GATEWAY also check gateway_id Reverts
+ 338904 Fix some white space. ........
+
+ * /, res/res_fax.c: Merged revisions 338904 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r338904 | irroot | 2011-10-02 16:17:32 +0200 (Sun, 02 Oct 2011) |
+ 8 lines Remove T38 Gateway capability when detaching framehook.
+ SET(FAXOPT(gateway)=no) does not remove the capability when
+ detaching the framehook. small patch to fix this problem.
+ ........
+
+2011-10-01 01:56 +0000 [r338855] TransNexus OSP Development <support@transnexus.com>
+
+ * configure: Update "configure" based on r338139.
+
+2011-09-30 22:08 +0000 [r338802] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 338801 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r338801 | rmudgett | 2011-09-30 17:06:48 -0500
+ (Fri, 30 Sep 2011) | 19 lines Merged revisions 338800 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r338800 | rmudgett | 2011-09-30 17:05:10 -0500 (Fri, 30 Sep 2011)
+ | 12 lines 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 19:25 +0000 [r338755] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Formatting changes only --Denna och
+ nedanstående rader kommer inte med i loggmeddelandet-- M
+ channels/chan_sip.c
+
+2011-09-30 18:59 +0000 [r338720] Jonathan Rose <jrose@digium.com>
+
+ * /, configs/queues.conf.sample: Merged revisions 338719 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r338719 | jrose | 2011-09-30 13:55:27 -0500
+ (Fri, 30 Sep 2011) | 9 lines Merged revisions 338718 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r338718 | jrose | 2011-09-30 13:54:30 -0500 (Fri, 30 Sep
+ 2011) | 1 line Adds documentation for QueueMemberStatus event
+ generation ........ ................
+
+2011-09-30 16:40 +0000 [r338665] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/chan_sip.c: Fix formatting of AMI header for SIP show
+ peer. ASTERISK-17486 exposed the problem for AMI parsers. (closes
+ issue ASTERISK-18649) Reported by: Jacek Konieczny Patches:
+ asterisk-sipshowpeer_response_end.patch (license #6298) patch
+ uploaded by Jacek Konieczny ........ Merged revisions 338663 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 338664 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2011-09-30 13:21 +0000 [r338623] Olle Johansson <oej@edvina.net>
+
+ * main/features.c: Preserve DTMF length in main/features.c Review:
+ https://reviewboard.asterisk.org/r/1463/ A small part of much
+ larger work with DTMF duration in Asterisk, funded by IPvision AS
+ in Denmark. Thanks to irroot for the review!
+
+2011-09-29 21:16 +0000 [r338557] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * tests/test_security_events.c, /, tests/test_locale.c,
+ tests/test_logger.c, tests/test_dlinklists.c,
+ tests/test_linkedlists.c, tests/test_amihooks.c: Merged revisions
+ 338556 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r338556 | pabelanger | 2011-09-29 17:14:34 -0400
+ (Thu, 29 Sep 2011) | 9 lines Merged revisions 338555 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r338555 | pabelanger | 2011-09-29 17:12:21 -0400 (Thu,
+ 29 Sep 2011) | 2 lines Test modules should depend on the
+ TEST_FRAMEWORK flag ........ ................
+
+2011-09-29 20:55 +0000 [r338553] Jason Parker <jparker@digium.com>
+
+ * /, tests/test_db.c, tests/test_netsock2.c: Merged revisions
+ 338552 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r338552 | qwell | 2011-09-29 15:54:55 -0500
+ (Thu, 29 Sep 2011) | 9 lines Merged revisions 338551 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r338551 | qwell | 2011-09-29 15:54:13 -0500 (Thu, 29 Sep
+ 2011) | 1 line Test modules have a support level of core.
+ ........ ................
+
+2011-09-29 12:22 +0000 [r338435] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged
+ revisions 338417 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r338417 | irroot | 2011-09-29 14:16:42 +0200
+ (Thu, 29 Sep 2011) | 19 lines Merged revisions 338416 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r338416 | irroot | 2011-09-29 14:13:05 +0200 (Thu, 29 Sep 2011) |
+ 12 lines 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
+ ........ ................
+
+2011-09-29 12:03 +0000 [r338377-338415] Olle Johansson <oej@edvina.net>
+
+ * cdr/cdr_pgsql.c, CHANGES: Add CLI command "cdr show pgsql status"
+ based on "cdr mysql status" Review:
+ https://reviewboard.asterisk.org/r/923/ Thanks all for the code
+ reviews and feedback.
+
+ * res/res_agi.c: Just formatting.
+
+2011-09-28 22:38 +0000 [r338284-338324] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 338323 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r338323 | rmudgett | 2011-09-28 17:36:57 -0500
+ (Wed, 28 Sep 2011) | 12 lines Merged revisions 338322 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r338322 | rmudgett | 2011-09-28 17:35:52 -0500 (Wed, 28 Sep 2011)
+ | 5 lines 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, /: Merged revisions 338253 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r338253 | rmudgett | 2011-09-28 16:22:05 -0500
+ (Wed, 28 Sep 2011) | 14 lines Merged revisions 338235 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r338235 | rmudgett | 2011-09-28 16:17:45 -0500 (Wed, 28 Sep 2011)
+ | 7 lines 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 20:55 +0000 [r338229] Jason Parker <jparker@digium.com>
+
+ * build_tools/cflags.xml, channels/chan_usbradio.c,
+ build_tools/cflags-devmode.xml, agi/agi.xml, utils/utils.xml, /,
+ build_tools/embed_modules.xml, tests/test_db.c,
+ tests/test_netsock2.c: Merged revisions 338228 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r338228 | qwell | 2011-09-28 15:54:35 -0500
+ (Wed, 28 Sep 2011) | 9 lines Merged revisions 338227 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r338227 | qwell | 2011-09-28 15:52:47 -0500 (Wed, 28 Sep
+ 2011) | 1 line Add support levels to non-module sections of
+ menuselect (cflags, utils, etc). ........ ................
+
+2011-09-28 20:28 +0000 [r338226] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 338225 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r338225 | rmudgett | 2011-09-28 15:26:39 -0500
+ (Wed, 28 Sep 2011) | 12 lines Merged revisions 338224 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r338224 | rmudgett | 2011-09-28 15:24:41 -0500 (Wed, 28 Sep 2011)
+ | 5 lines 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 17:00 +0000 [r338187-338188] Terry Wilson <twilson@digium.com>
+
+ * CHANGES: Update CHANGES to reflect autopausebusy not being in
+ Asterisk 10
+
+ * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add
+ autopausebusy and autopauseunavail queue options Make it possible
+ to autopause on a busy or unavailable response from a device.
+ (closes issue ASTERISK-16112) Reported by: jlpedrosa Patches:
+ autopausebusy.txt by twilson Review:
+ https://reviewboard.asterisk.org/r/1399/
+
+2011-09-28 07:30 +0000 [r338136-338139] TransNexus OSP Development <support@transnexus.com>
+
+ * configure.ac: Updated for checking OSP Toolkit version 4.0.0.
+
+ * apps/app_osplookup.c: Updated for OSP Toolkit 4.0.0.
+
+2011-09-27 20:15 +0000 [r338086] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, apps/app_macro.c: Merged revisions 338085 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r338085 | pabelanger | 2011-09-27 16:13:14 -0400
+ (Tue, 27 Sep 2011) | 9 lines Merged revisions 338084 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r338084 | pabelanger | 2011-09-27 16:10:13 -0400 (Tue,
+ 27 Sep 2011) | 2 lines Upgrade app_macro to core ........
+ ................
+
+2011-09-27 12:45 +0000 [r338042] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Whitespace (red blobs) fixes
+
+2011-09-26 19:40 +0000 [r337975] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, main/pbx.c, cdr/cdr_sqlite3_custom.c, /,
+ include/asterisk/cel.h, cdr/cdr_syslog.c, tests/test_gosub.c,
+ 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_custom.c,
+ cdr/cdr_manager.c, apps/app_voicemail.c: Merged revisions 337974
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r337974 | rmudgett | 2011-09-26 14:35:23 -0500
+ (Mon, 26 Sep 2011) | 37 lines Merged revisions 337973 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337973 | rmudgett | 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011)
+ | 30 lines 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 19:20 +0000 [r337855-337910] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * /, contrib/init.d/rc.archlinux.asterisk: Merged revisions 337902
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r337902 | irroot | 2011-09-23 21:18:14 +0200
+ (Fri, 23 Sep 2011) | 10 lines Merged revisions 337898 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337898 | irroot | 2011-09-23 21:14:30 +0200 (Fri, 23 Sep 2011) |
+ 4 lines Spelling fix ........ ................
+
+ * /, apps/app_queue.c: Merged revisions 337840 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r337840 | irroot | 2011-09-23 10:39:22 +0200
+ (Fri, 23 Sep 2011) | 17 lines Merged revisions 337839 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337839 | irroot | 2011-09-23 10:34:03 +0200 (Fri, 23 Sep 2011) |
+ 11 lines 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 00:47 +0000 [r337776] Russell Bryant <russell@russellbryant.com>
+
+ * /, configs/res_pktccops.conf.sample: Merged revisions 337775 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r337775 | russell | 2011-09-22 19:45:35 -0500
+ (Thu, 22 Sep 2011) | 18 lines Merged revisions 337774 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337774 | russell | 2011-09-22 19:44:19 -0500 (Thu, 22 Sep 2011)
+ | 11 lines 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 21:42 +0000 [r337722] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 337721 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r337721 | rmudgett | 2011-09-22 16:37:41 -0500
+ (Thu, 22 Sep 2011) | 25 lines Merged revisions 337720 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337720 | rmudgett | 2011-09-22 16:29:46 -0500 (Thu, 22 Sep 2011)
+ | 18 lines 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 16:35 +0000 [r337600] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c, include/asterisk/event_defs.h,
+ main/security_events.c, channels/sip/security_events.c (added),
+ main/event.c, CHANGES, channels/sip/include/security_events.h
+ (added), channels/sip/include/sip.h,
+ include/asterisk/security_events_defs.h,
+ configs/logger.conf.sample: Merged revisions 337595,337597 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ........ r337595 | jrose | 2011-09-22 10:35:50 -0500 (Thu, 22 Sep
+ 2011) | 12 lines Generate Security events in chan_sip using new
+ Security Events Framework Security Events Framework was added in
+ 1.8 and support was added for AMI to generate events at that
+ time. This patch adds support for chan_sip to generate security
+ events. (closes issue ASTERISK-18264) Reported by: Michael L.
+ Young Patches: security_events_chan_sip_v4.patch (license #5026)
+ by Michael L. Young Review:
+ https://reviewboard.asterisk.org/r/1362/ ........ r337597 | jrose
+ | 2011-09-22 10:47:05 -0500 (Thu, 22 Sep 2011) | 10 lines Forgot
+ to svn add new files to r337595 Part of Generating security
+ events for chan_sip (issue ASTERISK-18264) Reported by: Michael
+ L. Young Patches: security_events_chan_sip_v4.patch (License
+ #5026) by Michael L. Young Reviewboard:
+ https://reviewboard.asterisk.org/r/1362/ ........
+
+2011-09-22 11:46 +0000 [r337432-337543] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * /, res/res_srtp.c: Merged revisions 337542 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r337542 | irroot | 2011-09-22 13:44:22 +0200
+ (Thu, 22 Sep 2011) | 14 lines Merged revisions 337541 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337541 | irroot | 2011-09-22 13:39:49 +0200 (Thu, 22 Sep 2011) |
+ 8 lines 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: Merged revisions 337487 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r337487 | irroot | 2011-09-22 11:26:26 +0200
+ (Thu, 22 Sep 2011) | 16 lines Merged revisions 337486 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337486 | irroot | 2011-09-22 11:22:26 +0200 (Thu, 22 Sep 2011) |
+ 10 lines 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, /: Merged revisions 337431 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r337431 | irroot | 2011-09-22 08:29:09 +0200
+ (Thu, 22 Sep 2011) | 25 lines Merged revisions 337430 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337430 | irroot | 2011-09-22 08:18:33 +0200 (Thu, 22 Sep 2011) |
+ 19 lines 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 21:26 +0000 [r337343-337385] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, apps/app_voicemail.c: More silly spacing changes ..... Merged
+ revisions 337353 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ..... Merged
+ revisions 337380 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, apps/app_voicemail.c: ................ ........ Dumb little
+ spacing fix. ........ Merged revisions 337344 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ Merged revisions 337345 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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 ........ Merged
+ revisions 337325 from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ Merged revisions 337342 from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+
+2011-09-21 11:21 +0000 [r337262-337283] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * /, configs/sip.conf.sample: Merged revisions 337263 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r337263 | irroot | 2011-09-21 13:15:48 +0200 (Wed, 21 Sep 2011) |
+ 1 line Whitespace fixup from SRTP patch ........
+
+ * /, apps/app_originate.c, CHANGES: Merged revisions 337261 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ........ r337261 | irroot | 2011-09-21 12:42:06 +0200 (Wed, 21
+ Sep 2011) | 10 lines Adds a timeout argument to app_originate the
+ default is 30s this will be used if the timout supplied is
+ invalid or no timeout is supplied. Contributed by: jacco (thank
+ you for the work) Review:
+ https://reviewboard.asterisk.org/r/1310/ ........
+
+2011-09-21 09:39 +0000 [r337179-337220] Olle Johansson <oej@edvina.net>
+
+ * main/pbx.c, /, CHANGES, configs/extensions.conf.sample: Merged
+ revisions 337219 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r337219 | oej | 2011-09-21 11:32:50 +0200 (Ons, 21 Sep 2011) | 13
+ lines Make ast_pbx_run() not default to s@default if extension is
+ not found Review: https://reviewboard.asterisk.org/r/1446/ This
+ is a bug - or architecture mistake - that has been in Asterisk
+ for a very long time. It was exposed by the AMI originate action
+ and possibly some other applications. Most channel drivers checks
+ if an extension exists BEFORE starting a pbx on an inbound call,
+ so most calls will not depend on this issue. Thanks everyone
+ involved in the review and on IRC and the mailing list for a
+ quick review and all the feedback. (closes issue ASTERISK-18578)
+ ........
+
+ * res/res_rtp_asterisk.c, /, configs/rtp.conf.sample, CHANGES:
+ Merged revisions 337178 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r337178 | oej | 2011-09-21 10:51:41 +0200 (Ons, 21 Sep 2011) | 14
+ lines Change strictrtp option to default to yes in the RTP module
+ Suggested by Kapejod on Facebook Review:
+ https://reviewboard.asterisk.org/r/1448/ (closes issue
+ ASTERISK-18587) Thanks for quick feedback to kpfleming and
+ Tilghman --Denna och nedanstående rader kommer inte med i
+ loggmeddelandet-- M CHANGES M configs/rtp.conf.sample M
+ res/res_rtp_asterisk.c ........
+
+2011-09-20 23:02 +0000 [r337124] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_dial.c, include/asterisk/app.h, apps/app_meetme.c,
+ apps/app_minivm.c, main/app.c, apps/app_confbridge.c,
+ apps/app_followme.c, apps/app_voicemail.c: Merged revisions
+ 337120 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r337120 | mjordan | 2011-09-20 17:49:36 -0500
+ (Tue, 20 Sep 2011) | 28 lines Merged revisions 337118 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011)
+ | 21 lines 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 22:54 +0000 [r337121-337123] Richard Mudgett <rmudgett@digium.com>
+
+ * /, funcs/func_strings.c: Merged revisions 337119 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r337119 | rmudgett | 2011-09-20 17:47:45 -0500 (Tue, 20 Sep 2011)
+ | 16 lines Fix crash with STRREPLACE function. The
+ ast_func_read() function calls the .read2 callback with the len
+ parameter set to zero indicating no size restrictions on the
+ supplied ast_str buffer. The value was used to dimension a local
+ starts[] array with the array subsequently used. * Reworked the
+ strreplace() function to perform the string replacement in a
+ straight forward manner. Eliminated the need for the starts[]
+ array. (closes issue ASTERISK-18545) Reported by: Federico Alves
+ Patches: jira_asterisk_18545_v10.patch (license #5621) patch
+ uploaded by rmudgett Tested by: rmudgett, Federico Alves ........
+
+ * /: Updated 10 merge property.
+
+ * /: Restore branch-10 merge properties.
+
+2011-09-20 22:29 +0000 [r337117] Leif Madsen <leif@leifmadsen.com>
+
+ * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 337115 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337115 | lmadsen | 2011-09-20 17:18:25 -0500 (Tue, 20 Sep 2011)
+ | 7 lines 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 21:05 +0000 [r337063] Kinsey Moore <kmoore@digium.com>
+
+ * main/pbx.c, /, tests/test_pbx.c: Merged revisions 337062 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r337062 | kmoore | 2011-09-20 16:05:01 -0500
+ (Tue, 20 Sep 2011) | 18 lines Merged revisions 337061 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337061 | kmoore | 2011-09-20 16:04:11 -0500 (Tue, 20 Sep 2011) |
+ 11 lines 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 19:13 +0000 [r336988-337009] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_ss7.c: Merged revisions 337008 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r337008 | rmudgett | 2011-09-20 14:12:24 -0500
+ (Tue, 20 Sep 2011) | 22 lines Merged revisions 337007 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337007 | rmudgett | 2011-09-20 14:10:30 -0500 (Tue, 20 Sep 2011)
+ | 15 lines 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: Merged revisions 336978 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r336978 | rmudgett | 2011-09-20 13:14:40 -0500
+ (Tue, 20 Sep 2011) | 28 lines Merged revisions 336977 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336977 | rmudgett | 2011-09-20 13:12:17 -0500 (Tue, 20 Sep 2011)
+ | 21 lines 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 16:56 +0000 [r336937] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * channels/sip/sdp_crypto.c, /, channels/chan_sip.c,
+ channels/sip/include/sdp_crypto.h, channels/sip/include/srtp.h,
+ configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h:
+ Merged revisions 336936 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r336936 | irroot | 2011-09-20 18:51:59 +0200 (Tue, 20 Sep 2011) |
+ 14 lines Allow Setting Auth Tag Bit length Based on invite or
+ config option Update the SIP SRTP API to allow use of 32 or 80
+ bit taglen. Curently only 80 bit is supported. The outgoing
+ invite will use the taglen of the incoming invite preventing
+ one-way audio. (Closes issue ASTERISK-17895) Review:
+ https://reviewboard.asterisk.org/r/1173/ ........
+
+2011-09-20 01:11 +0000 [r336879] Russell Bryant <russell@russellbryant.com>
+
+ * res/res_rtp_asterisk.c, /: Merged revisions 336878 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r336878 | russell | 2011-09-19 20:03:55 -0500
+ (Mon, 19 Sep 2011) | 43 lines Merged revisions 336877 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336877 | russell | 2011-09-19 19:56:20 -0500 (Mon, 19 Sep 2011)
+ | 36 lines 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 22:28 +0000 [r336837] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 336792 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r336792 | twilson | 2011-09-19 17:13:34 -0500
+ (Mon, 19 Sep 2011) | 9 lines Merged revisions 336791 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r336791 | twilson | 2011-09-19 17:07:58 -0500 (Mon, 19
+ Sep 2011) | 2 lines Don't interfere with T.38 reinvites This is
+ an update to the fix for ASTERISK-18340 and ASTERISK-17725
+ ........ ................
+
+2011-09-19 21:42 +0000 [r336735-336790] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, funcs/func_strings.c: Merged revisions 336789 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r336789 | tilghman | 2011-09-19 16:41:16 -0500 (Mon, 19 Sep 2011)
+ | 2 lines Ensure substring will not be found in the previous
+ match. ........
+
+ * Makefile, /, configure, include/asterisk/autoconfig.h.in,
+ main/Makefile, codecs/gsm/Makefile, configure.ac, Makefile.rules,
+ include/asterisk/optional_api.h: Merged revisions 336734 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r336734 | tilghman | 2011-09-19 15:29:40 -0500
+ (Mon, 19 Sep 2011) | 18 lines Merged revisions 336733 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336733 | tilghman | 2011-09-19 15:27:03 -0500 (Mon, 19 Sep 2011)
+ | 11 lines 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 20:23 +0000 [r336732] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_echo.c, apps/app_saycounted.c, apps/app_mp3.c,
+ apps/app_morsecode.c, res/res_musiconhold.c, apps/app_queue.c,
+ apps/app_mixmonitor.c: Merged revisions 336717 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r336717 | jrose | 2011-09-19 15:16:23 -0500
+ (Mon, 19 Sep 2011) | 14 lines Merged revisions 336716 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336716 | jrose | 2011-09-19 15:07:36 -0500 (Mon, 19 Sep 2011) |
+ 7 lines 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 19:03 +0000 [r336660-336662] Richard Mudgett <rmudgett@digium.com>
+
+ * apps/app_dial.c, /, UPGRADE-1.8.txt: Merged revisions 336659 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r336659 | rmudgett | 2011-09-19 13:51:19 -0500
+ (Mon, 19 Sep 2011) | 38 lines Merged revisions 336658 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336658 | rmudgett | 2011-09-19 13:46:40 -0500 (Mon, 19 Sep 2011)
+ | 31 lines 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 ........ ................
+
+ * /: Update merge 10 branch merge propterty.
+
+ * /: Restore 10 branch merge properties.
+
+2011-09-19 16:22 +0000 [r336600] Jason Parker <jparker@digium.com>
+
+ * cel/cel_odbc.c, configs/cel_odbc.conf.sample, sounds/Makefile:
+ Remove weird mergeinfo props that make merges annoying sometimes.
+
+2011-09-19 15:48 +0000 [r336574] Leif Madsen <leif@leifmadsen.com>
+
+ * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 336572
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ 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/
+ ........
+
+2011-09-19 15:36 +0000 [r336571] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 336570 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r336570 | rmudgett | 2011-09-19 10:32:00 -0500
+ (Mon, 19 Sep 2011) | 11 lines Merged revisions 336569 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336569 | rmudgett | 2011-09-19 10:25:34 -0500 (Mon, 19 Sep 2011)
+ | 4 lines Rework sig_pri_hangup() to be simpler and clearer. JIRA
+ AST-675 ........ ................
+
+2011-09-19 13:57 +0000 [r336505] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 336502 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r336502 | oej | 2011-09-19 15:38:53 +0200 (Mån,
+ 19 Sep 2011) | 12 lines Merged revisions 336501 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336501 | oej | 2011-09-19 15:33:50 +0200 (Mån, 19 Sep 2011) | 5
+ lines Add diversion header to a 302 redirect response if we have
+ diversion data (closes issue ASTERISK-18143) patch by oej
+ ........ ................
+
+2011-09-19 13:41 +0000 [r336503] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * /, channels/chan_h323.c: Merged revisions 336500 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r336500 | irroot | 2011-09-19 15:31:50 +0200
+ (Mon, 19 Sep 2011) | 19 lines Merged revisions 336499 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336499 | irroot | 2011-09-19 15:27:52 +0200 (Mon, 19 Sep 2011) |
+ 13 lines 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 12:20 +0000 [r336382-336453] Olle Johansson <oej@edvina.net>
+
+ * main/manager.c, /: Merged revisions 336441 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r336441 | oej | 2011-09-19 14:15:06 +0200 (Mån,
+ 19 Sep 2011) | 9 lines Merged revisions 336440 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336440 | oej | 2011-09-19 14:06:48 +0200 (Mån, 19 Sep 2011) | 2
+ lines Make sure manager_debug option is reset at reload ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 336381 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r336381 | oej | 2011-09-19 12:05:00 +0200 (Mån,
+ 19 Sep 2011) | 16 lines Merged revisions 336378 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336378 | oej | 2011-09-19 11:40:44 +0200 (Mån, 19 Sep 2011) | 9
+ lines 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 22:12 +0000 [r336315-336317] Terry Wilson <twilson@digium.com>
+
+ * /, funcs/func_frame_trace.c: Merged revisions 336316 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r336316 | twilson | 2011-09-16 17:11:39 -0500
+ (Fri, 16 Sep 2011) | 9 lines Merged revisions 336314 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r336314 | twilson | 2011-09-16 17:10:56 -0500 (Fri, 16
+ Sep 2011) | 2 lines Whitespace fix ........ ................
+
+ * /, funcs/func_frame_trace.c: Merged revisions 336313 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r336313 | twilson | 2011-09-16 17:07:00 -0500
+ (Fri, 16 Sep 2011) | 12 lines Merged revisions 336312 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336312 | twilson | 2011-09-16 17:04:25 -0500 (Fri, 16 Sep 2011)
+ | 5 lines 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 21:20 +0000 [r336311] Jonathan Rose <jrose@digium.com>
+
+ * main/channel.c, main/rtp_engine.c, /, channels/chan_sip.c,
+ include/asterisk/frame.h: Merged revisions 336307 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r336307 | jrose | 2011-09-16 16:09:20 -0500
+ (Fri, 16 Sep 2011) | 20 lines Merged revisions 336294 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336294 | jrose | 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) |
+ 13 lines 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 19:11 +0000 [r336236] Sean Bright <sean@malleable.com>
+
+ * /, UPGRADE-1.8.txt: Merged revisions 336235 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r336235 | seanbright | 2011-09-16 15:10:39 -0400
+ (Fri, 16 Sep 2011) | 9 lines Merged revisions 336234 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r336234 | seanbright | 2011-09-16 15:06:27 -0400 (Fri,
+ 16 Sep 2011) | 2 lines Make a note that inotify won't work with
+ an NFS mounted spooler directory. ........ ................
+
+2011-09-16 10:16 +0000 [r336095-336168] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * channels/chan_misdn.c, /: Merged revisions 336167 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r336167 | irroot | 2011-09-16 12:12:03 +0200
+ (Fri, 16 Sep 2011) | 22 lines Merged revisions 336166 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336166 | irroot | 2011-09-16 12:09:17 +0200 (Fri, 16 Sep 2011) |
+ 16 lines 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: Merged revisions 336094 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r336094 | irroot | 2011-09-15 17:54:46 +0200
+ (Thu, 15 Sep 2011) | 26 lines Merged revisions 336093 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336093 | irroot | 2011-09-15 17:46:21 +0200 (Thu, 15 Sep 2011) |
+ 20 lines 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/
+ ........ ................
+
+2011-09-15 15:19 +0000 [r336092] David Vossel <dvossel@digium.com>
+
+ * /, main/format_cap.c: Merged revisions 336091 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r336091 | dvossel | 2011-09-15 10:19:10 -0500 (Thu, 15 Sep 2011)
+ | 2 lines Removes some no-op code found in format_cap.c. ........
+
+2011-09-15 12:50 +0000 [r336043] Olle Johansson <oej@edvina.net>
+
+ * CREDITS, /, apps/app_meetme.c, CHANGES: Merged revisions 336042
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r336042 | oej | 2011-09-15 14:46:38 +0200 (Tor, 15 Sep 2011) | 12
+ lines Meetme: Introducing a new option "k" to kill a conference
+ if there's only a single member left. When using Meetme as a
+ modular call bridge from third party applications, it's handy to
+ make it behave like a normal call bridge. When the second to last
+ person exists, the last person will be kicked out of the
+ conference when this option is enabled. (closes issue
+ ASTERISK-18234) Review: https://reviewboard.asterisk.org/r/1376/
+ Patch by oej, sponsored by ClearIT, Solna, Sweden ........
+
+2011-09-15 08:40 +0000 [r335993] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * /, channels/chan_agent.c: Merged revisions 335991 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r335991 | irroot | 2011-09-15 10:29:12 +0200
+ (Thu, 15 Sep 2011) | 17 lines Merged revisions 335978 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335978 | irroot | 2011-09-15 10:15:22 +0200 (Thu, 15 Sep 2011) |
+ 11 lines 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 18:38 +0000 [r335853-335913] Richard Mudgett <rmudgett@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 335912 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r335912 | rmudgett | 2011-09-14 13:31:15 -0500
+ (Wed, 14 Sep 2011) | 20 lines Merged revisions 335911 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ 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) Reported
+ by: Michael Keuter ........ ................
+
+ * channels/chan_dahdi.c, /: Merged revisions 335852 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r335852 | rmudgett | 2011-09-14 11:00:37 -0500
+ (Wed, 14 Sep 2011) | 18 lines Merged revisions 335851 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ 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) 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 13:29 +0000 [r335792] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/manager.c, /: Merged revisions 335791 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r335791 | mnicholson | 2011-09-14 08:28:50 -0500
+ (Wed, 14 Sep 2011) | 11 lines Merged revisions 335790 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335790 | mnicholson | 2011-09-14 08:28:16 -0500 (Wed, 14 Sep
+ 2011) | 4 lines The tech and data members of
+ fast_originate_helper are not string fields. ASTERISK-17709
+ ........ ................
+
+2011-09-13 22:11 +0000 [r335722] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_directed_pickup.c: Merged revisions 335721 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r335721 | rmudgett | 2011-09-13 17:10:44 -0500
+ (Tue, 13 Sep 2011) | 9 lines Merged revisions 335720 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r335720 | rmudgett | 2011-09-13 17:10:15 -0500 (Tue, 13
+ Sep 2011) | 1 line Remove obsolete todo comment about
+ PICKUPRESULT. ........ ................
+
+2011-09-13 21:52 +0000 [r335719] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * main/dnsmgr.c: Additional updates for parsing dnsmgr.conf Review:
+ https://reviewboard.asterisk.org/r/1432/
+
+2011-09-13 21:40 +0000 [r335718] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * main/asterisk.c: do parse defaultlanguage from asterisk.conf Do
+ parse the option "defaultlanguage" from the [options] section of
+ asterisk.conf, as in the sample config file. Otherwise the
+ build-time default language (normally "en") is always the default
+ one. Review: https://reviewboard.asterisk.org/r/1342/
+ Signed-off-by: Tzafrir Cohen (License #5035)
+ <tzafrir.cohen@xorcom.com> Original-Commit:
+ http://svn.digium.com/svn/asterisk/branches/1.8@335716
+ Original-Commit:
+ http://svn.digium.com/svn/asterisk/branches/10@335717
+
+2011-09-13 18:56 +0000 [r335657] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, configure, configure.ac: Merged revisions 335656 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r335656 | tilghman | 2011-09-13 13:55:33 -0500
+ (Tue, 13 Sep 2011) | 11 lines Merged revisions 335655 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335655 | tilghman | 2011-09-13 13:52:38 -0500 (Tue, 13 Sep 2011)
+ | 4 lines 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 18:49 +0000 [r335654] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, main/manager.c, /: Merged revisions 335653 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r335653 | mnicholson | 2011-09-13 13:47:57 -0500
+ (Tue, 13 Sep 2011) | 12 lines Merged revisions 335618 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335618 | mnicholson | 2011-09-13 13:20:52 -0500 (Tue, 13 Sep
+ 2011) | 5 lines Don't limit the size of appdata for manager
+ originate actions. ASTERISK-17709 Patch by: tilghman (with
+ modifications) ........ ................
+
+2011-09-13 18:11 +0000 [r335555-335603] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * UPGRADE.txt, main/dsp.c: Clean up dsp.conf parsing Review:
+ https://reviewboard.asterisk.org/r/1434/
+
+ * UPGRADE.txt, cdr/cdr_csv.c: Clean up cdr.conf parsing for [csv]
+ section Review: https://reviewboard.asterisk.org/r/1427/
+
+ * main/dnsmgr.c, UPGRADE.txt: Clean up dnsmgr.conf parsing Review:
+ https://reviewboard.asterisk.org/r/1432/
+
+2011-09-13 07:35 +0000 [r335511] Russell Bryant <russell@russellbryant.com>
+
+ * include/asterisk/event.h, /, res/ais/evt.c, main/event.c: Merged
+ revisions 335510 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r335510 | russell | 2011-09-13 02:24:34 -0500
+ (Tue, 13 Sep 2011) | 22 lines Merged revisions 335497 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335497 | russell | 2011-09-13 02:11:36 -0500 (Tue, 13 Sep 2011)
+ | 15 lines 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 15:56 +0000 [r335435] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, /: Merged revisions 335434 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r335434 | mnicholson | 2011-09-12 10:55:48 -0500
+ (Mon, 12 Sep 2011) | 13 lines Merged revisions 335433 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335433 | mnicholson | 2011-09-12 10:54:41 -0500 (Mon, 12 Sep
+ 2011) | 6 lines Properly set caller_warning and callee_warning
+ before we try to use them. ASTERISK-18199 Patch by: elguero
+ Testing by: rtang ........ ................
+
+2011-09-12 14:33 +0000 [r335385] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Documentation updates
+
+2011-09-12 14:24 +0000 [r335354] Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 335346 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r335346 | kmoore | 2011-09-12 09:22:15 -0500
+ (Mon, 12 Sep 2011) | 17 lines Merged revisions 335341 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335341 | kmoore | 2011-09-12 09:21:17 -0500 (Mon, 12 Sep 2011) |
+ 10 lines 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) ........ ................
+
+2011-09-12 14:22 +0000 [r335324-335349] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Small documentation updates
+
+ * CREDITS, channels/chan_sip.c, include/asterisk/indications.h,
+ UPGRADE.txt, configs/sip.conf.sample, channels/sip/include/sip.h:
+ New sip.conf option for setting default tonezone for channel or
+ individual devices Review:
+ https://reviewboard.asterisk.org/r/1429/ (closes issue
+ ASTERISK-18497) Thanks to russellb for peer review.
+
+ * /, channels/chan_sip.c: Merged revisions 335323 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r335323 | oej | 2011-09-12 15:47:13 +0200 (Mån,
+ 12 Sep 2011) | 19 lines Merged revisions 335319 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335319 | oej | 2011-09-12 15:25:30 +0200 (Mån, 12 Sep 2011) | 12
+ lines 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 13:27 +0000 [r335322] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 335321 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r335321 | kmoore | 2011-09-12 08:27:04 -0500
+ (Mon, 12 Sep 2011) | 16 lines Merged revisions 335320 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335320 | kmoore | 2011-09-12 08:25:42 -0500 (Mon, 12 Sep 2011) |
+ 9 lines 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 11:15 +0000 [r335261] Stefan Schmidt <sst@sil.at>
+
+ * /, channels/chan_sip.c: Merged revisions 335260 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r335260 | schmidts | 2011-09-12 11:11:45 +0000
+ (Mon, 12 Sep 2011) | 12 lines Merged revisions 335259 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335259 | schmidts | 2011-09-12 11:09:19 +0000 (Mon, 12 Sep 2011)
+ | 6 lines 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/ ........
+ ................
+
+2011-09-12 03:10 +0000 [r335170-335212] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * UPGRADE.txt: Be more specific on which section has changed.
+
+ * main/cdr.c, UPGRADE.txt: Iterate though cdr.conf setting Review:
+ https://reviewboard.asterisk.org/r/1426/
+
+2011-09-11 17:09 +0000 [r335129] Terry Wilson <twilson@digium.com>
+
+ * configs/res_config_sqlite3.conf.sample (added),
+ res/res_config_sqlite3.c (added): Add SQLite 3 realtime support
+
+2011-09-09 16:28 +0000 [r335079] Matthew Jordan <mjordan@digium.com>
+
+ * channels/chan_unistim.c, apps/app_dial.c, main/pbx.c,
+ addons/chan_ooh323.c, channels/chan_sip.c,
+ 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: Merged revisions 335078 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r335078 | mjordan | 2011-09-09 11:27:01 -0500
+ (Fri, 09 Sep 2011) | 29 lines Merged revisions 335064 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011)
+ | 23 lines 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/ ........
+ ................
+
+2011-09-09 07:28 +0000 [r335015] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * funcs/func_dialplan.c, /, apps/app_readexten.c, CHANGES: Merged
+ revisions 335014 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r335014 | irroot | 2011-09-09 09:23:53 +0200 (Fri, 09 Sep 2011) |
+ 9 lines Move code for VALID_EXTEN from app_readexten to
+ func_dialplan Mark VALID_EXTEN deprecated. Review:
+ https://reviewboard.asterisk.org/r/1396/ ........
+
+2011-09-08 22:30 +0000 [r334955] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/logger.c: Merged revisions 334954 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r334954 | rmudgett | 2011-09-08 17:28:56 -0500
+ (Thu, 08 Sep 2011) | 17 lines Merged revisions 334953 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334953 | rmudgett | 2011-09-08 17:27:40 -0500 (Thu, 08 Sep 2011)
+ | 10 lines 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. ........
+ ................
+
+2011-09-08 13:36 +0000 [r334907] Jonathan Rose <jrose@digium.com>
+
+ * main/cdr.c, main/pbx.c: Removes colorful verb statements
+ erroneously commited with r332760
+
+2011-09-07 19:38 +0000 [r334845] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 334844 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r334844 | pabelanger | 2011-09-07 15:37:24 -0400
+ (Wed, 07 Sep 2011) | 11 lines Merged revisions 334843 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334843 | pabelanger | 2011-09-07 15:35:52 -0400 (Wed, 07 Sep
+ 2011) | 4 lines Cleanup chan_iax2.c log messages Review:
+ https://code.asterisk.org/code/cru/CR-AST-11 ........
+ ................
+
+2011-09-07 19:35 +0000 [r334842] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c: Merged revisions 334841 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r334841 | rmudgett | 2011-09-07 14:33:38 -0500
+ (Wed, 07 Sep 2011) | 17 lines Merged revisions 334840 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334840 | rmudgett | 2011-09-07 14:31:44 -0500 (Wed, 07 Sep 2011)
+ | 10 lines 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 15:37 +0000 [r334683-334792] Stefan Schmidt <sst@sil.at>
+
+ * /, main/features.c: Merged revisions 334747 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r334747 | schmidts | 2011-09-07 15:10:37 +0000
+ (Wed, 07 Sep 2011) | 9 lines Merged revisions 334682 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07
+ Sep 2011) | 3 lines 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. ........
+ ................
+
+ * main/features.c: clean up wrong merged stuff
+
+ * /, main/features.c: Merged revisions 334682 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07 Sep 2011)
+ | 3 lines 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. ........
+
+ * 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 08:17 +0000 [r334618-334623] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * /, CHANGES, apps/app_queue.c: Merged revisions 334621 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r334621 | alecdavis | 2011-09-07 20:14:50 +1200
+ (Wed, 07 Sep 2011) | 9 lines Merged revisions 334620 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r334620 | alecdavis | 2011-09-07 20:12:49 +1200 (Wed, 07
+ Sep 2011) | 2 lines peroid typo ........ ................
+
+ * main/logger.c: log Asterisk Version number, Build etc into each
+ log file Allow tracking of previous versions through log file
+ records to be tracked. Each time log file is created or opened,
+ log Asterisk Version, Buildinfo. etc. alecdavis (license 585)
+ Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1409/
+
+ * main/pbx.c, /: Merged revisions 334617 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r334617 | alecdavis | 2011-09-07 19:45:00 +1200
+ (Wed, 07 Sep 2011) | 17 lines Merged revisions 334616 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334616 | alecdavis | 2011-09-07 19:33:39 +1200 (Wed, 07 Sep
+ 2011) | 10 lines 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/ ........
+ ................
+
+2011-09-07 00:54 +0000 [r334574] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/frame.c, contrib/realtime/mysql/iaxfriends.sql,
+ contrib/realtime/postgresql/realtime.sql,
+ configs/sip.conf.sample, CHANGES,
+ contrib/realtime/mysql/sipfriends.sql: Implement the '!' negation
+ element to negate codecs directly in the allow keyword. This
+ permits the list of codecs to be specified in one configuration
+ line, instead of two or more, generally with the aim of either
+ allowing all codecs with the exception of a few or disallowing
+ most but permitting a few. Review:
+ https://reviewboard.asterisk.org/r/1411/
+
+2011-09-06 16:15 +0000 [r334519] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * /, apps/app_voicemail.c: Merged revisions 334455 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r334455 | irroot | 2011-09-06 15:58:56 +0200
+ (Tue, 06 Sep 2011) | 18 lines Merged revisions 334453 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334453 | irroot | 2011-09-06 15:48:03 +0200 (Tue, 06 Sep 2011) |
+ 13 lines 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/ ........
+ ................
+
+2011-09-06 16:08 +0000 [r334517] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * configs/iax.conf.sample, /, CHANGES, channels/chan_iax2.c: Merged
+ revisions 334514 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r334514 | pabelanger | 2011-09-06 11:47:59 -0400 (Tue, 06 Sep
+ 2011) | 6 lines authdebug is now disabled by default To enable
+ this functionaility again set authdebug = yes in iax.conf Review:
+ https://reviewboard.asterisk.org/r/1414/ ........
+
+2011-09-06 16:04 +0000 [r334472-334515] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * /, apps/app_voicemail.c: Revert r334472 due to properties going
+ missing
+
+ * /, apps/app_voicemail.c: Merged revisions 334455 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r334455 | irroot | 2011-09-06 15:58:56 +0200
+ (Tue, 06 Sep 2011) | 18 lines Merged revisions 334453 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334453 | irroot | 2011-09-06 15:48:03 +0200 (Tue, 06 Sep 2011) |
+ 13 lines 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 21:09 +0000 [r334304-334358] Richard Mudgett <rmudgett@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 334357 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r334357 | rmudgett | 2011-09-02 16:08:16 -0500
+ (Fri, 02 Sep 2011) | 26 lines Merged revisions 334355 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334355 | rmudgett | 2011-09-02 15:59:49 -0500 (Fri, 02 Sep 2011)
+ | 19 lines 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/ ........
+ ................
+
+ * /, include/asterisk/config.h, main/config.c: Merged revisions
+ 334297 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r334297 | rmudgett | 2011-09-02 12:15:08 -0500
+ (Fri, 02 Sep 2011) | 46 lines Merged revisions 334296 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334296 | rmudgett | 2011-09-02 12:10:58 -0500 (Fri, 02 Sep 2011)
+ | 39 lines 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 17:41 +0000 [r334231-334236] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * main/pbx.c, /: Merged revisions 334235 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r334235 | tilghman | 2011-09-01 12:39:32 -0500
+ (Thu, 01 Sep 2011) | 9 lines Merged revisions 334234 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r334234 | tilghman | 2011-09-01 12:38:33 -0500 (Thu, 01
+ Sep 2011) | 2 lines Remove 1.6 compatibility documentation from
+ 1.8, as it no longer applies. ........ ................
+
+ * res/res_config_odbc.c, /: Merged revisions 334230 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r334230 | tilghman | 2011-09-01 12:30:19 -0500
+ (Thu, 01 Sep 2011) | 25 lines Merged revisions 334229 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334229 | tilghman | 2011-09-01 12:28:09 -0500 (Thu, 01 Sep 2011)
+ | 18 lines 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 18:54 +0000 [r334158] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 334157 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r334157 | mnicholson | 2011-08-31 13:53:40 -0500
+ (Wed, 31 Aug 2011) | 11 lines Merged revisions 334156 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334156 | mnicholson | 2011-08-31 13:50:33 -0500 (Wed, 31 Aug
+ 2011) | 4 lines Disable T.38 when we get a invite with image
+ media port set to 0 ASTERISK-17678 ........ ................
+
+2011-08-31 18:11 +0000 [r334115] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_sip.c: Optimize chan_sip.c check_rtp_timeout()
+ function. * Make check_rtp_timeout() remember the values returned
+ by ast_rtp_instance_get_timeout(),
+ ast_rtp_instance_get_hold_timeout(), and
+ ast_rtp_instance_get_keepalive() instead of repeatedly calling
+ them. (closes issue ASTERISK-18319) Reported by: Rob Gagnon
+ Patches: issue-18319-trunk-r333066.diff (License #6159) patch
+ uploaded by Rob Gagnon Review:
+ https://reviewboard.asterisk.org/r/1377/
+
+2011-08-31 16:31 +0000 [r334067] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, res/res_fax.c: Merged revisions 334064 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r334064 | mnicholson | 2011-08-31 11:31:00 -0500 (Wed, 31 Aug
+ 2011) | 4 lines only alter the gateway_timeout when attching the
+ gateway to a channel ASTERISK-18219 ........
+
+2011-08-31 16:02 +0000 [r334011-334014] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 334013 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r334013 | rmudgett | 2011-08-31 11:00:49 -0500
+ (Wed, 31 Aug 2011) | 30 lines Merged revisions 334012 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334012 | rmudgett | 2011-08-31 10:57:12 -0500 (Wed, 31 Aug 2011)
+ | 23 lines 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: Merged revisions 334010
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r334010 | rmudgett | 2011-08-31 10:23:11 -0500
+ (Wed, 31 Aug 2011) | 50 lines Merged revisions 334009 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334009 | rmudgett | 2011-08-31 10:20:31 -0500 (Wed, 31 Aug 2011)
+ | 43 lines 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 15:20 +0000 [r334008] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 334007 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r334007 | kmoore | 2011-08-31 10:19:30 -0500
+ (Wed, 31 Aug 2011) | 14 lines Merged revisions 334006 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r334006 | kmoore | 2011-08-31 10:18:37 -0500 (Wed, 31 Aug 2011) |
+ 7 lines 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 22:16 +0000 [r333963] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c, /,
+ addons/ooh323c/src/ooCalls.h, addons/ooh323c/src/oochannels.c,
+ addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c: Merged
+ revisions 333961-333962 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r333961 | may | 2011-08-31 01:21:53 +0400 (Wed,
+ 31 Aug 2011) | 11 lines Merged revisions 333947 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r333947 | may | 2011-08-31 01:16:30 +0400 (Wed, 31 Aug 2011) | 5
+ lines cleanups in ACF/ARJ GK replies processing fixed long (24
+ sec) pause if acf/arj proccessed before ast_cond_wait called to
+ wait this ........ ................ r333962 | may | 2011-08-31
+ 01:53:42 +0400 (Wed, 31 Aug 2011) | 3 lines security fix. really
+ drop call if signalling addr is not same as socket addr
+ ................
+
+2011-08-30 14:03 +0000 [r333896] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, res/res_fax.c: Merged revisions 333895 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r333895 | mnicholson | 2011-08-30 09:01:31 -0500 (Tue, 30 Aug
+ 2011) | 6 lines Replaced FAXOPT(gwtimeout) with a second
+ parameter to FAXOPT(gateway). Patch by: irroot Review:
+ https://reviewboard.asterisk.org/r/1385/ ASTERISK-18219 ........
+
+2011-08-29 21:43 +0000 [r333838] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 333837 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r333837 | twilson | 2011-08-29 16:41:13 -0500
+ (Mon, 29 Aug 2011) | 22 lines Merged revisions 333836 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r333836 | twilson | 2011-08-29 16:38:31 -0500 (Mon, 29 Aug 2011)
+ | 15 lines 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 21:17 +0000 [r333789] Richard Mudgett <rmudgett@digium.com>
+
+ * /, include/asterisk/channel.h, addons/chan_mobile.c: Merged
+ revisions 333786 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r333786 | rmudgett | 2011-08-29 16:12:29 -0500
+ (Mon, 29 Aug 2011) | 13 lines Merged revisions 333784-333785 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r333784 | rmudgett | 2011-08-29 16:05:43 -0500 (Mon, 29 Aug 2011)
+ | 2 lines Fix deadlock potential of
+ chan_mobile.c:mbl_ast_hangup(). ........ r333785 | rmudgett |
+ 2011-08-29 16:06:16 -0500 (Mon, 29 Aug 2011) | 1 line Add some do
+ not hold locks notes to channel.h ........ ................
+
+2011-08-29 18:28 +0000 [r333736] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, res/res_fax_spandsp.c: Merged revisions 333716 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r333716 | mnicholson | 2011-08-29 13:22:58 -0500 (Mon, 29 Aug
+ 2011) | 5 lines It is possible for the gateway to be attached
+ when the channel is still negotiating T.38. This change handles
+ that case. ASTERISK-18329 ........
+
+2011-08-29 17:31 +0000 [r333689] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c, /, CHANGES: Merged revisions 333681 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r333681 | twilson | 2011-08-29 12:28:59 -0500 (Mon, 29 Aug 2011)
+ | 7 lines Use realtime text when it is negotiated This patch make
+ use of wirte_text() realtime text instead of send_text() if T.140
+ is in native formats. ASTERISK-17937 Review:
+ https://reviewboard.asterisk.org/r/1356/ ........
+
+2011-08-29 17:14 +0000 [r333632] Matthew Jordan <mjordan@digium.com>
+
+ * apps/app_voicemail.c: Merged revisions 333631 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r333631 | mjordan | 2011-08-29 12:12:55 -0500
+ (Mon, 29 Aug 2011) | 9 lines Merged revisions 333630 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r333630 | mjordan | 2011-08-29 12:11:15 -0500 (Mon, 29
+ Aug 2011) | 1 line Fixed improperly formatted TestEvent AMI
+ message in app_voicemail ........ ................
+
+2011-08-29 15:58 +0000 [r333571] Jonathan Rose <jrose@digium.com>
+
+ * /, res/res_jabber.c: Merged revisions 333570 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r333570 | jrose | 2011-08-29 10:56:56 -0500
+ (Mon, 29 Aug 2011) | 11 lines Merged revisions 333569 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r333569 | jrose | 2011-08-29 10:55:34 -0500 (Mon, 29 Aug 2011) |
+ 4 lines Accidental use of variable client->status instead of
+ client->state in from ASTERISK-18078 (issue ASTERISK-18078)
+ ........ ................
+
+2011-08-28 09:57 +0000 [r333509] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_vpb.cc: chan_vpb: remove unused variables (gcc4.6)
+ GCC 4.6 detects variables that get assined to, but never used
+ later. Also removes some remmed-out lines that become invalid.
+ (closes issue ASTERISK-18336) Signed-off-by: Tzafrir Cohen
+ (License #5035) <tzafrir.cohen@xorcom.com>,
+
+2011-08-26 16:38 +0000 [r333428] Jonathan Rose <jrose@digium.com>
+
+ * /, res/res_jabber.c: Merged revisions 333410 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r333410 | jrose | 2011-08-26 11:28:03 -0500
+ (Fri, 26 Aug 2011) | 19 lines Merged revisions 333378 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r333378 | jrose | 2011-08-26 11:19:07 -0500 (Fri, 26 Aug 2011) |
+ 13 lines [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 16:12 +0000 [r333371] Matthew Jordan <mjordan@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 333370 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r333370 | mjordan | 2011-08-26 10:58:37 -0500
+ (Fri, 26 Aug 2011) | 26 lines Merged revisions 333339 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r333339 | mjordan | 2011-08-26 08:36:36 -0500 (Fri, 26 Aug 2011)
+ | 20 lines 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 19:13 +0000 [r333276] Jonathan Rose <jrose@digium.com>
+
+ * /, res/res_jabber.c: Merged revisions 333266 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r333266 | jrose | 2011-08-25 14:00:05 -0500
+ (Thu, 25 Aug 2011) | 20 lines Merged revisions 333265 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r333265 | jrose | 2011-08-25 13:47:42 -0500 (Thu, 25 Aug 2011) |
+ 14 lines 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 19:01 +0000 [r333159-333269] Jason Parker <jparker@digium.com>
+
+ * Makefile, /: Merged revisions 333268 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r333268 | qwell | 2011-08-25 14:01:18 -0500
+ (Thu, 25 Aug 2011) | 9 lines Merged revisions 333267 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r333267 | qwell | 2011-08-25 14:00:55 -0500 (Thu, 25 Aug
+ 2011) | 2 lines Fix for DESTDIR spaces patch. ........
+ ................
+
+ * Makefile, build_tools/mkpkgconfig, /, configure, configure.ac,
+ makeopts.in, sounds/Makefile: Merged revisions 333203 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r333203 | qwell | 2011-08-25 10:29:56 -0500
+ (Thu, 25 Aug 2011) | 15 lines Merged revisions 333201 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ 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/ ........
+ ................
+
+ * channels/chan_local.c: Fix typo from r333070
+
+2011-08-24 16:52 +0000 [r333117] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, res/res_fax.c: Merged revisions 333115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r333115 | mnicholson | 2011-08-24 11:51:42 -0500 (Wed, 24 Aug
+ 2011) | 4 lines Changed the "timeout" option to "gwtimeout".
+ ASTERISK-18219 ........
+
+2011-08-24 09:17 +0000 [r333070-333075] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_local.c: Formatting changes - Removing some red
+ white space and adding some curly brackets.
+
+ * CHANGES: Add documentation for new manager event in chan_local
+ AST-17623
+
+ * channels/chan_local.c: Add manager event for local channel
+ semi-bridge (issue AST-17623) Review:
+ https://reviewboard.asterisk.org/r/1154
+
+2011-08-23 18:17 +0000 [r332881-333014] Richard Mudgett <rmudgett@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 333011 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r333011 | rmudgett | 2011-08-23 13:15:49 -0500
+ (Tue, 23 Aug 2011) | 19 lines Merged revisions 333010 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r333010 | rmudgett | 2011-08-23 13:14:01 -0500 (Tue, 23 Aug 2011)
+ | 12 lines 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 ........ ................
+
+ * /, main/config.c: Merged revisions 332940 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r332940 | rmudgett | 2011-08-22 16:23:40 -0500
+ (Mon, 22 Aug 2011) | 14 lines Merged revisions 332939 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332939 | rmudgett | 2011-08-22 16:22:24 -0500 (Mon, 22 Aug 2011)
+ | 7 lines Minor code optimizations. * Simplify
+ ast_category_browse() logic for easier understanding. * Remove
+ dead code in ast_variable_delete() and simplify some of its
+ logic. ........ ................
+
+ * /, apps/app_queue.c: Merged revisions 332875,332878 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r332875 | rmudgett | 2011-08-22 14:41:03 -0500
+ (Mon, 22 Aug 2011) | 1 line Fix merge property. ................
+ r332878 | rmudgett | 2011-08-22 14:46:25 -0500 (Mon, 22 Aug 2011)
+ | 25 lines Merged revisions 332874 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332874 | rmudgett | 2011-08-22 14:32:19 -0500 (Mon, 22 Aug 2011)
+ | 18 lines 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 19:56 +0000 [r332880] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, channels/chan_gtalk.c: Merged revisions 332877 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r332877 | pabelanger | 2011-08-22 15:43:33 -0400
+ (Mon, 22 Aug 2011) | 13 lines Merged revisions 332876 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332876 | pabelanger | 2011-08-22 15:41:24 -0400 (Mon, 22 Aug
+ 2011) | 6 lines Revert previous commit It seems google is still
+ making changes to the protocol. (issue ASTERISK-18301) ........
+ ................
+
+2011-08-22 19:52 +0000 [r332879] Richard Mudgett <rmudgett@digium.com>
+
+ * /: Fix merge 10 branch merge properties.
+
+2011-08-22 19:19 +0000 [r332844] Matthew Jordan <mjordan@digium.com>
+
+ * include/asterisk/test.h, main/manager.c, /, main/file.c,
+ main/test.c, main/app.c, configs/manager.conf.sample,
+ include/asterisk/manager.h, apps/app_voicemail.c: Merged
+ revisions 332817 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332817 | mjordan | 2011-08-22 13:15:51 -0500 (Mon, 22 Aug 2011)
+ | 4 lines 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 18:33 +0000 [r332762-332831] Richard Mudgett <rmudgett@digium.com>
+
+ * res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged
+ revisions 332830 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r332830 | rmudgett | 2011-08-22 13:32:09 -0500
+ (Mon, 22 Aug 2011) | 15 lines Merged revisions 332816 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332816 | rmudgett | 2011-08-22 13:14:59 -0500 (Mon, 22 Aug 2011)
+ | 8 lines 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: Merged revisions 332761 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r332761 | rmudgett | 2011-08-22 12:05:35 -0500
+ (Mon, 22 Aug 2011) | 22 lines Merged revisions 332759 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332759 | rmudgett | 2011-08-22 12:00:03 -0500 (Mon, 22 Aug 2011)
+ | 15 lines 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 ........ ................
+
+2011-08-22 17:05 +0000 [r332760] Jonathan Rose <jrose@digium.com>
+
+ * main/cdr.c, main/pbx.c, configs/cdr.conf.sample,
+ include/asterisk/cdr.h, CHANGES: Add option for logging congested
+ calls as CONGESTION instead of NO_ANSWER in CDR This patch adds a
+ CDR option to cdr.conf that will allow CDR files to log calls
+ ending with congestion in a way that is unique from other
+ unanswered calls. (closes issue ASTERISK-14842) Reported by: Alec
+ Davis Patches: cdr_congestion.diff.txt (License #5546) patch
+ uploaded by Alec Davis
+
+2011-08-22 16:31 +0000 [r332757] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, res/res_fax.c, include/asterisk/res_fax.h: Merged revisions
+ 332756 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r332756 | mnicholson | 2011-08-22 11:29:45 -0500 (Mon, 22 Aug
+ 2011) | 4 lines add a way to disable and/or modify the gateway
+ timeout ASTERISK-18219 ........
+
+2011-08-21 14:34 +0000 [r332701] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, channels/chan_gtalk.c: Merged revisions 332700 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r332700 | pabelanger | 2011-08-21 10:33:23 -0400
+ (Sun, 21 Aug 2011) | 12 lines Merged revisions 332699 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332699 | pabelanger | 2011-08-21 10:31:31 -0400 (Sun, 21 Aug
+ 2011) | 5 lines Fix outgoing calls in chan_gtalk (closes issue
+ ASTERISK-18301) Reported by: az1324 ........ ................
+
+2011-08-19 20:00 +0000 [r332655] Kinsey Moore <kmoore@digium.com>
+
+ * /, apps/app_confbridge.c: Merged revisions 332654 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r332654 | kmoore | 2011-08-19 14:59:34 -0500 (Fri, 19 Aug 2011) |
+ 8 lines Make CONFBRIDGE_INFO behave more nicely CONFBRIDGE_INFO
+ doesn't behave as well in edge cases as MEETME_INFO. With this
+ patch, CONFBRIDGE_INFO should behave in a much more reasonable
+ manner when presented with invalid conferences and keywords.
+ Review: https://reviewboard.asterisk.org/r/1359/ ........
+
+2011-08-19 17:24 +0000 [r332615] Richard Mudgett <rmudgett@digium.com>
+
+ * res/res_config_ldap.c: Fix infinite loop releasing the same
+ memory in ldap_loadentry(). * Fixed memory leak of vars in
+ ldap_loadentry(). * Fixed potential NULL ptr dereference of vars
+ in ldap_loadentry().
+
+2011-08-18 21:39 +0000 [r332561] Terry Wilson <twilson@digium.com>
+
+ * main/netsock2.c, /: Merged revisions 332560 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r332560 | twilson | 2011-08-18 16:34:04 -0500
+ (Thu, 18 Aug 2011) | 12 lines Merged revisions 332559 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332559 | twilson | 2011-08-18 16:26:01 -0500 (Thu, 18 Aug 2011)
+ | 5 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 ........ ................
+
+2011-08-18 19:30 +0000 [r332505] Kinsey Moore <kmoore@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 332504 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r332504 | kmoore | 2011-08-18 14:29:15 -0500
+ (Thu, 18 Aug 2011) | 15 lines Merged revisions 332503 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332503 | kmoore | 2011-08-18 14:28:00 -0500 (Thu, 18 Aug 2011) |
+ 8 lines 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 14:49 +0000 [r332388-332448] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * build_tools/cflags.xml, build_tools/cflags-devmode.xml, /: Merged
+ revisions 332447 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r332447 | tilghman | 2011-08-18 09:48:40 -0500
+ (Thu, 18 Aug 2011) | 9 lines Merged revisions 332446 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r332446 | tilghman | 2011-08-18 09:46:54 -0500 (Thu, 18
+ Aug 2011) | 2 lines Move BETTER_BACKTRACES out of development
+ mode, as it's useful when DEBUG_THREADS is enabled. ........
+ ................
+
+ * Makefile, agi/Makefile, utils/Makefile, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ Makefile.moddir_rules, makeopts.in, sounds/Makefile: Merged
+ revisions 332369 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r332369 | tilghman | 2011-08-17 14:24:59 -0500
+ (Wed, 17 Aug 2011) | 17 lines Merged revisions 332355 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332355 | tilghman | 2011-08-17 14:21:36 -0500 (Wed, 17 Aug 2011)
+ | 10 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/
+ ........ ................
+
+2011-08-17 18:31 +0000 [r332337] Terry Wilson <twilson@digium.com>
+
+ * /, res/res_timing_timerfd.c: Merged revisions 332321 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r332321 | twilson | 2011-08-17 13:09:49 -0500
+ (Wed, 17 Aug 2011) | 17 lines Merged revisions 332320 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332320 | twilson | 2011-08-17 12:35:27 -0500 (Wed, 17 Aug 2011)
+ | 10 lines 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-18142, ASTERISK-18166,
+ ASTERISK-18197, AST-486, AST-495, AST-507 and possibly others.
+ Review: https://reviewboard.asterisk.org/r/1361/ ........
+ ................
+
+2011-08-17 16:18 +0000 [r332270] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ channels/sig_pri.c: Merged revisions 332265 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r332265 | rmudgett | 2011-08-17 11:01:29 -0500
+ (Wed, 17 Aug 2011) | 33 lines Merged revisions 332264 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332264 | rmudgett | 2011-08-17 10:51:08 -0500 (Wed, 17 Aug 2011)
+ | 26 lines 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 ........ ................
+
+2011-08-16 20:15 +0000 [r332178] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * tests/test_substitution.c, tests/test_heap.c, /,
+ tests/test_expr.c, tests/test_ast_format_str_reduce.c,
+ tests/test_logger.c, tests/test_gosub.c, tests/test_app.c,
+ tests/test_dlinklists.c, tests/test_event.c, 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_acl.c, tests/test_locale.c, tests/test_utils.c,
+ tests/test_devicestate.c, tests/test_aoc.c, tests/test_astobj2.c,
+ tests/test_poll.c, tests/test_amihooks.c: Merged revisions 332177
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r332177 | pabelanger | 2011-08-16 16:11:49 -0400
+ (Tue, 16 Aug 2011) | 11 lines Merged revisions 332176 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332176 | pabelanger | 2011-08-16 16:10:13 -0400 (Tue, 16 Aug
+ 2011) | 4 lines Flag test modules as 'core' Review:
+ https://reviewboard.asterisk.org/r/1369/ ........
+ ................
+
+2011-08-16 17:53 +0000 [r332120] Jonathan Rose <jrose@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 332119 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r332119 | jrose | 2011-08-16 12:45:38 -0500
+ (Tue, 16 Aug 2011) | 23 lines Merged revisions 332118 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332118 | jrose | 2011-08-16 12:38:19 -0500 (Tue, 16 Aug 2011) |
+ 16 lines 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 17:23 +0000 [r332117] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/features.c, CHANGES, configs/features.conf.sample,
+ main/asterisk.c: Merged revisions 332101 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r332101 | rmudgett | 2011-08-16 12:17:28 -0500
+ (Tue, 16 Aug 2011) | 140 lines Merged revisions 332100 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332100 | rmudgett | 2011-08-16 11:31:36 -0500 (Tue, 16 Aug 2011)
+ | 133 lines 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 15:21 +0000 [r332028-332044] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/sip/include/sip.h: Merged revisions 332042 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ........ r332042 | mnicholson | 2011-08-16 10:20:48 -0500 (Tue,
+ 16 Aug 2011) | 2 lines fix a code comment AST-580 ........
+
+ * /, UPGRADE.txt, CHANGES: Merged revisions 332029 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r332029 | mnicholson | 2011-08-16 10:17:16 -0500 (Tue, 16 Aug
+ 2011) | 2 lines Moved notes about 'storesipcause' to UPGRADE.txt
+ from CHANGES AST-580 ........
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged
+ revisions 332027 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r332027 | mnicholson | 2011-08-16 10:08:40 -0500
+ (Tue, 16 Aug 2011) | 9 lines Merged revisions 332026 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r332026 | mnicholson | 2011-08-16 10:06:31 -0500 (Tue,
+ 16 Aug 2011) | 2 lines use DEFAULT_STORE_SIP_CAUSE to set the
+ default value for the 'storesipcause' option AST-580 ........
+ ................
+
+2011-08-16 14:47 +0000 [r332024] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_local.c: Formatting changes while working with
+ DTMF...
+
+2011-08-16 14:41 +0000 [r332023] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Merged
+ revisions 332022 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r332022 | mnicholson | 2011-08-16 09:40:37 -0500
+ (Tue, 16 Aug 2011) | 16 lines In 10 and trunk this option is
+ disabled by default. Merged revisions 332021 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r332021 | mnicholson | 2011-08-16 09:20:43 -0500 (Tue, 16 Aug
+ 2011) | 7 lines Added the 'storesipcause' option to sip.conf to
+ allow the user to disable the setting of HASH(SIP_CAUSE,<chan
+ name>) on the channel. Having chan_sip set HASH(SIP_CAUSE,<chan
+ name>) on the channel carries a significant performance penalty
+ because of the usage of the MASTER_CHANNEL() dialplan function.
+ AST-580 ........ ................
+
+2011-08-15 17:36 +0000 [r331957] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 331956 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r331956 | rmudgett | 2011-08-15 12:35:03 -0500
+ (Mon, 15 Aug 2011) | 20 lines Merged revisions 331955 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331955 | rmudgett | 2011-08-15 12:24:08 -0500 (Mon, 15 Aug 2011)
+ | 13 lines 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 15:24 +0000 [r331903] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * main/rtp_engine.c, /: Merged revisions 331894 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r331894 | pabelanger | 2011-08-15 11:22:45 -0400
+ (Mon, 15 Aug 2011) | 12 lines Merged revisions 331886 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331886 | pabelanger | 2011-08-15 11:21:16 -0400 (Mon, 15 Aug
+ 2011) | 5 lines Fix noisy message when briding channels (closes
+ issue ASTERISK-18270) Reported by: Federico Alves ........
+ ................
+
+2011-08-15 15:15 +0000 [r331869] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 331868 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r331868 | dvossel | 2011-08-15 10:14:13 -0500
+ (Mon, 15 Aug 2011) | 12 lines Merged revisions 331867 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331867 | dvossel | 2011-08-15 10:12:16 -0500 (Mon, 15 Aug 2011)
+ | 6 lines Fixes locking inversion issues present in the handling
+ of the sip REFER method. (closes issue ASTERISK-18082) Reported
+ by: James Van Vleet ........ ................
+
+2011-08-15 13:27 +0000 [r331830] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Formatting guideline fixes
+
+2011-08-12 19:06 +0000 [r331776] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 331775 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r331775 | mnicholson | 2011-08-12 14:03:31 -0500
+ (Fri, 12 Aug 2011) | 17 lines Merged revisions 331774 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331774 | mnicholson | 2011-08-12 14:01:27 -0500 (Fri, 12 Aug
+ 2011) | 11 lines 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 19:01 +0000 [r331773] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 331772 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r331772 | rmudgett | 2011-08-12 13:59:45 -0500
+ (Fri, 12 Aug 2011) | 15 lines Merged revisions 331771 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331771 | rmudgett | 2011-08-12 13:58:40 -0500 (Fri, 12 Aug 2011)
+ | 8 lines 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
+ ........ ................
+
+2011-08-12 18:03 +0000 [r331717] Jonathan Rose <jrose@digium.com>
+
+ * apps/app_dial.c, /, apps/app_meetme.c: Merged revisions 331644
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r331644 | jrose | 2011-08-12 11:18:57 -0500
+ (Fri, 12 Aug 2011) | 9 lines Merged revisions 331635 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r331635 | jrose | 2011-08-12 10:49:17 -0500 (Fri, 12 Aug
+ 2011) | 1 line Fixes 32bit compilation warnings brought on by
+ 331634 in app_dial and app_meetme ........ ................
+
+2011-08-12 17:56 +0000 [r331716] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 331715 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r331715 | rmudgett | 2011-08-12 12:54:47 -0500
+ (Fri, 12 Aug 2011) | 29 lines Merged revisions 331714 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331714 | rmudgett | 2011-08-12 12:47:57 -0500 (Fri, 12 Aug 2011)
+ | 22 lines 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 16:32 +0000 [r331660] Terry Wilson <twilson@digium.com>
+
+ * /, tests/test_netsock2.c: Merged revisions 331659 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r331659 | twilson | 2011-08-12 11:31:21 -0500
+ (Fri, 12 Aug 2011) | 11 lines Merged revisions 331658 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331658 | twilson | 2011-08-12 11:30:26 -0500 (Fri, 12 Aug 2011)
+ | 4 lines Fix netsock2 multiple zero-expansion test Remove
+ erroneous single bracket. ........ ................
+
+2011-08-12 16:22 +0000 [r331657] Kinsey Moore <kmoore@digium.com>
+
+ * /, main/logger.c: Merged revisions 331654 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r331654 | kmoore | 2011-08-12 11:21:37 -0500
+ (Fri, 12 Aug 2011) | 19 lines Merged revisions 331649 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331649 | kmoore | 2011-08-12 11:20:25 -0500 (Fri, 12 Aug 2011) |
+ 12 lines 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 ........ ................
+
+2011-08-11 21:55 +0000 [r331580] Jason Parker <jparker@digium.com>
+
+ * apps/app_dial.c, /, apps/app_meetme.c: Merged revisions 331579
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r331579 | qwell | 2011-08-11 16:54:54 -0500
+ (Thu, 11 Aug 2011) | 13 lines Merged revisions 331578 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331578 | qwell | 2011-08-11 16:46:39 -0500 (Thu, 11 Aug 2011) |
+ 6 lines 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 21:44 +0000 [r331577] Richard Mudgett <rmudgett@digium.com>
+
+ * /, funcs/func_shell.c: Merged revisions 331576 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r331576 | rmudgett | 2011-08-11 16:42:21 -0500
+ (Thu, 11 Aug 2011) | 16 lines Merged revisions 331575 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ 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 ........ ................
+
+2011-08-10 22:24 +0000 [r331519] Kinsey Moore <kmoore@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 331518 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r331518 | kmoore | 2011-08-10 17:23:49 -0500
+ (Wed, 10 Aug 2011) | 17 lines Merged revisions 331517 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331517 | kmoore | 2011-08-10 17:23:08 -0500 (Wed, 10 Aug 2011) |
+ 10 lines 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 20:51 +0000 [r331419-331463] Richard Mudgett <rmudgett@digium.com>
+
+ * /, main/logger.c: Merged revisions 331462 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r331462 | rmudgett | 2011-08-10 15:41:35 -0500
+ (Wed, 10 Aug 2011) | 37 lines Merged revisions 331461 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331461 | rmudgett | 2011-08-10 15:29:59 -0500 (Wed, 10 Aug 2011)
+ | 30 lines 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/ ........
+ ................
+
+ * /, main/features.c: Merged revisions 331420 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r331420 | rmudgett | 2011-08-10 14:07:53 -0500 (Wed, 10 Aug 2011)
+ | 2 lines Make sure feature_request_and_dial() initializes
+ outstate if passed in. ........
+
+ * /, main/features.c, CHANGES: Merged revisions 331418 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r331418 | rmudgett | 2011-08-10 13:25:08 -0500 (Wed, 10 Aug 2011)
+ | 6 lines Revert -r318141. It was a band-aid that only partially
+ fixed parking. A better fix is on reviewboard review 1358. (issue
+ ASTERISK-17374) ........
+
+2011-08-10 15:45 +0000 [r331371] Jonathan Rose <jrose@digium.com>
+
+ * channels/chan_sip.c, CHANGES: SIP display-name needed to be empty
+ for Avaya IP500 In order to address a compatability issue with
+ certain features on certain devices which rely on display name
+ content to change behavior, initreqprep in chan_sip.c has been
+ changed to no longer substitute cid_number into the display name
+ when cid_name isn't present. Instead, it will send no display
+ name in that case. (closes issue ASTERISK-16198) Reported by:
+ Walter Doekes Review: https://reviewboard.asterisk.org/r/1341/
+
+2011-08-10 13:49 +0000 [r331317] Kinsey Moore <kmoore@digium.com>
+
+ * main/manager.c, /: Merged revisions 331316 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r331316 | kmoore | 2011-08-10 08:48:41 -0500
+ (Wed, 10 Aug 2011) | 15 lines Merged revisions 331315 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331315 | kmoore | 2011-08-10 08:47:46 -0500 (Wed, 10 Aug 2011) |
+ 8 lines 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 23:17 +0000 [r331266] Richard Mudgett <rmudgett@digium.com>
+
+ * main/pbx.c, /, channels/chan_sip.c, main/features.c,
+ channels/chan_iax2.c, apps/app_parkandannounce.c: Merged
+ revisions 331265 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r331265 | rmudgett | 2011-08-09 18:12:49 -0500
+ (Tue, 09 Aug 2011) | 22 lines Merged revisions 331248 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331248 | rmudgett | 2011-08-09 17:12:59 -0500 (Tue, 09 Aug 2011)
+ | 15 lines 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 17:12 +0000 [r331202] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooGkClient.c, addons/chan_ooh323.c, /,
+ addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooq931.c:
+ Merged revisions 331147,331200 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r331147 | may | 2011-08-09 20:16:55 +0400 (Tue,
+ 09 Aug 2011) | 11 lines Merged revisions 331146 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331146 | may | 2011-08-09 20:13:09 +0400 (Tue, 09 Aug 2011) | 4
+ lines 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 ........
+ ................ r331200 | may | 2011-08-09 20:36:39 +0400 (Tue,
+ 09 Aug 2011) | 9 lines Setup IP proto version for call in GK mode
+ Added additional check for IP semantics before parse destination
+ by ast_parse_args due to it can parse numeric as IP. (closes
+ issue ASTERISK-18218) Reported by: slesru Patch:
+ ASTERISK-18218.patch ................
+
+2011-08-09 17:08 +0000 [r331201] Kinsey Moore <kmoore@digium.com>
+
+ * funcs/func_enum.c, UPGRADE.txt, main/enum.c: Allow ENUM query
+ functions to report lookup errors The ENUM dialplan functions do
+ not report DNS query errors properly. It is useful to
+ differentiate between failed query (e.g. non-existent domain) vs.
+ no data records of the appropriate type. This is required to make
+ overlapped dialing work. (closes issue ASTERISK-13769) Review:
+ https://reviewboard.asterisk.org/r/1355/ Patch-by: Timo Teras
+
+2011-08-09 16:02 +0000 [r331140-331144] Jason Parker <jparker@digium.com>
+
+ * /, doc/asterisk.8: Merged revisions 331143 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r331143 | qwell | 2011-08-09 10:59:54 -0500
+ (Tue, 09 Aug 2011) | 9 lines Merged revisions 331142 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r331142 | qwell | 2011-08-09 10:58:16 -0500 (Tue, 09 Aug
+ 2011) | 1 line Regenerate asterisk man page from sgml. ........
+ ................
+
+ * /, doc/asterisk.8, configs/asterisk.conf.sample,
+ configs/voicemail.conf.sample, doc/asterisk.sgml: Merged
+ revisions 331139 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r331139 | qwell | 2011-08-09 10:50:07 -0500
+ (Tue, 09 Aug 2011) | 19 lines Merged revisions 306999 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r306999 | lathama | 2011-02-08 14:22:35 -0600 (Tue, 08 Feb 2011)
+ | 12 lines 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 ........ ................
+
+ * doc/asterisk.8, configs/asterisk.conf.sample,
+ configs/voicemail.conf.sample, doc/asterisk.sgml: Merged
+ revisions 331138 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r331138 | qwell | 2011-08-09 10:47:20 -0500 (Tue, 09 Aug 2011) |
+ 1 line Revert merge of r306999, due to merge conflict. ........
+
+2011-08-08 22:59 +0000 [r331042-331098] Terry Wilson <twilson@digium.com>
+
+ * /, UPGRADE.txt, CHANGES, include/asterisk/manager.h: Merged
+ revisions 331097 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r331097 | twilson | 2011-08-08 17:59:01 -0500 (Mon, 08 Aug 2011)
+ | 5 lines Bump the AMI protocol version to 1.2 As a result of
+ converting Unlink events that were missed in the AMI 1.1 update
+ to Bridge events, the AMI protocol version is being incremented.
+ ........
+
+ * main/channel.c, /, CHANGES: Merged revisions 331041 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r331041 | twilson | 2011-08-08 16:12:51 -0500 (Mon, 08 Aug 2011)
+ | 6 lines Replace AMI Unlink events with Bridge events A previous
+ update converted some of the Link and Unlink events to Bridge
+ events, but a couple of Unlink events were missed. This patch
+ rectifies the situation. (closes issues ASTERISK-17455) ........
+
+2011-08-08 20:54 +0000 [r331000-331040] Kinsey Moore <kmoore@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 331039 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r331039 | kmoore | 2011-08-08 15:53:30 -0500
+ (Mon, 08 Aug 2011) | 18 lines Merged revisions 331038 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r331038 | kmoore | 2011-08-08 15:52:45 -0500 (Mon, 08 Aug 2011) |
+ 11 lines In-queue MOH stops after a periodic announcement If the
+ seek value is past the end of file when resuming G.722 MOH, MOH
+ will cease to function for the duration of the MOH session
+ through all starts and stops until saved state is cleared.
+ Adjusting the code to guarantee a single valid read (which is
+ already assumed) fixes the bug. (closes issue ASTERISK-18077)
+ Review: https://reviewboard.asterisk.org/r/1328/ Tested-by:
+ Jonathan Rose <jrose@digium.com> ........ ................
+
+ * configs/queues.conf.sample, apps/app_queue.c: Log queue member
+ name when state_interface is set for ADDMEMBER and REMOVEMEMBER
+ events app_queue logs the events ADDMEMBER and REMOVEMEMBER with
+ the agent field set to the interface value rather than the
+ membername value when a member is added with a state_interface
+ value set. However all other member related queue events are
+ logged with the membername when a state_interface is set. This
+ patch makes these fields optionally more consistent and correct.
+ (closes issue ASTERISK-14769) Review:
+ https://reviewboard.asterisk.org/r/1286 Patch-by: Jamuel Starkey
+ Tested-by: Kinsey Moore <kmoore@digium.com>
+
+ * apps/app_queue.c: app_queue: Add StateInterface to output of
+ "queue show" and "QueueStatus" This patch adds the
+ state_interface of the queue member struct to the output of
+ "queue show" (CLI command) and "QueueStatus" (AMI action) when
+ displaying relevant queue member information. For the AMI event
+ message the variable StateInterface has been added. (closes issue
+ ASTERISK-18071) Review: https://reviewboard.asterisk.org/r/1300/
+ Patch-by: Jamuel Starkey
+
+2011-08-05 15:57 +0000 [r330941] David Vossel <dvossel@digium.com>
+
+ * /, codecs/codec_resample.c: Merged revisions 330940 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r330940 | dvossel | 2011-08-05 10:53:49 -0500 (Fri, 05 Aug 2011)
+ | 2 lines The slin resampler is no longer dependent on an
+ external library, but the dependency was not removed correctly.
+ ........
+
+2011-08-05 08:47 +0000 [r330903] Alexandr Anikin <may@telecom-service.ru>
+
+ * addons/ooh323c/src/ooGkClient.c, /,
+ addons/ooh323c/src/ooCmdChannel.c: Merged revisions 330899 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r330899 | may | 2011-08-05 11:38:28 +0400 (Fri,
+ 05 Aug 2011) | 11 lines Merged revisions 330827 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r330827 | may | 2011-08-04 23:37:16 +0400 (Thu, 04 Aug 2011) | 4
+ lines change gk client behaivour on rrq/grq failures to setup
+ timers and next tries after timeout instead of complete failure
+ in the ooh323 stack ........ ................
+
+2011-08-04 20:53 +0000 [r330845] Terry Wilson <twilson@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 330844 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r330844 | twilson | 2011-08-04 15:51:23 -0500
+ (Thu, 04 Aug 2011) | 11 lines Merged revisions 330843 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r330843 | twilson | 2011-08-04 15:29:19 -0500 (Thu, 04 Aug 2011)
+ | 4 lines Make libsrtp instructions more explicit when linking
+ fails (closes issue ASTERISK-18139) ........ ................
+
+2011-08-03 15:16 +0000 [r330707-330764] Kinsey Moore <kmoore@digium.com>
+
+ * /, main/Makefile: Merged revisions 330763 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r330763 | kmoore | 2011-08-03 10:15:26 -0500
+ (Wed, 03 Aug 2011) | 16 lines Merged revisions 330762 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r330762 | kmoore | 2011-08-03 10:14:36 -0500 (Wed, 03 Aug 2011) |
+ 9 lines 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, /: Merged revisions
+ 330706 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r330706 | kmoore | 2011-08-03 08:39:06 -0500
+ (Wed, 03 Aug 2011) | 17 lines Merged revisions 330705 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r330705 | kmoore | 2011-08-03 08:38:17 -0500 (Wed, 03 Aug 2011) |
+ 10 lines 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 20:54 +0000 [r330650] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, res/res_jabber.c: Merged revisions 330649 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r330649 | kpfleming | 2011-08-02 15:52:44 -0500
+ (Tue, 02 Aug 2011) | 9 lines Merged revisions 330648 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r330648 | kpfleming | 2011-08-02 15:51:56 -0500 (Tue, 02
+ Aug 2011) | 2 lines Convert an error message to actually be
+ helpful. ........ ................
+
+2011-08-02 16:19 +0000 [r330577-330593] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 330586 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r330586 | dvossel | 2011-08-02 11:17:59 -0500
+ (Tue, 02 Aug 2011) | 15 lines Merged revisions 330581 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r330581 | dvossel | 2011-08-02 11:15:08 -0500 (Tue, 02 Aug 2011)
+ | 8 lines 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: Merged revisions 330579 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r330579 | dvossel | 2011-08-02 11:08:57 -0500
+ (Tue, 02 Aug 2011) | 9 lines Merged revisions 330578 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r330578 | dvossel | 2011-08-02 11:07:02 -0500 (Tue, 02
+ Aug 2011) | 2 lines Optimization to buffer initialization fix.
+ ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 330576 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r330576 | dvossel | 2011-08-02 10:55:36 -0500
+ (Tue, 02 Aug 2011) | 12 lines Merged revisions 330575 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r330575 | dvossel | 2011-08-02 10:53:21 -0500 (Tue, 02 Aug 2011)
+ | 5 lines Fixes uninitialized string buffer in log message.
+ (closes issue ASTERISK-17200) Reported by: lmadsen ........
+ ................
+
+2011-08-01 15:24 +0000 [r330435] Kinsey Moore <kmoore@digium.com>
+
+ * /, main/say.c: Merged revisions 330434 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r330434 | kmoore | 2011-08-01 10:23:29 -0500
+ (Mon, 01 Aug 2011) | 16 lines Merged revisions 330433 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r330433 | kmoore | 2011-08-01 10:22:10 -0500 (Mon, 01 Aug 2011) |
+ 9 lines 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 ........ ................
+
+2011-07-31 00:19 +0000 [r330370-330379] Richard Mudgett <rmudgett@digium.com>
+
+ * main/astobj2.c: Fixed compiler warning and a couple prototype
+ mismatches.
+
+ * main/channel.c, /: Merged revisions 330369 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r330369 | rmudgett | 2011-07-30 18:57:56 -0500
+ (Sat, 30 Jul 2011) | 11 lines Merged revisions 330368 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r330368 | rmudgett | 2011-07-30 18:56:29 -0500 (Sat, 30 Jul 2011)
+ | 4 lines Remove some redundant locking code in
+ ast_do_masquerade(). Also updated some comments. ........
+ ................
+
+2011-07-30 15:54 +0000 [r330313] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * main/channel.c, /: Merged revisions 330312 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r330312 | irroot | 2011-07-30 17:34:41 +0200
+ (Sat, 30 Jul 2011) | 15 lines Merged revisions 330311 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r330311 | irroot | 2011-07-30 17:25:16 +0200 (Sat, 30 Jul 2011) |
+ 9 lines 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 19:34 +0000 [r330273] Russell Bryant <russell@russellbryant.com>
+
+ * include/asterisk/astobj2.h, tests/test_astobj2.c,
+ channels/chan_iax2.c, main/astobj2.c: astobj2: Avoid using
+ temporary objects + ao2_find() with OBJ_POINTER. There is a
+ fairly common pattern making its way through the code base where
+ we put a temporary object on the stack so we can call ao2_find()
+ with OBJ_POINTER. The purpose is so that it can be passed into
+ the object hash function. However, this really seems like a hack
+ and potentially error prone. This patch is a first stab at
+ approach to avoid having to do that. It adds a new flag, OBJ_KEY,
+ which can be used instead of OBJ_POINTER in these situations.
+ Then, the hash function can know whether it was given an object
+ or some custom data to hash. The patch also changes some uses of
+ ao2_find() for iax2_user and iax2_peer objects to reflect how
+ OBJ_KEY would be used. So long, and thanks for all the fish.
+ Review: https://reviewboard.asterisk.org/r/1184/
+
+2011-07-29 17:20 +0000 [r330205-330221] Sean Bright <sean@malleable.com>
+
+ * /, formats/format_wav.c: Merged revisions 330217 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r330217 | seanbright | 2011-07-29 13:19:42 -0400
+ (Fri, 29 Jul 2011) | 9 lines Merged revisions 330213 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r330213 | seanbright | 2011-07-29 13:18:56 -0400 (Fri,
+ 29 Jul 2011) | 2 lines Correct the check for O_RDONLY. ........
+ ................
+
+ * /, formats/format_wav.c: Merged revisions 330204 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r330204 | seanbright | 2011-07-29 12:58:40 -0400
+ (Fri, 29 Jul 2011) | 9 lines Merged revisions 330203 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r330203 | seanbright | 2011-07-29 12:58:08 -0400 (Fri,
+ 29 Jul 2011) | 2 lines Only write to wav files that were opened
+ to be written to. ........ ................
+
+2011-07-29 05:27 +0000 [r330163] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, apps/app_confbridge.c: Merged revisions 330162 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r330162 | pabelanger | 2011-07-29 01:25:18 -0400 (Fri, 29 Jul
+ 2011) | 4 lines Fix typo pointed out on #asterisk Thanks notten
+ ........
+
+2011-07-28 21:46 +0000 [r330109] Terry Wilson <twilson@digium.com>
+
+ * /, main/term.c: Merged revisions 330108 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r330108 | twilson | 2011-07-28 16:44:31 -0500
+ (Thu, 28 Jul 2011) | 9 lines Merged revisions 330107 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r330107 | twilson | 2011-07-28 16:42:41 -0500 (Thu, 28
+ Jul 2011) | 2 lines Make console colors work for
+ TERM=xterm-256color ........ ................
+
+2011-07-28 17:16 +0000 [r330052] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 330051 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r330051 | rmudgett | 2011-07-28 12:10:37 -0500
+ (Thu, 28 Jul 2011) | 29 lines Merged revisions 330050 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ................ r330050 | rmudgett | 2011-07-28 12:04:24 -0500
+ (Thu, 28 Jul 2011) | 22 lines 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 15:46 +0000 [r329996] Jason Parker <jparker@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 329995 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r329995 | qwell | 2011-07-28 10:45:49 -0500
+ (Thu, 28 Jul 2011) | 13 lines Merged revisions 329994 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329994 | qwell | 2011-07-28 10:45:24 -0500 (Thu, 28 Jul 2011) |
+ 6 lines 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 15:30 +0000 [r329993] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, res/res_fax.c: Merged revisions 329992 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r329992 | mnicholson | 2011-07-28 10:28:21 -0500
+ (Thu, 28 Jul 2011) | 13 lines Merged revisions 329991 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329991 | mnicholson | 2011-07-28 10:26:56 -0500 (Thu, 28 Jul
+ 2011) | 6 lines 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 13:04 +0000 [r329897-329953] Sean Bright <sean@malleable.com>
+
+ * configs/confbridge.conf.sample, /: Merged revisions 329952 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ........ r329952 | seanbright | 2011-07-28 09:03:58 -0400 (Thu,
+ 28 Jul 2011) | 4 lines The default conf-usermenu says that '8'
+ can be used to leave the conference, so put that in the sample
+ user menu. '5' is supposed to extend the conference, but there
+ doesn't appear to be a concept of that in the menu actions.
+ ........
+
+ * /, apps/app_confbridge.c: Merged revisions 329950 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r329950 | seanbright | 2011-07-28 08:43:55 -0400 (Thu, 28 Jul
+ 2011) | 1 line Correct the spelling of 'conference.' ........
+
+ * /, channels/chan_sip.c: Merged revisions 329896 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r329896 | seanbright | 2011-07-28 07:35:27 -0400
+ (Thu, 28 Jul 2011) | 9 lines Merged revisions 329895 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r329895 | seanbright | 2011-07-28 07:34:33 -0400 (Thu,
+ 28 Jul 2011) | 2 lines Make the output of Externhost in 'sip show
+ settings' more consistent. ........ ................
+
+2011-07-27 21:22 +0000 [r329835-329856] Jonathan Rose <jrose@digium.com>
+
+ * main/cdr.c, main/pbx.c, include/asterisk/cdr.h, CHANGES:
+ reverting 329840 due to failing tests. Going to change this
+ feature to be purely optional.
+
+ * main/cdr.c, main/pbx.c, include/asterisk/cdr.h, CHANGES: Adds cdr
+ logging of calls resulting in CONGESTION Applies a patch made a
+ long time ago by alecdavis which adds a CDR feature for logging
+ calls that failed due to congestion. (closes issue #15907)
+ Reported by: alecdavis Patches: cdr_congestion.diff.txt uploaded
+ by alecdavis (license #5546) Review:
+ https://reviewboard.asterisk.org/r/454/
+
+2011-07-27 19:19 +0000 [r329775] Sean Bright <sean@malleable.com>
+
+ * /, Makefile.moddir_rules: Merged revisions 329771 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r329771 | seanbright | 2011-07-27 15:18:47 -0400
+ (Wed, 27 Jul 2011) | 15 lines Merged revisions 329767 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329767 | seanbright | 2011-07-27 15:17:46 -0400 (Wed, 27 Jul
+ 2011) | 8 lines Explicitly sort the module list so that the
+ menuselect lists are sorted. (closes ASTERISK-18141) Reported by:
+ Richard Miller Patches: sort-order.diff uploaded by seanbright
+ (License #5060) Tested by: leifmadsen ........ ................
+
+2011-07-27 18:12 +0000 [r329711] Jonathan Rose <jrose@digium.com>
+
+ * /, configs/indications.conf.sample: Merged revisions 329710 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r329710 | jrose | 2011-07-27 13:11:07 -0500
+ (Wed, 27 Jul 2011) | 14 lines Merged revisions 329709 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329709 | jrose | 2011-07-27 13:10:30 -0500 (Wed, 27 Jul 2011) |
+ 8 lines 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 15:26 +0000 [r329671] Sean Bright <sean@malleable.com>
+
+ * /, main/loader.c: Merged revisions 329670 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r329670 | seanbright | 2011-07-27 11:25:53 -0400 (Wed, 27 Jul
+ 2011) | 2 lines Sort the module list so that 'module show' is
+ alphabetical. ........
+
+2011-07-27 04:27 +0000 [r329615] Tilghman Lesher <tilghman@meg.abyt.es>
+
+ * /, cdr/cdr_odbc.c: Merged revisions 329614 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r329614 | tilghman | 2011-07-26 23:25:26 -0500
+ (Tue, 26 Jul 2011) | 13 lines Merged revisions 329613 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329613 | tilghman | 2011-07-26 23:23:46 -0500 (Tue, 26 Jul 2011)
+ | 6 lines 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 14:27 +0000 [r329530-329564] Jonathan Rose <jrose@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 329538 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r329538 | jrose | 2011-07-26 09:19:34 -0500
+ (Tue, 26 Jul 2011) | 11 lines Merged revisions 329529 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329529 | jrose | 2011-07-26 09:04:55 -0500 (Tue, 26 Jul 2011) |
+ 5 lines 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. ........ ................
+
+ * include/asterisk/app.h, /, configs/voicemail.conf.sample,
+ main/app.c, apps/app_voicemail.c: Merged revisions 329528 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r329528 | jrose | 2011-07-26 08:52:34 -0500
+ (Tue, 26 Jul 2011) | 24 lines Merged revisions 329527 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) |
+ 17 lines 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 19:57 +0000 [r329473] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, main/enum.c: Merged revisions 329472 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r329472 | pabelanger | 2011-07-25 15:55:33 -0400
+ (Mon, 25 Jul 2011) | 9 lines Merged revisions 329471 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r329471 | pabelanger | 2011-07-25 15:49:40 -0400 (Mon,
+ 25 Jul 2011) | 2 lines Decrease verbose messages to debug, to
+ help clean up CLI. ........ ................
+
+2011-07-25 14:07 +0000 [r329391-329432] Gregory Nietsky <gregory@distrotech.co.za>
+
+ * include/asterisk/dsp.h, main/dsp.c: dsp_process was enhanced to
+ work with alaw and ulaw in addition to slin. noticed that some
+ functions could be refactored here it is. Reported by: irroot
+ Tested by: irroot, mnicholson Review:
+ https://reviewboard.asterisk.org/r/1304/
+
+ * channels/chan_sip.c, channels/sip/include/sip.h: Remove
+ lastmsgssent from sip it has not been working since 1.6 Clean up
+ the return values to be consistant not currently used Add doxygen
+ returns MWI Event is sent on Register (closes issue
+ ASTERISK-17866) Reported by: one47 Tested by: irroot, mvanbaak
+ Review: https://reviewboard.asterisk.org/r/1172/
+
+2011-07-22 21:15 +0000 [r329332-329335] Richard Mudgett <rmudgett@digium.com>
+
+ * main/pbx.c, /: Merged revisions 329334 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10 ........
+ r329334 | rmudgett | 2011-07-22 16:14:22 -0500 (Fri, 22 Jul 2011)
+ | 1 line Make use less redundant loop construct for iterating
+ over hints. ........
+
+ * main/pbx.c, /: Merged revisions 329331 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r329331 | rmudgett | 2011-07-22 15:43:07 -0500
+ (Fri, 22 Jul 2011) | 55 lines Merged revisions 329299 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329299 | rmudgett | 2011-07-22 10:44:58 -0500 (Fri, 22 Jul 2011)
+ | 48 lines 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/
+ ........ ................
+
+2011-07-21 20:26 +0000 [r329258] Russell Bryant <russell@russellbryant.com>
+
+ * channels/chan_dahdi.c, /, main/features.c,
+ include/asterisk/netsock2.h, CHANGES, channels/sig_pri.c,
+ include/asterisk/rtp_engine.h: Merged revisions 329257 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ........ r329257 | russell | 2011-07-21 15:22:36 -0500 (Thu, 21
+ Jul 2011) | 2 lines s/1.10/10.0/ ........
+
+2011-07-21 18:06 +0000 [r329146-329205] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
+ revisions 329204 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r329204 | rmudgett | 2011-07-21 13:05:18 -0500
+ (Thu, 21 Jul 2011) | 13 lines Merged revisions 329203 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329203 | rmudgett | 2011-07-21 13:04:09 -0500 (Thu, 21 Jul 2011)
+ | 6 lines 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: Merged revisions 329200 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r329200 | rmudgett | 2011-07-21 12:32:02 -0500
+ (Thu, 21 Jul 2011) | 24 lines Merged revisions 329199 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329199 | rmudgett | 2011-07-21 12:30:57 -0500 (Thu, 21 Jul 2011)
+ | 17 lines 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: Merged revisions 329145 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/10
+ ................ r329145 | rmudgett | 2011-07-21 11:52:17 -0500
+ (Thu, 21 Jul 2011) | 16 lines Merged revisions 329144 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329144 | rmudgett | 2011-07-21 11:46:21 -0500 (Thu, 21 Jul 2011)
+ | 9 lines 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) ........
+ ................
+
+2011-07-21 16:22 +0000 [r329106-329130] Jason Parker <jparker@digium.com>
+
+ * UPGRADE-1.10.txt (removed), UPGRADE-10.txt (added), UPGRADE.txt:
+ Fix UPGRADE.txt files for Asterisk 10.
+
+ * /: Remove another 2.0 property.
+
+2011-07-21 16:05 +0000 [r329105] Russell Bryant <russell@russellbryant.com>
+
+ * /: Fix merge properties to reflect Asterisk 10 branch
+
projects / thirdparty / asterisk.git / commitdiff
