[svn-commits] lmadsen: tag 1.4.33-rc1 r266647 - /tags/1.4.33-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Jun 1 10:59:16 CDT 2010
Author: lmadsen
Date: Tue Jun 1 10:59:14 2010
New Revision: 266647
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=266647
Log:
Importing files for 1.4.33-rc1 release.
Added:
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tags/1.4.33-rc1/.version (with props)
tags/1.4.33-rc1/ChangeLog (with props)
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+2010-06-01 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.33-rc1 Released.
+
+2010-06-01 15:17 +0000 [r266585] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c: Prevent CLI prompt from distorting output of
+ lines shorter than the prompt. Uses the VT100 method of clearing
+ the line from the cursor position to the end of the line: Esc-0K
+ (closes issue #17160) Reported by: coolmig Patches:
+ 20100531__issue17160.diff.txt uploaded by tilghman (license 14)
+ Tested by: coolmig
+
+2010-06-01 14:57 +0000 [r266579-266580] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * channels/chan_sip.c: Fix formatting issue with previous patch.
+
+ * channels/chan_sip.c: Missing fallback to audio fax feature when
+ T.38 re-INVITE failed When a T.38 re-INVITE failed with an 488 or
+ 606 answer, we should fallback to audio fax by send a
+ re-re-INVITE without T.38. The function is backported from 1.6
+ asterisk. (closes issue #16795) Reported by: vrban (closes issue
+ #16692) Reported by: vrban Patches:
+ t38_fallback_to_audio_v3.patch uploaded by vrban (license 756)
+ Tested by: lmadsen, vrban, haggard
+ https://reviewboard.asterisk.org/r/514/
+
+2010-05-30 04:43 +0000 [r266437] Tilghman Lesher <tlesher at digium.com>
+
+ * contrib/init.d/rc.debian.asterisk: Reverting patch and reopening
+ issue #16784, as patch breaks color display.
+
+2010-05-26 21:11 +0000 [r266142] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c, main/logger.c: Use sigaction for signals which
+ should persist past the initial trigger, not signal. If you call
+ signal() in a Solaris signal handler, instead of just resetting
+ the signal handler, it causes the signal to refire, because the
+ signal is not marked as handled prior to the signal handler being
+ called. This effectively causes Solaris to immediately exceed the
+ threadstack in recursive signal handlers and crash. (closes issue
+ #17000) Reported by: rmcgilvr Patches:
+ 20100526__issue17000.diff.txt uploaded by tilghman (license 14)
+ Tested by: rmcgilvr
+
+2010-05-26 20:33 +0000 [r266140] David Vossel <dvossel at digium.com>
+
+ * channels/chan_dahdi.c: add dahdi_func_write to zap_tech structure
+ This was supposed to be committed with r263292, the back-port of
+ teh DAHDI buffer policy dial string option
+
+2010-05-26 18:21 +0000 [r266004] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Make AgentComplete message more consistent. At
+ times, the "Member" field was not specified during the event.
+ It's there now. (closes issue #15638) Reported by: elbriga
+ Patches: patchAppQueueAgentComplete.diff uploaded by elbriga
+ (license 482)
+
+2010-05-26 16:21 +0000 [r265910] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_config_pgsql.c: Not finding rows in the DB does not rise
+ to the level of a warning. (closes issue #17062) Reported by:
+ drookie Patches: 20100525__issue17062.diff.txt uploaded by
+ tilghman (license 14)
+
+2010-05-25 17:11 +0000 [r265613] David Vossel <dvossel at digium.com>
+
+ * channels/chan_dahdi.c: fixes build issue with zaptel (closes
+ issue #17394) Reported by: aragon Patches: half_buffer_fix.diff
+ uploaded by dvossel (license 671) Tested by: aragon
+
+2010-05-25 16:48 +0000 [r265610] Matthew Nicholson <mnicholson at digium.com>
+
+ * apps/app_queue.c: Don't mark the cdr records of unanswered queue
+ calls with "NOANSWER". This restores the behavior prior to
+ r258670. (closes issue #17334) Reported by: jvandal Patches:
+ queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested
+ by: aragon, jvandal
+
+2010-05-25 13:33 +0000 [r265570] Terry Wilson <twilson at digium.com>
+
+ * include/asterisk/options.h, main/asterisk.c, Makefile,
+ doc/manager.txt, main/manager.c: Merged revisions 265320,265467
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r265320 | twilson | 2010-05-24 14:06:40 -0500 (Mon, 24
+ May 2010) | 14 lines Add the FullyBooted AMI event It is possible
+ to connect to the manager interface before all Asterisk modules
+ are loaded. To ensure that an application does not send AMI
+ actions that might require a module that has not yet loaded, the
+ application can listen for the FullyBooted manager event. It will
+ be sent upon connection if all modules have been loaded, or as
+ soon as loading is complete. The event: Event: FullyBooted
+ Privilege: system,all Status: Fully Booted Review:
+ https://reviewboard.asterisk.org/r/639/ ........ r265467 |
+ twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line
+ Merge the rest of the FullyBooted patch ........
+
+2010-05-24 19:37 +0000 [r265365] David Vossel <dvossel at digium.com>
+
+ * main/channel.c: fixes segfault when using generic plc
+
+2010-05-21 20:59 +0000 [r264996-265089] Mark Michelson <mmichelson at digium.com>
+
+ * include/asterisk/file.h, apps/app_queue.c: Don't hang up on a
+ queue caller if the file we attempt to play does not exist. This
+ also fixes a documentation mistake in file.h that made my
+ original attempt to correct this problem not work correctly.
+ (closes issue #17061) Reported by: RoadKill
+
+ * include/asterisk/channel.h: Fix grammatical error in comment.
+
+ * main/channel.c, main/autoservice.c, include/asterisk/channel.h:
+ Allow ast_safe_sleep to defer specific frames until after the
+ sleep has concluded. From reviewboard Background: A Digium
+ customer discovered a somewhat odd bug. The setup is that parties
+ A and B are bridged, and party A places party B on hold. While
+ party B is listening to hold music, he mashes a bunch of DTMF.
+ Party A takes party B off hold while this is happening, but party
+ B continues to hear hold music. I could reproduce this about 1 in
+ 5 times. The issue: When DTMF features are enabled and a user
+ presses keys, the channel that the DTMF is streamed to is placed
+ in an ast_safe_sleep for 100 ms, the duration of the emulated
+ tone. If an AST_CONTROL_UNHOLD frame is read from the channel
+ during the sleep, the frame is dropped. Thus the unhold
+ indication is never made to the channel that was originally
+ placed on hold. The fix: Originally, I discussed with Kevin
+ possible ways of fixing the specific problem reported. However,
+ we determined that the same type of problem could happen in other
+ situations where ast_safe_sleep() is used. Using autoservice as a
+ model, I modified ast_safe_sleep_conditional() to defer specific
+ frame types so they can be re-queued once the sleep has finished.
+ I made a common function for determining if a frame should be
+ deferred so that there are not two identical switch blocks to
+ maintain. Review: https://reviewboard.asterisk.org/r/674/
+
+2010-05-20 23:23 +0000 [r264820] Richard Mudgett <rmudgett at digium.com>
+
+ * main/callerid.c: ast_callerid_parse() had a path that left name
+ uninitialized. Several callers of ast_callerid_parse() do not
+ initialize the name parameter before calling thus there is the
+ potential to use an uninitialized pointer.
+
+2010-05-20 15:59 +0000 [r264541] Mark Michelson <mmichelson at digium.com>
+
+ * include/asterisk/options.h, main/loader.c, main/channel.c,
+ include/asterisk/channel.h: 1.4 version of PLC fix. Analogous to
+ trunk revision 264452, but without the change to chan_sip since
+ it is not necessary in this branch.
+
+2010-05-19 20:01 +0000 [r264334] Matthew Nicholson <mnicholson at digium.com>
+
+ * apps/app_speech_utils.c: Set quieted flag when receiving a dtmf
+ tone during playback in speechbackground. (closes issue #16966)
+ Reported by: asackheim
+
+2010-05-19 17:41 +0000 [r264248] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/options.h, configure, configure.ac: Internal
+ timing is now on by default, if you're using DAHDI 2.3 or above.
+ The reason for ensuring DAHDI 2.3 or above is that this version
+ ensures that a timer is always available, whereas in previous
+ versions, it was possible for DAHDI to be loaded, but have no
+ drivers to actually generate timing. If internal_timing was
+ turned on in this circumstance, a complete lack of audio would
+ result. This is the reason why internal_timing was not on by
+ default. However, now that DAHDI ensures the availability of a
+ timer, there is no reason for this setting to be off (and in
+ fact, it solves a great many initial user problems). (closes
+ issue #15932) Reported by: dimas Patches:
+ 20100519__issue15932.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman
+
+2010-05-19 08:23 +0000 [r264056] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * configs/indications.conf.sample: fix incorrectly typed
+ indications for [nz] stutter and dialrecall (closes issue #17359)
+ Reported by: alecdavis Patches: bug17359.diff.txt uploaded by
+ alecdavis (license 585)
+
+2010-05-19 06:32 +0000 [r263949] Tilghman Lesher <tlesher at digium.com>
+
+ * main/dsp.c: Because progress is called multiple times, across
+ several frames, we must persist states when detecting multitone
+ sequences. (closes issue #16749) Reported by: dant Patches:
+ dsp.c-bug16749-1.patch uploaded by dant (license 670) Tested by:
+ dant
+
+2010-05-18 18:54 +0000 [r263769] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_directory.c: Modify directory name reading to be
+ interrupted with operator or pound escape. In the case of
+ accidentally entering the wrong first three letters for the
+ reading, users could be very frustrated if the name listing is
+ very long. This allows interrupting the reading by pressing 0 or
+ #. 0 will attempt to execute a configured operator (o) extension
+ and # will exit and proceed in the dialplan. ABE-2200
+
+2010-05-17 22:00 +0000 [r263637-263639] Mark Michelson <mmichelson at digium.com>
+
+ * main/devicestate.c: Fix logic error when checking for a devstate
+ provider. When using strsep, if one of the list of specified
+ separators is not found, it is the first parameter to strsep
+ which is now NULL, not the pointer returned by strsep. This issue
+ isn't especially severe in that the worst it is likely to do is
+ waste some cycles when a device with no '/' and no ':' is passed
+ to ast_device_state.
+
+ * main/pbx.c: Remove arbitrary size limitation for hints. (closes
+ issue #17257) Reported by: tim_ringenbach Patches:
+ hints_crash_fix.diff uploaded by tim ringenbach (license 540)
+
+2010-05-17 14:35 +0000 [r263374-263456] Leif Madsen <lmadsen at digium.com>
+
+ * main/http.c: Manager cookies are not compatible with RFC2109. The
+ Version field in the cookies we're setting contain quotes around
+ the version number which is not compatible with RFC2109 and
+ breaks some implementations. (closes issue #17231) Reported by:
+ ecarruda Patches: manager_rfc2109-trunk-v1.patch uploaded by
+ ecarruda (license 559) manager_rfc2109-1.6.2-v1.patch uploaded by
+ ecarruda (license 559) Tested by: ecarruda, russell
+
+ * sounds/Makefile: Update link to new version of core sounds. The
+ latest version of the core sounds files 1.4.19 now includes the
+ missing queue-minute sound file which is called by app_queue but
+ which has been missing. (closes issue #17123) Reported by:
+ n8ideas
+
+2010-05-17 13:01 +0000 [r263292] David Vossel <dvossel at digium.com>
+
+ * channels/chan_dahdi.c: backport of DAHDI buffer policy dial
+ string option
+
+2010-05-13 23:08 +0000 [r263112] Jeff Peeler <jpeeler at digium.com>
+
+ * main/channel.c, main/file.c: Fix internal timing not working with
+ Zaptel dahdi_compat.h was not being included in channel.c when
+ used with Zaptel and wasn't in file.c at all. (closes issue
+ #15250) Reported by: mneuhauser Patches: dahdi_compat.patch
+ uploaded by mneuhauser (license 425) Tested by: IgorG
+
+2010-05-12 17:00 +0000 [r262662] David Vossel <dvossel at digium.com>
+
+ * apps/app_meetme.c: fixes app_meetme dsp error We attempted to
+ detect silence after translating a frame from signed linear. This
+ caused a flooding of errors. To resolve this the code to detect
+ silence was moved before the translation. (closes issue #17133)
+ Reported by: jsdyer
+
+2010-05-11 19:55 +0000 [r262421] Jason Parker <jparker at digium.com>
+
+ * pbx/Makefile: Use a less silly method for modifying a
+ flex-generated file. The sed syntax that was used wasn't actually
+ valid, causing some versions to choke. This is the method that is
+ used in 1.6.x+ for similar changes. (closes issue #16696)
+ Reported by: bklang Patches: 16696-sedfix.diff uploaded by qwell
+ (license 4) Tested by: qwell
+
+2010-05-11 17:22 +0000 [r262321] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c, Makefile.rules: Fix issue #17302 a slightly
+ different way (mad props to Qwell)
+
+2010-05-10 16:34 +0000 [r262151] Tilghman Lesher <tlesher at digium.com>
+
+ * Makefile.rules: Allow compilation on Mac OS X 10.4 (Tiger)
+ (closes issue #17297) Reported by: jcovert Patches:
+ 20100506__issue17297.diff.txt uploaded by tilghman (license 14)
+ (closes issue #17302) Reported by: jcovert
+
+2010-05-06 20:10 +0000 [r261698-261735] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c: Only allow the operator key to be accepted
+ after leaving a voicemail. Or rather disallow the operator key
+ from being accepted when not offered, such as after finishing a
+ recording from within the mailbox options menu. ABE-2121 SWP-1267
+
+ * apps/app_voicemail.c: Revert 261698, code in trunk leads me to
+ believe unadvertised options are supported.
+
+ * apps/app_voicemail.c: Remove some hidden broken code in the
+ voicemail mailbox options menu. After finishing a recording from
+ within the mailbox options menu, pressing 0 exhibited strange
+ behavior with operator=yes turned on. Pressing 0 was not even
+ advertised as an option and the options from the vm-saveoper
+ prompt: "Press 1 to accept this recording. Otherwise, please
+ continue to hold" did not function correctly. While this of
+ course could be fixed, it didn't really seem to make sense even
+ if it was working properly. ABE-2121 SWP-1267
+
+2010-05-06 16:56 +0000 [r261608] Jason Parker <jparker at digium.com>
+
+ * sounds/Makefile: Use the versioned MOH tarballs, now that we have
+ them. This makes for more reproducibility. Prompted by a
+ discussion in #asterisk-dev
+
+2010-06-01 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.32 Released
+
+2010-05-26 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.32-rc2 Released
+
+2010-05-26 10:56 -0500 [r265891] Matt Nicholson <mnicholson at digium.com>
+
+ * Merged r265610 from 1.4:
+
+ Don't mark the cdr records of unanswered queue calls with "NOANSWER".
+ This restores the behavior prior to r258670.
+
+ (closes issue #17334)
+ Reported by: jvandal
+ Patches:
+ queue-cdr-fixes1.diff uploaded by mnicholson (license 96)
+ Tested by: aragon, jvandal
+
+2010-05-06 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.32-rc1 Released
+
+2010-05-05 16:42 +0000 [r261274] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * channels/chan_sip.c: Registration fix for SIP realtime. Make sure
+ realtime fields are not empty. (closes issue #17266) Reported by:
+ Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick
+ Lewis (license 657) Tested by: Nick_Lewis, sberney Review:
+ https://reviewboard.asterisk.org/r/643/
+
+2010-05-04 23:47 +0000 [r261093-261094] Tilghman Lesher <tlesher at digium.com>
+
+ * main/channel.c: Add a tiny corner case to the previous commit
+
+ * main/channel.c: Protect against overflow, when calculating how
+ long to wait for a frame. (closes issue #17128) Reported by:
+ under Patches: d.diff uploaded by under (license 914)
+
+2010-05-04 18:46 +0000 [r260923] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c: Voicemail transfer to operator should occur
+ immediately, not after main menu. There were two scenarios in the
+ advanced options that while using the operator=yes and review=yes
+ options, the transfer occurred only after exiting the main menu
+ (after sending a reply or leaving a message for an extension).
+ Now after the audio is processed for the reply or message the
+ transfer occurs immediately as expected. ABE-2107 ABE-2108
+
+2010-05-04 17:40 +0000 [r260887] tringenbach <tringenbach at localhost>:
+
+ * README-SERIOUSLY.bestpractices.txt: Fix FILTER() examples to work
+ in 1.4 Review: https://reviewboard.asterisk.org/r/644/
+
+2010-05-04 15:49 +0000 [r260801] Jason Parker <jparker at digium.com>
+
+ * build_tools/make_build_h: Fix fallout from removing from
+ configure script. Pointed out by philipp64 on #asterisk-dev
+
+2010-05-03 16:54 +0000 [r260661-260662] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * Makefile: Should have removed /usr/lib/ part. Thanks Qwell.
+
+ * Makefile: non-root make install PREFIX=/tmp fails. Prepend libdir
+ when executing mkpkgconfig allowing non-root installs to work.
+ (closes issue #17268) Reported by: pabelanger Patches:
+ issue17268.patch uploaded by pabelanger (license 224) Tested by:
+ pabelanger
+
+2010-05-03 14:57 +0000 [r260569] Leif Madsen <lmadsen at digium.com>
+
+ * doc/HOWTO_collect_debug_information.txt: Minor typo pointed out
+ by pabelanger on IRC.
+
+2010-04-30 22:22 +0000 [r260434] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c: Ensure channel state is not incorrectly
+ set in the case of a very early answer. The needringing bit was
+ being read in dahdi_read after answering thereby setting the
+ state to ringing from up. This clears needringing upon answering
+ so that is no longer possible. (closes issue #17067) Reported by:
+ tzafrir Patches: needringing.diff uploaded by tzafrir (license
+ 46)
+
+2010-04-30 20:08 +0000 [r260345] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_musiconhold.c: Fix potential crash from race condition
+ due to accessing channel data without the channel locked. In
+ res_musiconhold.c, there are several places where a channel's
+ stream's existence is checked prior to calling ast_closestream on
+ it. The issue here is that in several cases, the channel was not
+ locked while checking the stream. The result was that if two
+ threads checked the state of the channel's stream at
+ approximately the same time, then there could be a situation
+ where both threads attempt to call ast_closestream on the
+ channel's stream. The result here is that the refcount for the
+ stream would go below 0, resulting in a crash. I have added
+ proper channel locking to res_musiconhold.c to ensure that we do
+ not try to check chan->stream without the channel locked. A
+ Digium customer has been using this patch for several weeks and
+ has not had any crashes since applying the patch. ABE-2147
+
+2010-04-29 22:11 +0000 [r260195] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: DTMF CallerID detection problems. The code
+ handling DTMF CallerID drops digits on long CallerID numbers and
+ may timeout waiting for the first ring with shorter numbers. The
+ DTMF emulation mode was not turned off when processing DTMF
+ CallerID. When the emulation code gets behind in processing the
+ DTMF digits it can skip a digit. For shorter numbers, the timeout
+ may have been too short. I increased it from 2 seconds to 4
+ seconds. Four seconds is a typical time between rings for many
+ countries. (closes issue #16460) Reported by: sum Patches:
+ issue16460.patch uploaded by rmudgett (license 664)
+ issue16460_v1.6.2.patch uploaded by rmudgett (license 664) Tested
+ by: sum, rmudgett Review: https://reviewboard.asterisk.org/r/634/
+ JIRA SWP-562 JIRA AST-334 JIRA SWP-901
+
+2010-04-29 15:31 +0000 [r259858-260049] David Vossel <dvossel at digium.com>
+
+ * include/asterisk/audiohook.h, main/audiohook.c: Fixes crash in
+ audiohook_write_list The middle_frame in the audiohook_write_list
+ function was being freed if a audiohook manipulator returned a
+ failure. This is incorrect logic. This patch resolves this and
+ adds detailed descriptions of how this function should work and
+ why manipulator failures must be ignored. (closes issue #17052)
+ Reported by: dvossel Tested by: dvossel (closes issue #16196)
+ Reported by: atis Review: https://reviewboard.asterisk.org/r/623/
+
+ * main/channel.c, channels/chan_local.c: resolves deadlocks in
+ chan_local Issue_1. In the local_hangup() 3 locks must be held at
+ the same time... pvt, pvt->chan, and pvt->owner. Proper deadlock
+ avoidance is done when the channel to hangup is the outbound
+ chan_local channel, but when it is not the outbound channel we
+ have an issue... We attempt to do deadlock avoidance only on the
+ tech pvt, when both the tech pvt and the pvt->owner are locked
+ coming into that loop. By never giving up the pvt->owner channel
+ deadlock avoidance is not entirely possible. This patch resolves
+ that by doing deadlock avoidance on both the pvt->owner and the
+ pvt when trying to get the pvt->chan lock. Issue_2. ast_prod() is
+ used in ast_activate_generator() to queue a frame on the channel
+ and make the channel's read function get called. This function is
+ used in ast_activate_generator() while the channel is locked,
+ which mean's the channel will have a lock both from the generator
+ code and the frame_queue code by the time it gets to
+ chan_local.c's local_queue_frame code... local_queue_frame
+ contains some of the same crazy deadlock avoidance that
+ local_hangup requires, and this recursive lock prevents that
+ deadlock avoidance from happening correctly. This patch removes
+ ast_prod() from the channel lock so only one lock is held during
+ the local_queue_frame function. (closes issue #17185) Reported
+ by: schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel
+ (license 671) issue_17185_v2.diff uploaded by dvossel (license
+ 671) Tested by: schmoozecom, GameGamer43 Review:
+ https://reviewboard.asterisk.org/r/631/
+
+2010-04-28 21:07 +0000 [r259852] Leif Madsen <lmadsen at digium.com>
+
+ * config.guess: Update config.guess. Updating config.guess because
+ after installing Ubuntu Server 9.10 and running all the update
+ scripts, running ./configure would not continue because it was
+ unable to determine what kind of system I had. After updating
+ config.guess things started working again.
+
+2010-04-28 20:30 +0000 [r259748-259847] Jason Parker <jparker at digium.com>
+
+ * configure, configure.ac: Add AC_CONFIG_AUX_DIR to configure
+ script, so systems without install can use install-sh from our
+ source dir.
+
+ * makeopts.in: Missed this when removing $ID
+
+ * Makefile, configure, configure.ac: Remove usage of `id` since it
+ isn't useful and was causing breakge. Solaris `id` doesn't
+ support the -u argument. Instead of figuring out how to fix this
+ to work on Solaris, I decided to check why it was necessary and
+ where else it was used. It was only used in one place, and it
+ hasn't been needed for a very long time (I question whether it
+ was ever needed).
+
+2010-04-28 17:13 +0000 [r259664] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c: Do not play goodbye prompt after timeout of
+ message review. ABE-2124
+
+2010-04-27 21:53 +0000 [r259531] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: DAHDI "WARNING" message is confusing and
+ vague "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed
+ failed: Success" Changed the warning to "Failed to decode
+ CallerID on channel 'name'". The message before it is likely more
+ specific about why the CallerID decode failed. SWP-501 AST-283
+
+2010-04-27 21:48 +0000 [r259526] Leif Madsen <lmadsen at digium.com>
+
+ * sounds/Makefile: Update sounds files. * Add additional sounds
+ prompts for say_enumeration * Update the English conference
+ sounds prompts so they are better quality and all sound more
+ consistent * Clean up the core-sounds-XX.txt and
+ extra-sounds-XX.txt files to include all present sound files Both
+ core (en, fr, es) and extra (en, fr) sounds files have been
+ updated. (closes issue #16200) Reported by: murf (closes issue
+ #17137) Reported by: lmadsen
+
+2010-04-27 21:15 +0000 [r259352-259441] Jason Parker <jparker at digium.com>
+
+ * main/editline/configure, main/editline/configure.in: Add gar to
+ the check for AR for those silly OSes (Solaris) that don't have
+ ar.
+
+ * configure, configure.ac: Support the silly OSes that don't have
+ ar and strip. Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when
+ path isn't specified, and AC_PATH_TOOLS doesn't exist, we'll just
+ switch to AC_CHECK_TOOLS.
+
+2010-04-27 18:14 +0000 [r259270] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample:
+ hidecalleridname parameter in chan_dahdi.conf Issue #7321
+ implements a new chan_dahdi configuration option. However, a
+ change mentioned in the issue was never implemented. This is the
+ change that will allow the feature to work. I added a note to
+ chan_dahdi.conf.sample about the feature. (closes issue #17143)
+ Reported by: djensen99 Patches: diff.txt uploaded by djensen99
+ (license NA) (One line change) Tested by: djensen99
+
+2010-04-26 21:44 +0000 [r259018-259104] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c: Let compilation succeed warning-free when
+ DONT_OPTIMIZE is turned off.
+
+ * main/channel.c: Prevent Newchannel manager events for dummy
+ channels. No Newchannel manager event will be fired for channels
+ that are allocated to not match a registered technology type.
+ Thus bogus channels allocated solely for variable substitution or
+ CDR operations do not result in a Newchannel event. (closes issue
+ #16957) Reported by: atis Review:
+ https://reviewboard.asterisk.org/r/601
+
+2010-04-25 18:09 +0000 [r258775] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_monitor.c: When StopMonitor is called, ensure that it
+ will not be restarted by a channel event. (closes issue #16590)
+ Reported by: kkm Patches: resmonitor-16590-trunk.239289.diff
+ uploaded by kkm (license 888)
+
+2010-04-22 21:49 +0000 [r258670] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/cdr.c, main/channel.c, res/res_features.c: Fix broken CDR
+ behavior. This change allows a CDR record previously marked with
+ disposition ANSWERED to be set as BUSY or NO ANSWER. Additionally
+ this change partially reverts r235635 and does not set the
+ AST_CDR_FLAG_ORIGINATED flag on CDRs generated from ast_call().
+ To preserve proper CDR behavior, the AST_CDR_FLAG_DIALED flag is
+ now cleared from all brige CDRs in ast_bridge_call(). (closes
+ issue #16797) Reported by: VarnishedOtter Tested by: mnicholson
+
+2010-04-21 21:45 +0000 [r258432] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c: Fix looping forever when no input received
+ in certain voicemail menu scenarios. Specifically, prompting for
+ an extension (when leaving or forwarding a message) or when
+ prompting for a digit (when saving a message or changing
+ folders). ABE-2122 SWP-1268
+
+2010-04-20 16:16 +0000 [r257856-258029] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c: Play correct prompt when voicemail store
+ failure occurs after attempted forward. If a user's mailbox was
+ full and a message was attempted to be forwarded to said box,
+ warnings on the console would indicate failure. However, the
+ played prompt was that of success (vm-msgsaved). Now storage
+ failure is taken into account and the correct prompt
+ (vm-mailboxfull) is played when appropriate. ABE-2123 SWP-1262
+
+ * apps/app_voicemail.c: make app_voicemail compile with
+ IMAP_STORAGE
+
+2010-04-16 21:15 +0000 [r257686] Dwayne M. Hubbard <dwayne.hubbard at gmail.com>
+
+ * apps/app_mixmonitor.c: Make the mixmonitor thread process audio
+ frames faster Mantis issue 17078 reports MixMonitor recordings
+ have shorter durations than the call duration. This was because
+ the mixmonitor thread was not processing frames from the
+ audiohook fast enough. The mixmonitor thread would slowly fall
+ behind the most recent audio frame and when the channel hangs up,
+ the mixmonitor thread would exit without processing the same
+ number of frames as the channel; leaving the mixmonitor recording
+ shorter than actual call duration. This revision fixes this issue
+ by moving the ast_audiohook_trigger_wait() and the subsequent
+ audiohook.status check into the block where the
+ ast_audiohook_read_frame() function returns NULL. (closes issue
+ #17078) Reported by: geoff2010 Patches: dw-M17078.patch uploaded
+ by dhubbard (license 733) Tested by: dhubbard, geoff2010 Review:
+ https://reviewboard.asterisk.org/r/611/
+
+2010-04-15 21:23 +0000 [r257467-257544] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/app.h, main/app.c: Allow application options
+ with arguments to contain parentheses, through a variety of
+ escaping techniques. Fixes SWP-1194 (ABE-2143). Review:
+ https://reviewboard.asterisk.org/r/604/
+
+ * channels/chan_sip.c: Don't recreate peer, when responding to a
+ repeated deregistration attempt. When a reply to a deregistration
+ is lost in transmit, the client retries the deregistration.
+ Previously, this would cause a realtime/autocreate peer to be
+ loaded back into memory, after it had already been correctly
+ purged. Instead, we just want to resend the reply without loading
+ the peer. (closes issue #16908) Reported by: kkm Patches:
+ 20100412__issue16908.diff.txt uploaded by tilghman (license 14)
+ Tested by: kkm
+
+2010-04-15 19:40 +0000 [r257342-257426] Leif Madsen <lmadsen at digium.com>
+
+ * doc/backtrace.txt: Update backtrace.txt documentation. Update the
+ backtrace.txt documentation so it conforms to the same layout as
+ other documents we've been working on recently. Additionally, add
+ a bunch of new information about gathering backtraces for crashes
+ and deadlocks, along with ways of verifying your file before
+ uploading it. Create a couple of one line commands for people to
+ generate the files we need. (closes issue #17190) Reported by:
+ lmadsen Patches: backtrace.txt.patch-2 uploaded by lmadsen
+ (license 10) Tested by: lmadsen, pabelanger
+
+ * doc/backtrace.txt: Update address of the bug tracker.
+
+2010-04-14 23:08 +0000 [r257266] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: When forwarding a message, ensure that
+ prepending works correctly. This is a regression in 1.4, only.
+ (closes issue #17103) Reported by: mglazer Patches:
+ 20100408__issue17103.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman
+
+2010-04-13 16:46 +0000 [r257070] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/manager.c, configs/manager.conf.sample: Add an option to
+ restore past broken behavor of the Events manager action Before
+ r238915, certain values for the EventMask parameter of the Events
+ action would result in no response being returned. This patch
+ adds an option to restore that broken behavior. Also while fixing
+ this bug I discovered that passing an empty EventMasks parameter
+ would also result in no response being returned, this has been
+ fixed as well while being preserved when the broken behavior is
+ requested. (closes issue #17023) Reported by: nblasgen Review:
+ https://reviewboard.asterisk.org/r/602/
+
+2010-04-12 17:29 +0000 [r256900] Leif Madsen <lmadsen at digium.com>
+
+ * doc/HOWTO_collect_debug_information.txt (added): Add How-To
+ document on collecting debugging info for issues.asterisk.org
+ Paul Belanger has been helping a lot with bug tracking recently
+ and created this document that we can now point to when
+ additional debugging information is required. This document will
+ help those filing issues to know how to get the information
+ required when filing their issues. This will make things easier
+ on the developers. Initial text and changes by pabelanger. Tweaks
+ and editing by myself. (closes issue #17159) Reported by:
+ pabelanger Patches: HOWTO_collect_debug_information.txt.patch
+ uploaded by lmadsen (license 10) Tested by: tzafrir, pabelanger,
+ lmadsen
+
+2010-04-06 00:10 +0000 [r256225] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: DAHDI/PRI call to pri_channel_bridge() not
+ protected by PRI lock. SWP-1231 ABE-2163
+
+2010-05-03 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.31 Released
+
+2010-04-29 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.31-rc2 Released
+
+2010-04-29 10:31 +0000 [r260049] David Vossel <dvossel at digium.com>
+
+ * include/asterisk/audiohook.h, main/audiohook.c: Fixes crash in
+ audiohook_write_list. (closes issue 0017052) Reported by: dvossel
+ Tested by: dvossel. (closes issue 0016196) Reported by: atis.
+ Review: https://reviewboard.asterisk.org/r/623/
+
+2010-04-28 10:31 +0000 [r259858] David Vossel <dvossel at digium.com>
+
+ * channels/chan_local.c, main/channel.c: Resolves deadlocks in
+ chan_local. (closes issue 0017185) Reported by: schmoozecom
+ Patches: issue_17185_v1.diff uploaded by dvossel (license 671)
+ issue_17185_v2.diff uploaded by dvossel (license 671) Tested
+ by: schmoozecom, GameGamer43
+ Review: https://reviewboard.asterisk.org/r/631/
+
+2010-04-05 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.31-rc1 Released
+
+2010-04-02 23:45 +0000 [r256009-256014] Russell Bryant <russell at digium.com>
+
+ * channels/chan_local.c: Resolve a deadlock that occurs due to a
+ pointless call to ast_bridged_channel() (closes issue #16840)
+ Reported by: bzing2 Patches: patch.txt uploaded by bzing2
+ (license 902) issue_16840.rev1.diff uploaded by russell (license
+ 2) Tested by: bzing2, russell
+
+ * main/channel.c: Remove extremely verbose debug message.
+
+2010-03-31 19:09 +0000 [r255591] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: Ensure line terminators in email are
+ consistent. Fixes an issue with certain Mail Transport Agents,
+ where attachments are not interpreted correctly. (closes issue
+ #16557) Reported by: jcovert Patches:
+ 20100308__issue16557__1.4.diff.txt uploaded by tilghman (license
+ 14) 20100308__issue16557__1.6.0.diff.txt uploaded by tilghman
+ (license 14) 20100308__issue16557__trunk.diff.txt uploaded by
+ tilghman (license 14) Tested by: ebroad, zktech Reviewboard:
+ https://reviewboard.asterisk.org/r/544/
+
+2010-03-31 17:42 +0000 [r255503] Leif Madsen <lmadsen at digium.com>
+
+ * apps/app_dial.c, configs/sip.conf.sample: Add documentation
+ clarifying when 't' and 'T' can be used. (closes issue #17021)
+ Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad
+
+2010-03-30 20:56 +0000 [r255322-255409] Russell Bryant <russell at digium.com>
+
+ * channels/chan_h323.c: Don't kill Asterisk if the H323 listener
+ does not start.
+
+ * pbx/pbx_dundi.c: Don't make Asterisk not start if pbx_dundi fails
+ to initialize.
+
+2010-03-25 20:41 +0000 [r254714-254800] Jason Parker <jparker at digium.com>
+
+ * utils/Makefile: Don't remove local copies of utils in uninstall.
+
+ * main/astobj2.c, include/asterisk/astobj2.h: Fix DEBUG_THREADS
+ issue with out-of-tree modules. Take 2, without ABI breakage this
+ time. Review: https://reviewboard.asterisk.org/r/588/
+
+2010-03-25 18:51 +0000 [r254639] Russell Bryant <russell at digium.com>
+
+ * Makefile, /: Update Asterisk 1.4 to use menuselect trunk. Review:
+ https://reviewboard.asterisk.org/r/590/
+
+2010-03-25 17:33 +0000 [r254452-254552] Mark Michelson <mmichelson at digium.com>
+
+ * include/asterisk/acl.h: Add doxygen for acl.h Review:
+ https://reviewboard.asterisk.org/r/528
+
+ * main/rtp.c: Several fixes regarding RFC2833 DTMF detection. Here
+ is a copy and paste of the details from my request on reviewboard
+ that dealt with these changes: Fix 1. The first change in place
+ is to fix Mantis issue 15811, which deals with a situation where
+ Asterisk will incorrectly interpret out of order RFC2833 frames
+ as duplicate DTMF digits. For instance, we would receive a
+ sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1
+ seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1 seqno 7:
+ DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch when we
+ received the frame with seqno 5, we would interpret this as a new
+ DTMF 1. With this patch, we will check the seqno of the incoming
+ digit and not process the frame if the seqno is lower than the
+ last recorded seqno. Note that we do not record the seqno of the
+ dropped DTMF frame for future processing. While the above
+ situation is what was designed to be fixed, the patch is written
+ in such a way that the following would also be fixed too: seqno
+ 9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end) seqno 13:
+ DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno 15: DTMF 2
+ (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In this
+ second situation, the beginning of the DTMF 2 arrives before the
+ final end frame of the DTMF 1. With the patch, seqno 12 is no
+ processed and thus we properly interpret the DTMF. Fix 2. The
+ second change in place is to fix an issue like the following:
+ seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) *packet
+ lost* seqno 4: DTMF 1 (end) *packet lost* seqno 5: DTMF 1 (end)
+ *packet lost* seqno 6: DTMF 2 When we receive seqno 6, we had
[... 28094 lines stripped ...]
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