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authorlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2010-06-29 18:07:29 +0000
committerlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2010-06-29 18:07:29 +0000
commit13abb7ec3423789d0f1d7b2fa302c545cfb58e67 (patch)
treee9f12ae640607f4d853e7a64e495f506331d59c3
parent771592b6a9554635354841d4916f14d421c6b5f5 (diff)
Importing files for 1.4.34-rc1 release.
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.34-rc1@272976 f38db490-d61c-443f-a65b-d21fe96a405b
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+1.4.34-rc1
diff --git a/ChangeLog b/ChangeLog
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+2010-06-29 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.4.34-rc1 Released.
+
+2010-06-28 21:50 +0000 [r272921-272925] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c: Don't change ownership/group/permissions on run
+ directory, if it already exists. (closes issue #17076) Reported
+ by: stuarth Patches: 20100324__issue17076.diff.txt uploaded by
+ tilghman (license 14) Tested by: stuarth
+
+ * main/config.c: Also trim trailing blanks on #includes
+
+ * main/config.c: Change the way that we read include files, to
+ accommodate for changes in GCC 4.4. (closes issue #17472)
+ Reported by: seandarcy Patches: config2.patch uploaded by nivan
+ (license 1066) Tested by: nivan
+
+2010-06-28 18:47 +0000 [r272878-272881] Russell Bryant <russell@digium.com>
+
+ * tests/test_astobj2.c (added): Backport applicable parts of
+ test_astobj2.
+
+ * main/asterisk.c, Makefile, include/asterisk/test.h (added),
+ build_tools/cflags-devmode.xml, include/asterisk.h,
+ tests/Makefile, tests/test_skel.c, /, main/Makefile, tests
+ (added), include/asterisk/linkedlists.h, main/test.c (added):
+ Backport unit test API to 1.4. Review:
+ https://reviewboard.asterisk.org/r/750/
+
+2010-06-28 17:31 +0000 [r272804] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Decode URI in contact header of 302
+ response. ABE-2352
+
+2010-06-28 17:11 +0000 [r272688-272763] Russell Bryant <russell@digium.com>
+
+ * Makefile: Force SILENTMAKE where it is needed.
+
+ * Makefile: Backport method of setting SUBMAKE from trunk. By
+ setting the PRINT_DIR variable, SUBMAKE will print out the
+ directories it descends into, which is important for editors
+ (like vim) that watch the build output so that they can take you
+ to the file where an error occurred.
+
+2010-06-25 20:17 +0000 [r272562] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/voicemail_odbc_postgresql.txt: Make the structure of the
+ table specified before match the queries and results. (closes
+ issue #17557) Reported by: cmaj
+
+2010-06-24 21:58 +0000 [r272446] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: ss_thread calls pri_grab without lock
+ during overlap dial Recent changes to chan_dahdi with relation to
+ overlap dialing call pri_grab without first obtaining a lock.
+ (closes issue #17414) Reported by: pdf Patches: bug17414.patch
+ uploaded by jpeeler (license 325)
+
+2010-06-23 22:33 +0000 [r272367] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_queue.c: Send AgentComplete manager events in the event
+ of blind and attended transfers. (closes issue #16819) Reported
+ by: elbriga Patches: app_queue.diff uploaded by elbriga (license
+ 482)
+
+2010-06-23 20:57 +0000 [r272255] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * apps/app_meetme.c: First caller into a dynamic conference now
+ enter pin once. If MeetMe is configured to use dynamic conference
+ numbers, then the first caller (which creates the conference) had
+ to enter the PIN number twice. (closes issue #15878) Reported by:
+ shawkris Patches: issue15878.patch uploaded by pabelanger
+ (license 224) Tested by: pabelanger
+
+2010-06-23 18:40 +0000 [r272147] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Backport part of revision 136715 to fix
+ callerid in voicemail text files (IMAP only). (closes issue
+ #16945) Reported by: mneuhauser
+
+2010-06-22 17:31 +0000 [r271689-271902] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Decrease the module ref count in sip_hangup
+ when SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep
+ the ref count correct. (closes issue #16815) Reported by: rain
+ Patches: chan_sip-unref-fix.diff uploaded by rain (license 327)
+ (modified) Tested by: rain
+
+ * pbx/pbx_dundi.c: Allow users to specify a port for dundi peers.
+ (closes issue #17056) Reported by: klaus3000 Patches:
+ dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65)
+ Tested by: klaus3000
+
+ * configs/sip_notify.conf.sample, channels/chan_sip.c: Modify
+ chan_sip's packet generation api to automatically calculate the
+ Content-Length. This is done by storing packet content in a
+ buffer until it is actually time to send the packet, at which
+ time the size of the packet is calculated. This change was made
+ to ensure that the Content-Length is always correct. (closes
+ issue #17326) Reported by: kenner Tested by: mnicholson, kenner
+ Review: https://reviewboard.asterisk.org/r/693/
+
+2010-06-21 20:37 +0000 [r271552] Jeff Peeler <jpeeler@digium.com>
+
+ * pbx/pbx_ael.c: Do not use sizeof to calculate size of a heap
+ allocated character array. Change left out from 271399. (closes
+ issue #16053) Reported by: diLLec
+
+2010-06-18 20:52 +0000 [r271399-271444] Jeff Peeler <jpeeler@digium.com>
+
+ * pbx/pbx_ael.c: Check for newly added memory allocation failures
+ gracefully during AEL2 parsing.
+
+ * pbx/pbx_ael.c: Fix crash when parsing some heavily nested
+ statements in AEL on reload. Due to the recursion used when
+ compiling AEL in gen_prios, all the stack space was being
+ consumed when parsing some AEL that contained nesting 13 levels
+ deep. Changing a few large buffers to be heap allocated fixed the
+ crash, although I did not test how many more levels can now be
+ safely used. (closes issue #16053) Reported by: diLLec Tested by:
+ jpeeler
+
+2010-06-18 18:54 +0000 [r271339-271340] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/lock.h: Remove an unnecessary assignment that
+ causes a DEBUG_THREADS build failure on mac os x.
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/lock.h: Fix a build problem on Mac OS X with
+ DEBUG_THREADS enabled. This set of changes was already in trunk.
+
+2010-06-18 18:33 +0000 [r271335] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Eliminate deadlock potential in
+ dahdi_fixup(). (This is a backport of 269307, committed to trunk
+ by rmudgett.) Calling dahdi_indicate() when the channel private
+ lock is already held can cause a deadlock if the PRI lock is
+ needed because dahdi_indicate() will also get the channel private
+ lock. The pri_grab() function assumes that the channel private
+ lock is held once to avoid deadlock. (closes issue #17261)
+ Reported by: aragon
+
+2010-06-22 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.33.1 Released.
+
+ * channels/chan_dahdi.c: Merge revision 270404 from the 1.4 branch.
+
+ fixes FXS port still ringing when answered, as reported by Tzafrir
+ on dev-list.
+
+ (issue #17067)
+ Reported by: tzafrir
+ Tested by: alecdavis
+
+2010-06-17 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.4.33 Released.
+
+2010-06-10 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.4.33-rc2 Released.
+
+2010-06-10 Tilghman Lesher <tlesher@digium.com>
+
+ * Ensure signals are not blocked inside other signal handlers.
+
+ This eliminates the annoying <beep> on the console.
+
+ (closes issue 0017477)
+ Reported by: jvandal
+ Patches:
+ 20100610__issue17477.diff.txt uploaded by tilghman (license 14)
+
+2010-06-09 Paul Belanger <paul.belanger@polybeacon.com>
+
+ * Fix Debian init script to not use -c.
+
+ When using the init script as-is currently, it could cause issues on Debian
+ such as high CPU usage. This fix has worked for several people so I'm
+ implementing the change. We now handle color displays properly.
+
+ (closes issue 0016784)
+ Reported by: pabelanger
+ Patches:
+ 20100530__issue16784__2.diff.txt uploaded by tilghman (license 14)
+ Tested by: pabelanger, tilghman
+
+2010-06-01 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.4.33-rc1 Released.
+
+2010-06-01 15:17 +0000 [r266585] Tilghman Lesher <tlesher@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@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@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@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@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@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@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@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@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@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@digium.com>
+
+ * main/channel.c: fixes segfault when using generic plc
+
+2010-05-21 20:59 +0000 [r264996-265089] Mark Michelson <mmichelson@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@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@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@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@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@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@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@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@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@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@digium.com>
+
+ * channels/chan_dahdi.c: backport of DAHDI buffer policy dial
+ string option
+
+2010-05-13 23:08 +0000 [r263112] Jeff Peeler <jpeeler@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@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@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@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@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@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@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@digium.com>
+
+ * Asterisk 1.4.32 Released
+
+2010-05-26 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.4.32-rc2 Released
+
+2010-05-26 10:56 -0500 [r265891] Matt Nicholson <mnicholson@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@digium.com>
+
+ * Asterisk 1.4.32-rc1 Released
+
+2010-05-05 16:42 +0000 [r261274] Paul Belanger <paul.belanger@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@digium.com>
+
+ * Asterisk 1.4.31 Released
+
+2010-04-29 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.4.31-rc2 Released
+
+2010-04-29 10:31 +0000 [r260049] David Vossel <dvossel@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@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@digium.com>
+
+ * Asterisk 1.4.31-rc1 Released
+
+2010-04-02 23:45 +0000 [r256009-256014] Russell Bryant <russell@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@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@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@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@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@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@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
+ code in place that was supposed to properly end the previously
+ unended DTMF 1. The problem was that the code was essentially a
+ no-op. The code would set up an end frame for the DTMF 1 but
+ would immediately overwrite the frame with the begin for DTMF 2.
+ I changed process_dtmf_rfc2833() so that instead of returning a
+ single frame, it is given as an output parameter a list of
+ frames. Each frame that needs to be returned is appended to this
+ list. Fix 3. The final change is a minor one where an
+ AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco
+ DTMF or an RFC 3389 frame and no frame was returned, then we
+ would return &ast_null_frame. The problem is that earlier in the
+ function, we may have generated an AST_CONTROL_SRCCHANGE frame
+ and put it in the list of frames we wish to return. This frame
+ would be lost in such a case. The patch fixes this problem
+ Review: https://reviewboard.asterisk.org/r/558
+
+2010-03-25 15:57 +0000 [r254451] Terry Wilson <twilson@digium.com>
+
+ * main/file.c: Handle new SRCCHANGE control message here too
+
+2010-03-24 00:37 +0000 [r254235] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_monitor.c: Ensure that monitor recordings are written to
+ the correct location (again) This is an extension to 248860. As
+ such the dialplan test has been extended: ; non absolute path,
+ not combined exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test)
+ exten => 5040, n, dial(sip/5001) ; absolute path, not combined
+ exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2) exten =>
+ 5041, n, dial(sip/5001) ; no path, not combined exten => 5042, 1,
+ monitor(wav,monitor_test3) exten => 5042, n, dial(sip/5001) ;
+ combined: changemonitor from non absolute to no path (leaves
+ tmp/jeff) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m)
+ exten => 5043, n, changemonitor(monitor_test5) exten => 5043, n,
+ dial(sip/5001) ; combined: changemonitor from no path to non
+ absolute path exten => 5044, 1, monitor(wav,monitor_test6,m)
+ exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this
+ wasn't possible before exten => 5044, n, dial(sip/5001) ; non
+ absolute path, combined exten => 5045, 1,
+ monitor(wav,tmp/jeff/monitor_test8,m) exten => 5045, n,
+ dial(sip/5001) ; absolute path, combined exten => 5046, 1,
+ monitor(wav,/tmp/jeff/monitor_test9,m) exten => 5046, n,
+ dial(sip/5001) ; no path, combined exten => 5047, 1,
+ monitor(wav,monitor_test10,m) exten => 5047, n, dial(sip/5001) ;
+ combined: changemonitor from non absolute to absolute (leaves
+ tmp/jeff) exten => 5048, 1,
+ monitor(wav,tmp/jeff/monitor_test11,m) exten => 5048, n,
+ changemonitor(/tmp/jeff/monitor_test12) exten => 5048, n,
+ dial(sip/5001) ; combined: changemonitor from absolute to non
+ absolute (leaves /tmp/jeff) exten => 5049, 1,
+ monitor(wav,/tmp/jeff/monitor_test13,m) exten => 5049, n,
+ changemonitor(tmp/jeff/monitor_test14) exten => 5049, n,
+ dial(sip/5001) ; combined: changemonitor from no path to absolute
+ exten => 5050, 1, monitor(wav,monitor_test15,m) exten => 5050, n,
+ changemonitor(/tmp/jeff/monitor_test16) exten => 5050, n,
+ dial(sip/5001) ; combined: changemonitor from absolute to no path
+ (leaves /tmp/jeff) exten => 5051, 1,
+ monitor(wav,/tmp/jeff/monitor_test17,m) exten => 5051, n,
+ changemonitor(monitor_test18) exten => 5051, n, dial(sip/5001) ;
+ not combined: changemonitor from non absolute to no path (leaves
+ tmp/jeff) exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19)
+ exten => 5052, n, changemonitor(monitor_test20) exten => 5052, n,
+ dial(sip/5001) ; not combined: changemonitor from no path to non
+ absolute exten => 5053, 1, monitor(wav,monitor_test21) exten =>
+ 5053, n, changemonitor(tmp/jeff/monitor_test22) exten => 5053, n,
+ dial(sip/5001) ; not combined: changemonitor from non absolute to
+ absolute (leaves tmp/jeff) exten => 5054, 1,
+ monitor(wav,tmp/jeff/monitor_test23) exten => 5054, n,
+ changemonitor(/tmp/jeff/monitor_test24) exten => 5054, n,
+ dial(sip/5001) ; not combined: changemonitor from absolute to non
+ absolute (leaves /tmp/jeff) exten => 5055, 1,
+ monitor(wav,/tmp/jeff/monitor_test24) exten => 5055, n,
+ changemonitor(tmp/jeff/monitor_test25) exten => 5055, n,
+ dial(sip/5001) ; not combined: changemonitor from no path to
+ absolute exten => 5056, 1, monitor(wav,monitor_test26) exten =>
+ 5056, n, changemonitor(/tmp/jeff/monitor_test27) exten => 5056,
+ n, dial(sip/5001) ; not combined: changemonitor from absolute to
+ no path (leaves /tmp/jeff) exten => 5057, 1,
+ monitor(wav,/tmp/jeff/monitor_test28) exten => 5057, n,
+ changemonitor(monitor_test29) exten => 5057, n, dial(sip/5001)
+
+2010-03-23 22:45 +0000 [r254046-254161] Jason Parker <jparker@digium.com>
+
+ * main/astobj2.c, main/lock.c (removed), main/channel.c,
+ main/Makefile, include/asterisk/astobj2.h, UPGRADE.txt,
+ include/asterisk/lock.h: Revert revisions 254046 and 254098.
+
+ * UPGRADE.txt: Add note about the out-of-tree module ABI changes.
+
+ * main/astobj2.c, main/lock.c (added), main/channel.c,
+ main/Makefile, include/asterisk/astobj2.h,
+ include/asterisk/lock.h: Allow out-of-tree modules to load,
+ regardless of DEBUG_THREADS/DEBUG_CHANNEL_LOCKS differences. This
+ can be guaranteed by forcing the ABI to no longer change when
+ these compiler flags are set. An unfortunate side-effect to this
+ is that there is an ABI change here. However, there is some
+ mitigation. Existing modules *will* fail to load since they would
+ require functions that no longer exist. Review:
+ https://reviewboard.asterisk.org/r/508/
+
+2010-03-22 19:50 +0000 [r253799] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_features.c: Unconditionally copy the caller's account
+ code to the called party. (related to issue #16331)
+
+2010-03-21 14:26 +0000 [r253631-253670] Russell Bryant <russell@digium.com>
+
+ * main/Makefile: Fix final link on FreeBSD by adding the
+ PTHREAD_CFLAGS.
+
+ * main/sched.c, Makefile, apps/app_dial.c, channels/chan_dahdi.c,
+ main/manager.c, res/res_features.c, main/http.c, main/utils.c,
+ pbx/pbx_dundi.c, apps/app_followme.c: Resolve a number of FreeBSD
+ build issues.
+
+2010-03-18 17:57 +0000 [r253252-253349] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_userevent.c: Typo found while fixing issue #16961.
+
+ * doc/localchannel.txt: Synchronize text in localchannels.txt and
+ localchannels.tex. (issue #16963)
+
+ * doc/localchannel.txt: Update new Local channel documentation. The
+ original reporter, Kobaz, of an issue with a Local channel that
+ inspired the Local channel documentation provided some tweaks to
+ the documentation after testing what I had written. Hopefully
+ anything that was vague or unclear has been cleaned up by these
+ changes. (closes issue #16963) Reported by: kobaz Patches:
+ localchannel-2.txt uploaded by kobaz (license 834) Tested by:
+ kobaz, lmadsen
+
+2010-03-17 16:25 +0000 [r253158] Terry Wilson <twilson@digium.com>
+
+ * main/rtp.c, channels/chan_skinny.c, channels/chan_h323.c,
+ channels/chan_mgcp.c, channels/chan_sip.c,
+ include/asterisk/rtp.h: Revert API change in release branches
+ This re-renames ast_rtp_update_source to ast_rtp_new_source
+
+2010-03-17 00:26 +0000 [r253018] Leif Madsen <lmadsen@digium.com>
+
+ * configs/say.conf.sample: Add french snipset to say.conf. Add the
+ french snipset to say.conf. (Closes issue #15799)
+
+2010-03-16 20:52 +0000 [r252766-252928] Russell Bryant <russell@digium.com>
+
+ * Makefile.rules: Backport chan_sip build fix for Mac OSX 10.6 from
+ trunk.
+
+ * codecs/gsm/Makefile: Use uname -s, as done in trunk.
+
+ * codecs/gsm/Makefile: Apply codec_gsm Mac OS X 10.6 build fix that
+ is in trunk and 1.6.X.
+
+ * utils/Makefile: Don't treat warnings as errors for muted. muted
+ supports OS X, but uses functions marked as deprecated in 10.6.
+ However, the functions are still supported, so just ignore the
+ warnings for now and allow the build to proceed.
+
+2010-03-16 18:46 +0000 [r252761] Leif Madsen <lmadsen@digium.com>
+
+ * configs/extensions.ael.sample: Additional extensions.ael global
+ variable fixes. Fixing up a couple more overlapping global
+ variable namespaces shared with extensions.conf.sample. Also
+ noticed a few of the lines that were commented out didn't have
+ the closing semi-colon so I added that as well. (issue #17035)
+
+2010-03-15 21:43 +0000 [r252617] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/org.asterisk.asterisk.plist: Uh, yeah. Umask. I'm
+ stupid.
+
+2010-03-15 20:48 +0000 [r252531-252533] Leif Madsen <lmadsen@digium.com>
+
+ * configs/extensions.ael.sample: Update extensions.ael file to not
+ overlap extensions.conf. Updated the extensions.ael file so the
+ global variables don't overlap those that we have in
+ extensions.conf (sample files). This way unexpected things won't
+ happed hopefully if both pbx_ael and res_config are loaded.
+ (closes issue #17035) Reported by: pprindeville
+
+ * configure, configs/extensions.ael.sample: Revert last commit that
+ had bad changed to configure.
+
+ * configure, configs/extensions.ael.sample: Update extensions.ael
+ file to not overlap extensions.conf. Updated the extensions.ael
+ file so the global variables don't overlap those that we have in
+ extensions.conf (sample files). This way unexpected things won't
+ happed hopefully if both pbx_ael and res_config are loaded.
+ (closes issue #17035) Reported by: pprindeville
+
+2010-03-15 01:39 +0000 [r252361-252366] Tilghman Lesher <tlesher@digium.com>
+
+ * Makefile: Typo
+
+ * main/asterisk.c, Makefile,
+ contrib/init.d/org.asterisk.asterisk.plist (added): Launch
+ Asterisk on Mac OS X with launchd. Reviewboard:
+ https://reviewboard.asterisk.org/r/551/
+
+2010-03-13 00:30 +0000 [r252175] Terry Wilson <twilson@digium.com>
+
+ * main/rtp.c, channels/chan_mgcp.c, main/channel.c,
+ channels/chan_sip.c, channels/chan_skinny.c,
+ include/asterisk/rtp.h, channels/chan_h323.c,
+ configs/sip.conf.sample, include/asterisk/frame.h: Merged
+ revisions 252089 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 |
+ twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines
+ Only change the RTP ssrc when we see that it has changed This
+ change basically reverts the change reviewed in
+ https://reviewboard.asterisk.org/r/374/ and instead limits the
+ updating of the RTP synchronization source to only those times
+ when we detect that the other side of the conversation has
+ changed the ssrc. The problem is that SRCUPDATE control frames
+ are sent many times where we don't want a new ssrc, including
+ whenever Asterisk has to send DTMF in a normal bridge. This is
+ also not the first time that this mistake has been made. The
+ initial implementation of the ast_rtp_new_source function also
+ changed the ssrc--and then it was removed because of this same
+ issue. Then, we put it back in again to fix a different issue.
+ This patch attempts to only change the ssrc when we see that the
+ other side of the conversation has changed the ssrc. It also
+ renames some functions to make their purpose more clear. Review:
+ https://reviewboard.asterisk.org/r/540/ ........
+
+2010-03-12 19:58 +0000 [r251986-251997] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Forward declaring dahdi_pri was already
+ done.
+
+ * channels/chan_dahdi.c: Make chan_dahdi wakeup_sub() prototype not
+ conditional.
+
+2010-03-11 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.4.30 released
+
+2010-03-04 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.4.30-rc3 released
+
+2010-03-03 21:28 +0000 [r250613] Leif Madsen <lmadsen@digium.com>
+
+ * doc/localchannel.txt: Update existing Local channel
+ documentation. A complete re-write of the Local channel
+ documentation has been performed, with the existing information
+ from localchannel.txt and localchannel.tex merged in. (issue
+ #16637) Reported by: kobaz Patches: localchannel.tex uploaded by
+ lmadsen (license 10) localchannel.txt uploaded by lmadsen
+ (license 10) Tested by: lmadsen, jsmith, mmichelson
+
+2010-03-03 19:04 +0000 [r250480] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Make sure to clear red alarm after
+ polarity reversal. From the issue: The automatic overnight line
+ tests (or manual ones) used on UK (BT) lines causes a red alarm
+ on a dahdi / TDM400P connected channel. This is because the line
+ uses voltage tests (battery loss) and polarity reversal. The
+ polarity reversal causes chan_dahdi to initiate v23 CallerID
+ processing but during this the event DAHDI_EVENT_NOALARM is
+ ignored so that the alarm is never cleared. (closes issue #14163)
+ Reported by: jedi98 Patches: chan_dahdi-1.4-inalarm.diff uploaded
+ by jedi98 (license 653) Tested by: mattbrown, Chainsaw,
+ mikeeccleston
+
+2010-03-03 18:02 +0000 [r250394] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: fixes problem with duplicate TXREQ packets
+ When Asterisk receives an IAX2 TXREQ packet, try_transfer() will
+ call store_by_transfercallno() to link the chan_iax2_pvt struct
+ into iax_transfercallno_pvts. If a duplicate TXREQ packet is
+ received for the same call, the pvt struct will be linked into
+ iax_transfercallno_pvts multiple times. This patch fixes this.
+ Thanks rain for debugging this and providing a patch! (closes
+ issue #16904) Reported by: rain Patches:
+ iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested
+ by: rain, dvossel
+
+2010-03-02 21:08 +0000 [r250041-250050] Leif Madsen <lmadsen@digium.com>
+
+ * doc/imapstorage.txt: Update IMAP documentation. Update the IMAP
+ documentation to make it clear that storing voicemails in the
+ same folder as a large number of emails could potentially cause
+ significant slow downs when writing or retrieving voicemails.
+ (closes issue #16704) Reported by: TimeHider Tested by: lmadsen,
+ TimeHider
+
+ * configs/cdr.conf.sample: Update documentation to clarify purpose
+ of unanswered option. (closes issue #16267) Reported by: elsto
+ Patches: cdr.conf.sample.patch.txt uploaded by lmadsen (license
+ 10) Tested by: davidw, elsto
+
+ * doc/configuration.txt: Update documentation to not imply we
+ support overriding options. (issue #16855) Reported by: davidw
+
+2010-03-02 19:36 +0000 [r249845-249946] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * apps/app_echo.c: revert ability to exit echo app caused a
+ regression, as only supported VOICE, not VIDEO etc. Left in small
+ formatting change. (issue #16880)
+
+ * apps/app_echo.c: fixes ability to exit echo app when called from
+ a ISDN channel, null frames prevent '#' exit. Now only echo back
+ VOICE and DTMF frames (issue #16880) Reported by: alecdavis
+ Patches: based on echo_exit_1-6-1.diff.txt uploaded by alecdavis
+ (license 585) Tested by: alecdavis
+
+2010-03-01 19:35 +0000 [r249671] Sean Bright <sean@malleable.com>
+
+ * apps/app_voicemail.c: Fix crash in app_voicemail related to
+ message counting. We were passing a 'struct inprocess **' and
+ treating it like a 'struct inprocess *' causing a segfault.
+ (closes issue #16921) Reported by: whardier Patches:
+ 20100301_issue16921.patch uploaded by seanbright (license 71)
+ Tested by: whardier
+
+2010-03-01 17:02 +0000 [r249536] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_local.c: Modify queued frames from local channels
+ to not set the other side to up In this case, attended transfers
+ were broken due to ast_feature_request_and_dial detecting the
+ channel being set to up before the answer frame could be read and
+ therefore failing to mark the channel as ready. This fix is a
+ regression fix for 244785, which should continue to work properly
+ as well. (closes issue #16816) Reported by: jamhed Tested by:
+ jamhed, corruptor
+
+2010-02-27 23:51 +0000 [r249365] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * channels/chan_dahdi.c: overlap receiving: automatically send CALL
+ PROCEEDING when dialplan starts Following Q.931 5.2.4 When the
+ user has determined that sufficient call information has been
+ received the user shall stop T302 and send CALL PROCEEDING to the
+ network. Previously timeouts were possible if the dialplan took a
+ long time to issue any response back to the network. Verified
+ that our local TELCO also does the same. (issue #16789) Reported
+ by: alecdavis Patches: based on overlap_receiving_trunk.diff.txt
+ uploaded by alecdavis (license 585) Tested by: alecdavis (closes
+ issue #16789)
+
+2010-02-27 14:07 +0000 [r249234] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c: add a reference to the now-published IAX2
+ RFC
+
+2010-02-26 17:04 +0000 [r249100] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: For T.38 reINVITEs treat a 606 the same as a
+ 488. (closes issue #16792) Reported by: vrban Patches:
+ t38_606.patch uploaded by vrban (license 756)
+
+2010-02-25 21:22 +0000 [r248860] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_monitor.c: Ensure that monitor recordings are written to
+ the correct location (again) This is an extension to 248757. As
+ such the dialplan test has been extended: exten => 5040, 1,
+ monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
+ dial(sip/5001) exten => 5041, 1,
+ monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
+ dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
+ exten => 5042, n, dial(sip/5001) exten => 5043, 1,
+ monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n,
+ changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001)
+ exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n,
+ changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by
+ design and emits a warning exten => 5044, n, dial(sip/5001)
+
+2010-02-25 21:21 +0000 [r248859] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c: Some platforms clear /var/run at boot, which
+ makes connecting a remote console... difficult. Previously, we
+ only created the default /var/run/asterisk directory at install
+ time. While we could create it in the init script, that would not
+ work for those who start asterisk manually from the command line.
+ So the safest thing to do is to create it as part of the Asterisk
+ boot process. This also changes the ownership of the directory,
+ because the pid and ctl files are created after we setuid/setgid.
+ (closes issue #16802) Reported by: Brian Patches:
+ 20100224__issue16802.diff.txt uploaded by tilghman (license 14)
+ Tested by: tzafrir
+
+2010-02-25 18:06 +0000 [r248668-248757] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_monitor.c: Ensure that monitor recordings are written to
+ the correct location. Recordings should be placed in the monitor
+ directory when a non-absolute path is used. Exact dialplan used
+ for testing: exten => 5040, 1,
+ monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
+ dial(sip/5001) exten => 5041, 1,
+ monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
+ dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
+ exten => 5042, n, dial(sip/5001) ABE-2101
+
+ * apps/app_voicemail.c: Make deletion of temporary greetings work
+ properly with IMAP_STORAGE This same patch was merged in 220833,
+ but was skipped in this branch erroneously. (closes issue #16170)
+ Reported by: francesco_r
+
+2010-02-24 21:02 +0000 [r248582] Tilghman Lesher <tlesher@digium.com>
+
+ * main/logger.c: Remove color code sequences from verbose messages
+ that go to logfiles. (closes issue #16786) Reported by: dodo
+ Patches: logger2.patch uploaded by dodo (license 989) Tested by:
+ tilghman
+
+2010-02-23 16:26 +0000 [r248396] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes invite with replaces deadlock (closes
+ issue #16862) Reported by: pwalker Patches: replaces_deadlock_1.4
+ uploaded by dvossel (license 671) Tested by: pwalker, dvossel
+
+2010-02-22 13:52 +0000 [r248268] Olle Johansson <oej@edvina.net>
+
+ * apps/app_meetme.c: Don't log to debug unless debug is turned on
+
+2010-02-20 22:25 +0000 [r248106] Olle Johansson <oej@edvina.net>
+
+ * main/rtp.c: Make sure we support RTCP compound messages with zero
+ reports
+
+2010-02-19 19:11 +0000 [r248012] Tilghman Lesher <tlesher@digium.com>
+
+ * main/loader.c, /: Backport crash fix from trunk to 1.4, whereby
+ 'core show gracefully' could crash Asterisk. (closes issue
+ #16470) Reported by: kjotte
+
+2010-02-19 17:18 +0000 [r247910] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: Merged revision 247904 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
+ .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri,
+ 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing
+ consistent with other channel technologies. The processing of
+ DTMF tones on the receiving side of an ISDN channel is
+ inconsistent with the way it is handled in other channels,
+ especially DAHDI analog. This causes DTMF tones sent from an ISDN
+ phone to be doubled at the connected party. We are using the
+ following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes
+ Option one is necessary because the asterisk DSP DTMF detection
+ is better than mISDN's internal DSP. Not as many false positives.
+ Option two is necessary to transmit DTMF tones end to end when
+ mISDN channels are connected to SIP channels with out of band
+ DTMF for example. The symptom is that DTMF tones sent by an ISDN
+ phone are doubled on the way through asterisk when two mISDN
+ channels are connected with a Local channel in between or if it
+ is bridged to an analog channel. The doubling of DTMF tones is
+ because DTMF is passed inband to asterisk by the mISDN channel
+ and passed out of band once again after the release of the DTMF
+ tone. Passing it inband is wrong. Neither an analog channel nor
+ SIP channel passes DTMF inband if configured to inband DTMF.
+ Analog and SIP channels filter out the DTMF tones because they
+ use the voice frames returned by ast_dsp_process. But chan_misdn
+ passes the unfiltered input voice frames instead. To overcome one
+ aspect of the problem, the doubling of DTMF tones when two mISDN
+ channels are directly bridged, someone made an 'optimization',
+ where in that case the DTMF tone passed out-of-band to the peer
+ channel is not translated to an inband tone at the transmit side.
+ This optimization is bad because it does not work in general. For
+ example, analog channels or mISDN channels when bridged through
+ an intermediary local channel will generate DTMF tones from
+ out-of-band information. Also, of course, it must not be done
+ when there is no inband DTMF available. This patch fixes the
+ issue. Now chan_misdn will filter the received inband DTMF signal
+ the same as other channel types. Another change included: No need
+ to build an extra translation path because ast_process_dsp does
+ it if required. Patches: misdn-dtmf.patch JIRA ABE-2080
+
+2010-02-18 19:38 +0000 [r247651] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_features.c: Copy the calling party's account code to the
+ called party if they don't already have one. (closes issue
+ #16331) Reported by: bluefox Tested by: mnicholson
+
+2010-02-18 16:53 +0000 [r247502-247508] Leif Madsen <lmadsen@digium.com>
+
+ * README-SERIOUSLY.bestpractices.txt: Add additional link to best
+ practices document per jsmith.
+
+ * README-SERIOUSLY.bestpractices.txt (added): Add best practices
+ documentation. (issue #16808) Reported by: lmadsen (issue #16810)
+ Reported by: Nick_Lewis Tested by: lmadsen Review:
+ https://reviewboard.asterisk.org/r/507/
+
+2010-02-18 04:19 +0000 [r247422] Russell Bryant <russell@digium.com>
+
+ * Makefile, sounds/Makefile: Tweak argument handling for wget in
+ the sounds Makefile. 1) Fix the check to see if we are using wget
+ to not be full of fail. The configure script populates this
+ variable with the absolute path to wget if it is found, so it
+ didn't work. 2) Allow some extra arguments to be passed in for
+ wget. This is just a simple change to allow our Bamboo build
+ script to tell wget to be quiet and not fill up our logs with
+ download status output.
+
+2010-02-17 16:24 +0000 [r247168] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Make sure that when autofill is disabled that
+ callers not in the front of the queue cannot place calls. (closes
+ issue #16834) Reported by: kebl0155 Patches:
+ app_queue_no_autofill.v1.patch uploaded by kebl0155 (license 356)
+
+2010-02-15 23:42 +0000 [r246709] Tilghman Lesher <tlesher@digium.com>
+
+ * Makefile: Make the menuselect instructions correct by allowing
+ 'make menuselect' to actually solve dependency problems.
+ (Previously, it would fail out again with the same message about
+ running 'make menuselect', which was NOT at all helpful.)
+
+2010-02-12 23:30 +0000 [r246545] David Vossel <dvossel@digium.com>
+
+ * main/channel.c: lock channel during datastore removal On channel
+ destruction the channel's datastores are removed and destroyed.
+ Since there are public API calls to find and remove datastores on
+ a channel, a lock should be held whenever datastores are removed
+ and destroyed. This resolves a crash caused by a race condition
+ in app_chanspy.c. (closes issue #16678) Reported by:
+ tim_ringenbach Patches: datastore_destroy_race.diff uploaded by
+ tim ringenbach (license 540) Tested by: dvossel
+
+2010-02-12 18:52 +0000 [r246460] Jason Parker <jparker@digium.com>
+
+ * main/channel.c: Fix some silly formatting, and remove unnecessary
+ option_debug checks
+
+2010-02-10 17:44 +0000 [r246115] David Vossel <dvossel@digium.com>
+
+ * apps/app_queue.c: fixes random deadlock in app_queue with
+ use_weight during reload (closes issue #16677) Reported by:
+ tim_ringenbach Patches: app_queue_use_weight_deadlock.diff
+ uploaded by tim ringenbach (license 540)
+
+2010-02-10 13:37 +0000 [r245944] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/extensions.conf.sample: Include examples of FILTER usage
+ in extension patterns where a "." may be a risk.
+
+2010-02-10 08:24 +0000 [r245909] Olle Johansson <oej@edvina.net>
+
+ * res/res_smdi.c: Make sure that res_smdi loads regardless of
+ configuration, since chan_dahdi depends on res_smdi
+
+2010-02-09 22:55 +0000 [r245792] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: Fixes iaxs and iaxsl size off by one issue.
+ 2^15 = 32768 which is the maximum allowed iax2 callnumber.
+ Creating the iaxs and iaxsl array of size 32768 means the maximum
+ callnumber is actually out of bounds. This causes a nasty crash.
+ (closes issue #15997) Reported by: exarv Patches: iax_fix.diff
+ uploaded by dvossel (license 671)
+
+2010-02-08 20:39 +0000 [r245496] Jason Parker <jparker@digium.com>
+
+ * main/ast_expr2.fl, main/ast_expr2f.c: Remove reference of
+ documentation in source directory. People don't always build
+ Asterisk from source (distro packages, anybody?).
+
+2010-02-08 11:57 +0000 [r245422] Olle Johansson <oej@edvina.net>
+
+ * res/res_features.c: Res_features depends on res_adsi in 1.4
+
+2010-02-05 18:32 +0000 [r245044] Kevin P. Fleming <kpfleming@digium.com>
+
+ * contrib/firmware (removed), LICENSE: Remove contrib/firmware
+ directory as it is empty Remove explicit license for IAXy
+ firmware as it is no longer included in the tree
+
+2010-02-05 17:03 +0000 [r244926] Sean Bright <sean@malleable.com>
+
+ * main/asterisk.c: Update main copyright date.
+
+2010-02-04 23:20 +0000 [r244785] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_local.c: Change channel state on local channels for
+ busy,answer,ring. Previously local channels channel state never
+ changed. This became problematic when the state of the other side
+ of the local channel was lost, for example during a masquerade.
+ Changing the state of the local channel allows for the scenario
+ to be detected when the channel state is set to ringing, but the
+ peer isn't ringing. The specific problem scenario is described in
+ 164201. Although this was noted on one of the issues, here is the
+ tested dialplan verified to work: exten =>
+ 9700,1,Dial(Local/*9700@default&Local/#9700@default) exten =>
+ *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1)
+ exten => *9700,n,wait(3) ;3 works, 1 did not exten =>
+ *9700,n,Dial(SIP/5001) exten => #9700,1,Wait(1) ;1 works, 3 did
+ not exten =>
+ #9700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes
+ issue #14992) Reported by: davidw
+
+2010-02-01 23:13 +0000 [r244070-244242] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Backup and restore original textfile, for
+ prosthesis (gerund of prepend). Also, fix menuselect such that
+ changing voicemail build options correctly causes rebuild.
+ (closes issue #16415) Reported by: tomo1657 Patches:
+ prepention.patch uploaded by tomo1657 (license 484) (with
+ modifications by me to backport to 1.4)
+
+ * res/res_features.c: When a transferer hangs up during an attended
+ transfer BEFORE the transfer is answered, don't stop playing MOH.
+ (closes issue #16513) Reported by: litnimax Patches:
+ atxfer_moh_16513.patch uploaded by gknispel proformatique
+ (license 261) Tested by: litnimax
+
+ * main/channel.c, channels/chan_local.c: Revert previous chan_local
+ fix (r236981) and fix instead by destroying expired frames in the
+ queue. (closes issue #16525) Reported by: kobaz Patches:
+ 20100126__issue16525.diff.txt uploaded by tilghman (license 14)
+ 20100129__issue16525__1.6.0.diff.txt uploaded by tilghman
+ (license 14) Tested by: kobaz, atis (closes issue #16581)
+ Reported by: ZX81 (closes issue #16681) Reported by: alexr1
+
+2010-01-28 18:48 +0000 [r243862-243863] Leif Madsen <lmadsen@digium.com>
+
+ * BUGS: Oops, correct wrong link (https vs. http) in previous
+ commit.
+
+ * BUGS: Update location of bug tracker in documentation.
+
+2010-01-28 15:03 +0000 [r243779] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Fix a bogus third argument to
+ ast_copy_string().
+
+2010-01-27 20:35 +0000 [r243570-243691] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_queue.c: Revert 243570, I should have looked at this
+ closer. Will reopen the issue, but am leaving the review closed
+ as the change was pointless. (issue #16488)
+
+ * apps/app_queue.c: Extend announcement URL used with Queue from 80
+ chars to PATH_MAX. (closes issue #16488) Reported by: syspert
+ Patches: soundfilelen.pacth-2 uploaded by syspert (license 938)
+ Review: https://reviewboard.asterisk.org/r/475/
+
+2010-01-27 18:06 +0000 [r243486] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c: Use a safe list traversal while checking for
+ duplicate vars in pbx_builtin_setvar_helper.
+
+2010-01-26 23:55 +0000 [r243390] David Vossel <dvossel@digium.com>
+
+ * res/res_features.c: fixes bug with channel receiving wrong
+ privileges after call parking (closes issue #16429) Reported by:
+ Yasuhiro Konishi Patches: features.c.diff uploaded by Yasuhiro
+ Konishi (license 947) Tested by: dvossel
+
+2010-01-26 18:19 +0000 [r243258] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c: Remove unnecessary code in ast_read as issue
+ 16058 has been fully solved now.
+
+2010-01-25 21:50 +0000 [r242852-242969] Tilghman Lesher <tlesher@digium.com>
+
+ * main/Makefile, pbx/Makefile: Err, and use the new menuselect
+ define, too.
+
+ * build_tools/cflags.xml, build_tools/menuselect-deps.in,
+ configure, configure.ac: Only rebuild parsers by an option in
+ menuselect
+
+ * configure, main/Makefile, configure.ac, pbx/Makefile: Restore
+ FreeBSD to able-to-compile-ish-mode
+
+2010-01-25 20:08 +0000 [r242850-242851] Olle Johansson <oej@edvina.net>
+
+ * main/manager.c: Remove debugging that indeed should have been
+ gone before commit. Sorry.
+
+ * main/manager.c: Report error when writing to functions returns
+ error in AMI setvar action
+
+2010-01-25 05:42 +0000 [r242520-242728] Tilghman Lesher <tlesher@digium.com>
+
+ * main/Makefile, pbx/Makefile: Buildbot pointed out an error
+ (thanks, buildbot!)
+
+ * main/Makefile, pbx/Makefile: Oops, should have used CMD_PREFIX,
+ not ECHO_PREFIX, for the commands.
+
+ * main/Makefile: Make the build of the Asterisk expression parser
+ match that of the AEL parser.
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ pbx/ael/ael_lex.c, pbx/Makefile, makeopts.in: Only rebuild bison
+ and flex source files on demand, if bison and flex are detected
+ by the configure script. Changed after discussion on the -dev
+ list about possible unnecessary build failures, due to
+ checkouts/untars causing these special source files to possibly
+ be newer than their resulting C files. This should additionally
+ ensure that nobody need learn about extra Makefile arguments to
+ ensure the proper files get rebuilt when changes are made to
+ these special source files.
+
+2010-01-22 21:44 +0000 [r242423] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/Makefile: Rebuild from flex, bison sources when necessary.
+ (issue #14629) Reported by: Marquis Patches:
+ 20100121__issue14629.diff.txt uploaded by tilghman (license 14)
+
+2010-01-22 09:19 +0000 [r242226] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Initialize notify_types to NULL
+
+2010-01-22 01:48 +0000 [r242142] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/cdr.c: Add Dialed Number Identifier (DNID) field to cdr.
+ Branch support, retains ABI, if backend CDR collector is adaptive
+ then database requires 'dnid' field to be added, otherwise no
+ functional changes. Reported by: alecdavis Tested by: alecdavis
+ Patch cdr_dnid.diff2.txt uploaded by alecdavis (license 585)
+ Review: https://reviewboard.asterisk.org/r/455/
+
+2010-01-21 15:25 +0000 [r241932] Sean Bright <sean@malleable.com>
+
+ * configure, configure.ac: Fix configure check for
+ PTHREAD_ONCE_INIT when manually adding -Wall to CFLAGS. (closes
+ issue #16666) Reported by: romain_proformatique
+
+2010-01-21 05:53 +0000 [r241765] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_math.c: Guard against division by zero.
+
+2010-01-20 20:00 +0000 [r241626] David Vossel <dvossel@digium.com>
+
+ * Makefile: fixes parsing error in Makefile. Some echo lines were
+ missing "; . Thanks to jparker for pointing out the problem.
+
+2010-01-20 14:12 +0000 [r241543-241544] Sean Bright <sean@malleable.com>
+
+ * pbx/pbx_spool.c: Modify fix for issue 16554 to be more inline
+ with what is already in trunk. I should have taken a closer look
+ at trunk/1.6.x, as this bug has already been fixed in a much more
+ simple manner, by just settings o->vars to NULL after the
+ ast_pbx_outgoing_* calls. (issue #16554) Reported by: mav3rick
+
+ * pbx/pbx_spool.c: Fix a memory leak in pbx_spool when using SetVar
+ in a call file. In pbx_spool, when we are freeing our 'outgoing'
+ struct, we weren't deallocating the ast_variable list we had
+ built from SetVars in a call file. Adding a call to
+ ast_variables_destroy in our deallocation routine works, but only
+ if the variables have not already been passed into
+ ast_pbx_outgoing_app() or _exten(), both of which take care of
+ destroying the variable list for us. (closes issue #16554)
+ Reported by: mav3rick Patches: issue16554_20100119.patch uploaded
+ by seanbright (license 71) Tested by: mav3rick
+
+2010-01-20 09:38 +0000 [r241458] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/pbx.c: Update CDR variables as pbx starts Allows CDR
+ variables added in cdr.c:set_one_cid to become visable during the
+ call, by executing ast_cdr_update() early in __ast_pbx_run. Based
+ on cdr_update.diff3.txt (issue #16638) Reported by: alecdavis
+ Patches: cdr_update.diff3.txt uploaded by alecdavis (license 585)
+ Tested by: alecdavis
+
+2010-01-19 17:41 +0000 [r241228] Jason Parker <jparker@digium.com>
+
+ * Makefile: Allow parallel make (-j) to work properly. 1.4 changes
+ are quite different from the others. (issue #16489) Reported by:
+ Chainsaw Tested by: qwell
+
+2010-01-19 17:22 +0000 [r241227] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_agent.c: Fix deadlock in agent_read by removing
+ call to agent_logoff. One must always lock the agents list lock
+ before the agent private. agent_read locks the private
+ immediately, so locking the agents list lock is not an option
+ (which is what agent_logoff requires). Because agent_read already
+ has access to the agent private all that is necessary is to do
+ the required hanging up that agent_logoff performed. (closes
+ issue #16321) Reported by: valon24 Patches: bug16321.patch
+ uploaded by jpeeler (license 325)
+
+2010-01-18 19:54 +0000 [r241015] Sean Bright <sean@malleable.com>
+
+ * main/config.c: Plug a memory leak when reading configs with their
+ comments. While reading through configuration files with the
+ intent of returning their full contents (comments specifically)
+ we allocated some memory and then forgot to free it. This doesn't
+ fix 16554 but clears up a leak I had in the lab. (issue #16554)
+ Reported by: mav3rick Patches: issue16554_20100118.patch uploaded
+ by seanbright (license 71) Tested by: seanbright
+
+2010-01-18 16:51 +0000 [r240891] David Vossel <dvossel@digium.com>
+
+ * Makefile: updated transmit_silence option documentation in
+ asterisk.conf This patch updates the transmit_silence option to
+ better document why the option exists, and what it affects.
+ Thanks to russell for providing the verbage for this update.
+
+2010-01-18 13:27 +0000 [r240768] Olle Johansson <oej@edvina.net>
+
+ * utils/Makefile: Fix muted compilation in 1.4 only
+
+2010-01-15 23:06 +0000 [r240547] Russell Bryant <russell@digium.com>
+
+ * Makefile: Fix a spelling error in the asterisk.conf sample.
+
+2010-01-15 20:52 +0000 [r240414] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Disallow leaving more than maxmsg
+ voicemails. This is a possibility because our previous method
+ assumed that no messages are left in parallel, which is not a
+ safe assumption. Due to the vmu structure duplication, it was
+ necessary to track in-process messages via a separate structure.
+ If at some point, we switch vmu to an ao2-reference-counted
+ structure, which would eliminate the prior noted duplication of
+ structures, then we could incorporate this new in-process
+ structure directly into vmu. (closes issue #16271) Reported by:
+ sohosys Patches: 20100108__issue16271.diff.txt uploaded by
+ tilghman (license 14) 20100108__issue16271__trunk.diff.txt
+ uploaded by tilghman (license 14)
+ 20100108__issue16271__1.6.0.diff.txt uploaded by tilghman
+ (license 14) Tested by: jsutton
+
+2010-01-14 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.4.29
+
+2010-01-08 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.4.29-rc1
+
+2010-01-07 20:14 +0000 [r238409-238411] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: fixes crash in "scheduled_destroy" in
+ chan_iax A signed short was used to represent a callnumber. This
+ is makes it possible to attempt to access the iaxs array with a
+ negative index. (closes issue #16565) Reported by: jensvb
+
+ * channels/chan_sip.c: Change in sip show channels display format
+ allowing more digits for CID (closes issue 0016459) Reported by:
+ Rzadzins Patches: chan_sip_longer_cid.patch uploaded by Rzadzins
+ (license 953)
+
+2010-01-06 21:41 +0000 [r238230] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_cdr.c: Revise documentation on disposition values to
+ the actual values used. (closes issue #16289) Reported by:
+ wdoekes
+
+2010-01-06 15:18 +0000 [r237697-238009] Russell Bryant <russell@digium.com>
+
+ * apps/app_mp3.c: Resolve a crash due to an ast_frame not being
+ fully initialized. (closes issue #16531) Reported by: john8675309
+ (closes SWP-615)
+
+ * main/utils.c: Change a NOTICE log message to DEBUG where it
+ belongs. (closes issue #16479) Reported by: alexrecarey (closes
+ SWP-577)
+
+2010-01-04 21:45 +0000 [r237318-237573] Tilghman Lesher <tlesher@digium.com>
+
+ * main/say.c: Bounds checking for input string (closes issue
+ #16407) Reported by: qwell Patches: 20100104__issue16407.diff.txt
+ uploaded by tilghman (license 14)
+
+ * main/pbx.c: Regression in issue #15421 - Pattern matching (closes
+ issue #16482) Reported by: wdoekes Patches:
+ astsvn-16482-betterfix.diff uploaded by wdoekes (license 717)
+ 20091223__issue16482.diff.txt uploaded by tilghman (license 14)
+ Tested by: wdoekes, tilghman
+
+ * main/pbx.c, res/res_agi.c, include/asterisk/channel.h: Add a flag
+ to disable the Background behavior, for AGI users. This is in a
+ section of code that relates to two other issues, namely issue
+ #14011 and issue #14940), one of which was the behavior of
+ Background when called with a context argument that matched the
+ current context. This fix broke FreePBX, however, in a post-Dial
+ situation. Needless to say, this is an extremely difficult
+ collision of several different issues. While the use of an
+ exception flag is ugly, fixing all of the issues linked is rather
+ difficult (although if someone would like to propose a better
+ solution, we're happy to entertain that suggestion). (closes
+ issue #16434) Reported by: rickead2000 Patches:
+ 20091217__issue16434.diff.txt uploaded by tilghman (license 14)
+ 20091222__issue16434__1.6.1.diff.txt uploaded by tilghman
+ (license 14) Tested by: rickead2000
+
+ * channels/chan_local.c: It's also possible for the Local channel
+ to directly execute an Application. Reviewboard:
+ https://reviewboard.asterisk.org/r/452/
+
+2010-01-02 09:52 +0000 [r237135] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Release memory of the contact acl before
+ unloading module
+
+2009-12-30 21:57 +0000 [r236981] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c: Don't queue frames to channels that have
+ no means to process them. (closes issue #15609) Reported by:
+ aragon Patches:
+ 20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by
+ tilghman (license 14) Tested by: aragon Review:
+ https://reviewboard.asterisk.org/r/452/
+
+2009-12-30 20:25 +0000 [r236890] Jeff Peeler <jpeeler@digium.com>
+
+ * utils/astman.c: Remove conflicting function definitions
+ (asterisk.h) so LOW_MEMORY compiles.
+
+2009-12-28 15:12 +0000 [r236509-236585] Sean Bright <sean@malleable.com>
+
+ * include/asterisk/threadstorage.h, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Try a test
+ compile to see if PTHREAD_ONCE_INIT requires extra braces. There
+ was conditional code (based on build platform) to optioinally
+ wrap PTHREAD_ONCE_INIT in braces that was removed since it is
+ fixed in newer versions of Solaris/OpenSolaris, but I am still
+ running into it on Solaris 10 x86 so add a configure-time check
+ for it.
+
+ * apps/app_meetme.c: Avoid a crash with large numbers of MeetMe
+ conferences. Similar to changes made to Queue(), when we have
+ large numbers of conferences in meetme.conf (1000s) and we use
+ alloca()/strdupa(), we can blow out the stack and crash, so
+ instead just use a single fixed buffer. (closes issue #16509)
+ Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded
+ by seanbright (license 71) Tested by: seanbright
+
+2009-12-27 18:19 +0000 [r236433] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/rc.debian.asterisk: Turn on colors in the daemon,
+ since there's many requests for it on Ubuntu.
+
+2009-12-26 15:26 +0000 [r236357] Kevin P. Fleming <kpfleming@digium.com>
+
+ * sounds/Makefile: update to latest releases with zero uid/gid
+
+2009-12-23 15:21 +0000 [r236261] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Properly set T.38 attributes and don't
+ return before T.38 ports are configured when T.38 is found but no
+ audio stream is found. (closes issue #16318) Reported by:
+ bird_of_Luck Patches: t38-sdp-parsing-fix3.diff uploaded by
+ mnicholson (license 96), written by vrban and mnicholson Tested
+ by: vrban, mihaill
+
+2009-12-23 02:55 +0000 [r236184] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_agi.c: If EXEC only gets a single argument, don't crash
+ when the second is used. (closes issue #16504) Reported by:
+ bklang
+
+2009-12-22 16:58 +0000 [r236062] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes issue with p->method incorrectly set
+ to ACK It is possible for a second ACK to come in for a
+ retransmitted message. If an ack does not match an unacked
+ message in our queue, restore the previous p->method as this ACK
+ is completely ignored. (closes issue #16295) Reported by:
+ omolenkamp Patches: issue16295_v2.diff uploaded by dvossel
+ (license 671)
+
+2009-12-21 19:43 +0000 [r235940] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_monitor.c: Change Monitor to not assume file to write to
+ does not contain pathing. 227944 changed the fname_base argument
+ to always append the configured monitor path. This change was
+ necessary to properly compare files for uniqueness. If a full
+ path is given though, nothing needs to be appended and that is
+ handled correctly now. (closes issue #16377) (closes issue
+ #16376) Reported by: bcnit Patches:
+ res_monitor.c-issue16376-1.patch uploaded by dant (license 670)
+
+2009-12-21 16:45 +0000 [r235821] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_features.c: Send parking lot announcement to the channel
+ which parked the call, not the park-ee. (closes issue #16234)
+ Reported by: yeshuawatso Patches: 20091210__issue16234.diff.txt
+ uploaded by tilghman (license 14)
+ 20091221__issue16234__1.4.diff.txt uploaded by tilghman (license
+ 14) Tested by: yeshuawatso
+
+2009-12-18 22:39 +0000 [r235652] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, configure.ac: Revise verbiage, per #asterisk-dev
+ discussion
+
+2009-12-18 22:29 +0000 [r235635] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, include/asterisk/cdr.h: Correct CDR dispositions
+ for BUSY/FAILED This patch is simple in that it reorders the
+ disposition defines so that the fix for issue 12946 works
+ properly (the default CDR disposition was changed to
+ AST_CDR_NOANSWER). Also, the AST_CDR_FLAG_ORIGINATED flag was set
+ in ast_call to ensure all CDR records are written. The side
+ effects of CDR changes are scary, so I'm documenting the test
+ cases performed to attempt to catch any regressions. The
+ following tests were all performed using 1.4 rev 195881 vs head
+ (235571) + patch: A calls B C calls B (busy) Hangup C Hangup A
+ (Both SIP and features) A calls B A blind transfers to C Hangup C
+ (Both SIP and features) A calls B A attended transfers to C
+ Hangup C A calls B A attended transfers to C (SIP) C blind
+ transfers to A (features) Hangup A All of the test scenario CDRs
+ matched. The following tests were performed just with the patch
+ to ensure proper operation (with unanswered=yes): exten
+ =>s,1,Answer exten =>s,n,ResetCDR(w) exten =>s,n,ResetCDR(w)
+ exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w) (closes issue
+ #16180) Reported by: aatef Patches: bug16180.patch uploaded by
+ jpeeler (license 325)
+
+2009-12-18 21:18 +0000 [r235572] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, configure.ac: Point to the typical missing package,
+ not the cryptic "termcap support".
+
+2009-12-17 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.4.28
+
+2009-12-09 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.4.28-rc1
+
+2009-12-09 19:58 +0000 [r233782-233879] Russell Bryant <russell@digium.com>
+
+ * main/loader.c: Fix breakage of the "module load <module>" CLI
+ command.
+
+ * main/loader.c, formats/format_ilbc.c, formats/format_vox.c,
+ include/asterisk/module.h, formats/format_pcm.c,
+ formats/format_h263.c, formats/format_g723.c,
+ formats/format_h264.c, formats/format_jpeg.c,
+ formats/format_g726.c, formats/format_gsm.c,
+ formats/format_g729.c, formats/format_sln.c,
+ formats/format_wav.c, formats/format_ogg_vorbis.c,
+ formats/format_wav_gsm.c: Set a module load priority for format
+ modules. A recent change to app_voicemail made it such that the
+ module now assumes that all format modules are available while
+ processing voicemail configuration. However, when autoloading
+ modules, it was possible that app_voicemail was loaded before the
+ format modules. Since format modules don't depend on anything,
+ set a module load priority on them to ensure that they get loaded
+ first when autoloading. This version of the patch is specific to
+ Asterisk 1.4 and 1.6.0. These versions did not already support
+ module load priority in the module API. This adds a trivial
+ version of this which is just a module flag to include it in a
+ pass before loading "everything". Thanks to mmichelson for the
+ review! (closes issue #16412) Reported by: jiddings Tested by:
+ russell Review: https://reviewboard.asterisk.org/r/445/
+
+2009-12-08 00:02 +0000 [r233618] Atis Lezdins <atis@iq-labs.net>
+
+ * contrib/valgrind.supp: Merged revisions 233577 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r233577 |
+ atis | 2009-12-08 01:10:13 +0200 (Tue, 08 Dec 2009) | 8 lines Fix
+ compatibility with valgrind 3.3 and older. (noticed in issue
+ #16388) Reported by: parisioa Patches: valgrind.supp uloaded by
+ atis (license 242) Tested by: atis, parisioa ........
+
+2009-12-07 23:24 +0000 [r233471-233609] David Vossel <dvossel@digium.com>
+
+ * main/utils.c: hex escape control and non 7-bit clean characters
+ in uri_encode In ast_uri_encode, non 7-bit clean characters were
+ being hex escaped correctly, but control characters were not.
+ (issue #16299)
+
+ * channels/chan_sip.c: fixes missing Contact header angle brackets
+ (closes issue #16298) Reported by: mgernoth Patches:
+ reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested
+ by: dvossel
+
+2009-12-07 16:11 +0000 [r233392] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Allow SDP packets with only video session
+ information. (closes issue #16387) Reported by: zalex1953 Tested
+ by: mnicholson, zalex1953
+
+2009-12-04 21:54 +0000 [r233116-233279] David Vossel <dvossel@digium.com>
+
+ * configs/iax.conf.sample: clarify requirecalltoken option in
+ iax.sample.conf (closes issue #16223) Reported by: bklang
+ Patches: clarify-iax-requirecalltoken.patch uploaded by bklang
+ (license 919)
+
+ * apps/app_voicemail.c: document and rename strip_control() in
+ app_voicemail (closes issue #16291) Reported by: wdoekes
+
+2009-12-04 17:12 +0000 [r233092] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Only do frame payload check for HOLD frames. This
+ code was added for helping to debug the source of invalid HOLD
+ frames. However, a side effect of this is that it will
+ incorrectly report errors for frames that have an integer
+ payload. Make the check for this block specific to the HOLD frame
+ case.
+
+2009-12-04 16:59 +0000 [r233014-233091] Matthias Nick <mnick@digium.com>
+
+ * pbx/pbx_config.c: Parse global variables or expressions in hint
+ extensions Parse global variables or expressions in hint
+ extensions. Like: exten => 400,hint,DAHDI/i2/${GLOBAL(var)}
+ (closes issue #16166) Reported by: rmudgett Tested by: mnick,
+ rmudgett
+
+ * main/dsp.c: Warning message gets displayed only once Added
+ additional field 'int display_inband_dtmf_warning', which when
+ set to '1' displays the warning ('Inband DTMF is not supported on
+ codec %s. Use RFC2833'), and when set to '0' doesn't display the
+ warning. Otherwise you would get hundreds of warnings every
+ second. (closes issue #15769) Reported by: falves11 Patches:
+ patch_15769_14.txt uploaded by mnick (license 874) Tested by:
+ mnick, falves11
+
+2009-12-03 20:10 +0000 [r232820] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Deprecate "cz" in favor of "cs". Also,
+ change the use of language codes so that language registers as a
+ prefix, rather than an exact match. (closes issue #16272)
+ Reported by: patrol-cz Patches: 20091203__issue16272.diff.txt
+ uploaded by tilghman (license 14)
+
+2009-12-02 21:57 +0000 [r232581] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c: Send ack (response/message) after receiving
+ manager action userevent (closes issue #16264) Reported by: dimas
+ Patches: event-ack.patch uploaded by dimas (license 88)
+
+2009-12-02 19:03 +0000 [r232444] David Vossel <dvossel@digium.com>
+
+ * apps/app_queue.c: fixes app_queue ao2 error (closes issue #16369)
+ Reported by: vrban Patches: queue_issue_1.4.diff uploaded by
+ dvossel (license 671) Tested by: dvossel
+
+2009-12-02 17:04 +0000 [r232355] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_amd.c: Fix a bug where if you hung up very quickly after
+ calling AMD it would overwrite the AMDSTATUS of HANGUP with
+ TOOLONG. (closes issue #16239) Reported by: CGMChris
+
+2009-12-02 16:59 +0000 [r232268-232350] David Vossel <dvossel@digium.com>
+
+ * main/acl.c: ast_outaddrfor doesn't do htons() on port, looks odd
+ in strace. (closes issue #16290) Reported by: wdoekes
+
+ * funcs/func_groupcount.c: fixes segfault in func_groupcount closes
+ issue #16337) Reported by: Parantido Patches: issue_16337.diff
+ uploaded by dvossel (license 671) Tested by: Parantido, dvossel
+
+2009-12-02 04:05 +0000 [r232165] Terry Wilson <twilson@digium.com>
+
+ * main/channel.c: Fix compiling without devmode (closes issue
+ #16367) Reported by: falves11
+
+2009-12-02 00:42 +0000 [r232090] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Do not modify the gain settings on data
+ calls. (The digital flag actually represents a data call.)
+ (closes issue #15972) Reported by: udosw Patches:
+ transcap_digital_fix.diff.txt uploaded by alecdavis (license 585)
+ Tested by: alecdavis
+
+2009-12-01 23:25 +0000 [r232007] Russell Bryant <russell@digium.com>
+
+ * main/file.c: Fix a warning pointed out by buildbot.
+
+2009-12-01 21:52 +0000 [r231911-231926] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c: log channel name in dev mode as well
+
+ * main/channel.c: Fix crash with invalid frame data The crash was
+ happening as a result of a frame containing an invalid data
+ pointer, but was set with data length of zero. The few times the
+ issue was reproduced it _seemed_ that the frame was queued
+ properly, that is the data pointer was set to NULL. I never could
+ reproduce the crash so as a last resort the crash has been fixed,
+ but a check in __ast_read has been added to give as much
+ information about the source of problematic frames in the future.
+ (closes issue #16058) Reported by: atis
+
+2009-12-01 21:14 +0000 [r231853] David Vossel <dvossel@digium.com>
+
+ * main/pbx.c: WaitExten m option with no parameters generates frame
+ with zero datalen but non-null data ptr
+
+2009-12-01 15:34 +0000 [r231614-231740] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/file.c: Ignore unknown formats in ast_format_str_reduce()
+ and return an error if no know formats are found.
+
+ * apps/app_voicemail.c, include/asterisk/file.h, main/file.c,
+ main/app.c: Remove duplicate entries from voicemail format lists.
+ This prevents app_voicemail from entering an infinite loop when
+ the same format is specified twice in the format list. (closes
+ issue #15625) Reported by: Shagg63 Tested by: mnicholson Review:
+ https://reviewboard.asterisk.org/r/429/
+
+2009-11-30 17:14 +0000 [r231437-231441] David Vossel <dvossel@digium.com>
+
+ * main/rtp.c: fixes crash caused by RTP comfort noise payload
+ greater than 24 bytes AST-2009-010 (closes issue #16242) Reported
+ by: amorsen Patches: issue16242.diff uploaded by oej (license
+ 306) Tested by: amorsen, oej, dvossel
+
+ * apps/app_queue.c: app_queue crashes randomly, often during
+ call-transfers In app_queue, it is possible for a call_queue to
+ be destroyed while another object still holds a pointer to it.
+ This patch converts call_queue objects to ao2 objects allowing
+ them to be ref counted. This makes it safe for the queue_ent
+ object in queue_exec() to reference it's parent call_queue even
+ after it has left the queue. (closes issue #15686) Reported by:
+ Hatrix Patches: v2_queue_ao2.diff uploaded by dvossel (license
+ 671) Tested by: dvossel, aragon Review:
+ https://reviewboard.asterisk.org/r/427/
+
+2009-11-25 22:31 +0000 [r231298] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c: After a frame duplication failure, unlock the
+ channel before returning.
+
+2009-11-25 21:38 +0000 [r231233-231235] David Vossel <dvossel@digium.com>
+
+ * apps/app_dial.c: fixes solaris segfault on dial with verbosity >=
+ 3 (closes issue #16193) Reported by: asgaroth Patches:
+ bug_16193_1.4.21.2_vers.diff uploaded by snuffy (license 35)
+ Tested by: asgaroth, snuffy
+
+ * channels/chan_sip.c: fixes conditional jump or move depending on
+ uninitialised STACK value (closes issue #16261) Reported by:
+ edguy3 Patches: edguy16261.patch uploaded by edguy3 (license 917)
+
+2009-11-23 15:31 +0000 [r230772-230875] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: When 'sip set debug' is enabled, and the
+ last line of an incoming SIP message is not properly newline
+ terminated, ensure that that line is included in the debug
+ output. (part of issue #16268)
+
+ * main/editline/makelist.in, channels/chan_sip.c,
+ channels/ring_tone.h, channels/busy_tone.h: Correct fix for issue
+ #16268... the reporter's original patch was very close to
+ correct.
+
+ * channels/chan_sip.c: Ensure that SDP parsing does not ignore the
+ last line of the SDP. (closes issue #16268) Reported by: sgimeno
+
+2009-11-20 20:53 +0000 [r230627] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_features.c: Copy the peer CDR's userfield to the bridge
+ CDR if it exists. This is necessary for the recordagentcalls
+ option in chan_agent to store the recorded file name in the
+ bridge CDR. (closes issue #14590) Reported by: msetim Patches:
+ queue_agent_userfield.patch uploaded by Laureano (license 265)
+ Tested by: Laureano, mnicholson
+
+2009-11-19 21:22 +0000 [r230508] David Vossel <dvossel@digium.com>
+
+ * apps/app_mixmonitor.c: fixes MixMonitor thread not exiting when
+ StopMixMonitor is used (closes issue #16152) Reported by: AlexMS
+ Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license
+ 671) Tested by: dvossel, AlexMS Review:
+ https://reviewboard.asterisk.org/r/424/
+
+2009-11-19 16:09 +0000 [r230469] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/asterisk.c: Update copyright year in visible output. (cli)
+ Spotted by Stuart Henderson
+
+2009-11-30 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.4.27.1
+
+ * AST-2009-010
+
+ * SDP parser regression fix (issue #16268)
+
+2009-11-18 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.4.27
+
+2009-11-13 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.4.27-rc5
+
+2009-11-12 16:41 +0000 [r229669] David Vossel <dvossel@digium.com>
+
+ * funcs/func_audiohookinherit.c: fixes merging error, datastore was
+ being freed in the wrong function. (closes issue #16219) Reported
+ by: aragon
+
+2009-11-11 19:46 +0000 [r229498] David Brooks <dbrooks@digium.com>
+
+ * main/pbx.c: Solaris doesn't like NULL going to ast_log Solaris
+ will crash if NULL is passed to ast_log. This simple patch simply
+ uses S_OR to get around this. (closes issue #15392) Reported by:
+ yrashk
+
+2009-11-10 22:09 +0000 [r229360] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: If two pattern classes start with the same digit and
+ have the same number of characters, they will compare equal. The
+ example given in the issue report is that of [234] and [246],
+ which have these characteristics, yet they are clearly not
+ equivalent. The code still uses these two characteristics, yet
+ when the two scores compare equal, an additional check will be
+ done to compare all characters within the class to verify
+ equality. (closes issue #15421) Reported by: jsmith Patches:
+ 20091109__issue15421__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: jsmith, thedavidfactor
+
+2009-11-10 21:45 +0000 [r229355] David Ruggles <thedavidfactor@gmail.com>
+
+ * doc/externalivr.txt: Fix ExternalIVR Documentation Remove
+ documentation for event that doesn't function (closes issue
+ #16220) Reported by: thedavidfactor Patches:
+ externalivr.txt.20091110.1622.patch uploaded by thedavidfactor
+ (license 903)
+
+2009-11-10 20:03 +0000 [r229281] Joshua Colp <jcolp@digium.com>
+
+ * codecs/codec_g726.c: Remove broken support for direct transcoding
+ between G.726 RFC3551 and G.726 AAL2. On some systems the
+ translation core would actually consider g726aal2 -> g726 ->
+ signed linear to be a quicker path then g726aal2 -> signed linear
+ which exposed this problem. (closes issue #15504) Reported by:
+ globalnetinc
+
+2009-11-10 17:23 +0000 [r229191] David Ruggles <thedavidfactor@gmail.com>
+
+ * doc/externalivr.txt: Document ExternalIVR event tag collision
+ ExternalIVR uses the D tag for two different event types. This
+ documents that behavior and how to differentiate between the two
+ cases. Also includes a minor spelling fix and clarification
+ (closes issue #16211) Reported by: thedavidfactor Patches:
+ externalivr.txt.20091109.1507.patch uploaded by thedavidfactor
+ (license 903)
+
+2009-11-10 17:15 +0000 [r229167] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: don't crash on log message in solaris
+ AST-2009-006 (closes issue #16206) Reported by: bklang Tested by:
+ bklang
+
+2009-11-10 15:22 +0000 [r229091] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Reverted revision 202022. (closes issue
+ #16175) Reported by: paul-tg
+
+2009-11-09 Leif Madsen <lmadsen@digium.com>
+
+ * Release Astersik 1.4.27-rc4
+
+2009-11-09 15:37 +0000 [r228896] Leif Madsen <lmadsen@digium.com>
+
+ * main/channel.c: Update WARNING message. Update a WARNING message
+ to give a suggested fix when encountered. (closes issue #16198)
+ Reported by: atis Tested by: atis
+
+2009-11-09 14:16 +0000 [r228827] Matthew Nicholson <mnicholson@digium.com>
+
+ * include/asterisk/lock.h: Perform limited bounds checking when
+ destroying ast_mutex_t structures to make sure we don't try to
+ use negative indices. (closes issue #15588) Reported by: zerohalo
+ Patches: 20090820__issue15588.diff.txt uploaded by tilghman
+ (license 14) Tested by: zerohalo
+
+2009-11-06 22:33 +0000 [r228692] David Vossel <dvossel@digium.com>
+
+ * main/channel.c: fixes audiohook write crash occuring in chan_spy
+ whisper mode. After writing to the audiohook list in ast_write(),
+ frames were being freed incorrectly. Under certain conditions
+ this resulted in a double free crash. (closes issue #16133)
+ Reported by: wetwired (closes issue #16045) Reported by:
+ bluecrow76 Patches: issue16045.diff uploaded by dvossel (license
+ 671) Tested by: bluecrow76, dvossel, habile
+
+2009-11-06 18:32 +0000 [r228547] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Don't overwrite caller ID name on a trunk
+ with the configured fullname when using users.conf (issue
+ ABE-1989)
+
+2009-11-06 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.4.27-rc3
+
+2009-11-06 17:07 +0000 [r228418] David Vossel <dvossel@digium.com>
+
+ * codecs/codec_ilbc.c: fixes segfault in iLBC For reasons not yet
+ known, it appears possible for an ast_frame to have a datalen
+ greater than zero while the actual data is NULL during Packet
+ Loss Concealment. Most codecs don't support PLC so this doesn't
+ affect them. This patch catches the malformed frame and prevents
+ the crash from occuring. Additional efforts to determine why it
+ is possible for a frame to look like this are still being
+ investigated. (issue #16979)
+
+2009-11-06 16:41 +0000 [r228409] Joshua Colp <jcolp@digium.com>
+
+ * main/abstract_jb.c: Fix a bug caused by a partially invalid frame
+ (from the jitterbuffer) passing through the Asterisk core.
+ (closes issue #15560) Reported by: jvandal (closes issue #15709)
+ Reported by: covici
+
+2009-11-06 16:26 +0000 [r228378] Matthew Nicholson <mnicholson@digium.com>
+
+ * funcs/func_base64.c, main/utils.c: Properly handle '=' while
+ decoding base64 messages and null terminate strings returned from
+ BASE64_DECODE. (closes issue #15271) Reported by: chappell
+ Patches: base64_fix.patch uploaded by chappell (license 8) Tested
+ by: kobaz
+
+2009-11-06 15:41 +0000 [r228272-228338] David Vossel <dvossel@digium.com>
+
+ * main/astfd.c: fixes crash in astfd.c (closes issue #15981)
+ Reported by: slavon
+
+ * funcs/func_audiohookinherit.c: fixes memory leak in
+ func_audiohookinherit.c (closes issue 0015394) Reported by:
+ boroda Patches: bug15394_memoryleak_diff2.txt uploaded by dbrooks
+ (license 790) Tested by: dbrooks, boroda
+
+2009-11-05 19:14 +0000 [r228079] Jason Parker <jparker@digium.com>
+
+ * channels/chan_vpb.cc: Fix crash on VPB exception when no hardware
+ is present. (closes issue #14970) Reported by: tzafrir Patches:
+ vpb_exception.diff uploaded by tzafrir (license 46) Tested by:
+ markwaters
+
+2009-11-05 18:59 +0000 [r228078] David Brooks <dbrooks@digium.com>
+
+ * channels/chan_misdn.c: chan_misdn Asterisk 1.4.27-rc2 crash Crash
+ related to chan_misdn connection. Patch submitted by
+ gknispel_proformatique, tested by francesco_r. "I have many crash
+ since i have upgraded to Asterisk 1.4.27-rc2. Attached a full
+ bt." This patch zeros out an ast_frame. (closes issue #16041)
+ Reported by: francesco_r
+
+2009-11-04 23:47 +0000 [r227944] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_monitor.c: Fix incorrect filename comparsion after
+ monitor file change The logic to detect if a requested file is
+ indeed a different file from the current file was incorrect. The
+ main issue being confusion of the use of filename_base which was
+ previously set without pathing information and then compared to
+ another full path. Robust file comparison logic has been added to
+ properly check if two files are the same even if symlinks are
+ used. (closes issue #15313) Reported by: caspy Patches:
+ 20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license
+ 325) but mostly tilghman's work
+
+2009-11-04 20:52 +0000 [r227758-227827] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_dial.c: This patch modifies the Dial application to
+ monitor the calling channel for hangups while playing back
+ announcements. (closes issue #16005) Reported by: falves11
+ Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson
+ (license 96) Tested by: mnicholson, falves11 Review:
+ https://reviewboard.asterisk.org/r/407/
+
+ * channels/chan_sip.c: Modify the SDP parsing code to parse session
+ and media level items separately. With the new code, media level
+ proprieties should no longer be confused with session level
+ proprieties. This change also reorganizes some of the SDP parsing
+ code which should make it easier to manage in the future. (closes
+ issue #14994) Reported by: frawd Tested by: frawd, mnicholson,
+ file Review: https://reviewboard.asterisk.org/r/385/
+
+2009-11-04 19:25 +0000 [r227700-227735] Joshua Colp <jcolp@digium.com>
+
+ * static-http/prototype.js: Fix a security issue where it may be
+ possible for someone to execute a cross-site AJAX request
+ exploit. (AST-2009-009)
+
+ * channels/chan_sip.c: Fix a security issue where sending a
+ REGISTER with a differing username in the From URI and
+ Authorization header would reveal whether it was valid or not.
+ (AST-2009-008)
+
+2009-11-03 17:55 +0000 [r227275] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Make sure the outgoing flag is cleared if
+ a new channel fails to get created for outgoing calls. This is
+ the relevant portion of asterisk/trunk -r226648
+
+2009-11-03 15:36 +0000 [r227166] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix a bug where an RPID header could be
+ generated with a blank username in the URI. (closes issue #15909)
+ Reported by: kobaz
+
+2009-11-03 10:48 +0000 [r227088-227090] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Fixing bug before someone reports it...
+
+ * channels/chan_sip.c: Adding IP address in Contact ACL log message
+ and removing redundant message (based on kpfleming's feedback)
+
+ * channels/chan_sip.c: Use proper response code when violating
+ Contact ACL's. Review: https://reviewboard.asterisk.org/r/415/
+ Thanks kpfleming for a quick review. (EDVX-003)
+
+2009-11-02 20:52 +0000 [r226972] David Brooks <dbrooks@digium.com>
+
+ * channels/chan_sip.c: SIP channel name uniqueness SIP channel
+ names were supposed to be unique by way of a name suffix derived
+ from the pointer to the channel's private data. Uniqueness was
+ preserved on 32-bit systems, but not on 64-bit systems. This
+ patch, as suggested by kpfleming, replaces this suffix with a
+ simple incremented unsigned int. (closes issue #15152) Reported
+ by: palbrecht Review: https://reviewboard.asterisk.org/r/420/
+
+2009-11-02 18:08 +0000 [r226889] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c: Fix a bug where the recorded privacy
+ introduction file would not get removed if the caller hung up
+ while the called party had not yet answered. This was fixed by
+ introducing an argument to the 'n' option which, when enabled,
+ removes the introduction file under all scenarios. This was done
+ to preserve the behavior that has existed for quite some time.
+ (closes issue #14674) Reported by: ulogic Patches: bug14674.patch
+ uploaded by jpeeler (license 325)
+
+2009-11-02 17:14 +0000 [r226811] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/rc.redhat.asterisk: Don't allow two separate
+ instances of safe_asterisk when restarting from the init script.
+ (closes issue #14562) Reported by: davidw Patches: Initially
+ 20091022__issue14562.diff.txt uploaded by tilghman (license 14)
+ Modified to 20091030__Issue14562_diff.txt uploaded by davidw
+ (license 780) Tested by: davidw
+
+2009-11-02 15:31 +0000 [r226688-226736] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: fixes crash on iterator_destroy on
+ uninitialized iterator (closes issue #16162) Reported by: krn
+
+ * channels/chan_iax2.c: changes calltoken debug messages from
+ LOG_NOTICE to LOG_DEBUG like they are supposed to be (closes
+ issue #16144) Reported by: aragon
+
+2009-10-29 18:11 +0000 [r226531] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_local.c, doc/localchannel.txt: Add an option to
+ enabling passing music on hold start and stop requests through
+ instead of acting on them in chan_local. (closes issue #14709)
+ Reported by: dimas
+
+2009-10-28 20:06 +0000 [r226377-226382] Leif Madsen <lmadsen@digium.com>
+
+ * configs/sip.conf.sample: Update documentation in sip.conf.sample.
+ Update the documentation in sip.conf.sample in order to make it
+ more clear that directmedia/canreinvite do not cause Asterisk to
+ ignore reINVITEs. It is only used to stop Asterisk from
+ generating a reINVITE, but does not stop it from accepting them
+ if necessary. (closes issue #15644) Reported by: lmadsen
+
+ * doc/channelvariables.txt: Update CALLINGSUBADDR channel variable
+ documentation. (closes issue #15734) Reported by: alecdavis
+ Patches: channelvariables.tex.diff.txt uploaded by alecdavis
+ (license 585) Tested by: alecdavis
+
+2009-10-28 18:02 +0000 [r226138-226304] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/linkedlists.h: Fix documentation (pointed out by
+ TheDavidFactor on #-dev)
+
+ * main/manager.c: Manager output is not always NULL-terminated, so
+ force a NULL at the end of the filestream. (closes issue #15495)
+ Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded
+ by tilghman (license 14) Tested by: pdf
+
+2009-10-26 22:13 +0000 [r225957] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac: detect
+ ARM Linux EABI OSARCH as linux-gnu instead of linux-gnueabi * Set
+ OSARCH to linux-gnu even if host_os is linux-gnueabi * When
+ checking if we are Linux, check OSARCH rather than host_os The
+ newer ARM ABI ("EABI") shows the OS name 'linux-gnueabi' rather
+ than 'linux-gnu' . This patch sets OSARCH to be 'linux-gnu' even
+ in such a case. OSARCH is tested for the value of 'linux-gnu' in
+ one or two places in the tree. This patch also fixes the check
+ libcap to check for $OSARCH rather than $host_os . See also:
+ http://wiki.debian.org/ArmEabiPort
+
+2009-10-23 14:00 +0000 [r225581] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile: Don't force menuselect.makeopts to be rebuilt on every
+ build. For some reason the menuselect.makeopts file was listed as
+ PHONY in the Makefile, resulting in 'make' needing to rebuild it
+ for every build. This then resulted in the embedded module rules
+ being rebuilt on every build, which can be slow and is
+ unnecessary. This patch fixes the problem by properly allowing
+ 'make' to know when the menuselect.makeopts file needs to be
+ rebuilt (defining the proper dependencies).
+
+2009-10-22 21:51 +0000 [r225484] Leif Madsen <lmadsen@digium.com>
+
+ * doc/valgrind.txt, contrib/valgrind.supp (added): Clean valgrind
+ output by suppressing false errors. Update valgrind.txt
+ documentation and add valgrind.supp file in order to allow those
+ who are creating valgrind output to have less false errors in the
+ logfile. (closes issue #16007) Reported by: atis Patches:
+ valgrind.txt.diff uploaded by atis (license 242) asterisk2.supp
+ uploaded by atis (license 242) Tested by: atis, amorsen
+
+2009-10-21 20:58 +0000 [r225243] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: IAX2: VNAK loop caused by signaling frames
+ with no destination call number It is possible for the PBX thread
+ to queue up signaling frames before a destination call number is
+ received. This can result in signaling frames being sent out with
+ no destination call number. Since recent versions of Asterisk
+ require accurate destination callnumbers for all Full Frames,
+ this can cause a VNAK loop to occur. To resolve this no signaling
+ frames are sent until a destination callnumber is received, and
+ destination call numbers are now only required for iax_pvt
+ matching when the frame is an ACK. Review:
+ https://reviewboard.asterisk.org/r/413/
+
+2009-10-21 16:44 +0000 [r225169-225171] Russell Bryant <russell@digium.com>
+
+ * main/translate.c: Revert 225169, as this doesn't account for the
+ possibility of a list of frames.
+
+ * main/translate.c: Isolate the frame returned from
+ ast_translate().
+
+2009-10-21 16:02 +0000 [r225103-225105] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, apps/app_meetme.c, include/asterisk/channel.h: Fix
+ documentation for ast_softhangup() and correct the misuse
+ thereof. (closes issue #16103) Reported by: majorbloodnok
+
+ * apps/app_voicemail.c: Suffix is not needed for a match
+
+2009-10-21 14:37 +0000 [r225032] David Vossel <dvossel@digium.com>
+
+ * configs/iax.conf.sample, channels/chan_sip.c,
+ configs/sip.conf.sample, channels/chan_iax2.c: IAX/SIP
+ shrinkcallerid option The shrinking of caller id removes '(', '
+ ', ')', non-trailing '.', and '-' from the string. This means
+ values such as 555.5555 and test-test result in 555555 and
+ testtest. There are instances, such as Skype integration, where a
+ specific value is passed via caller id that must be preserved
+ unmodified. This patch makes the shrinking of caller id optional
+ in chan_sip and chan_iax in order to support such cases. By
+ default this option is on to preserve previous expected behavior.
+ (closes issue #15940) Reported by: dimas Patches: v2-15940.patch
+ uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c
+ uploaded by dvossel (license 671) Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/408/
+
+2009-10-21 02:59 +0000 [r224931] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/translate.h, main/dsp.c, main/frame.c,
+ main/translate.c, include/asterisk/dsp.h, codecs/codec_dahdi.c,
+ include/asterisk/frame.h: Isolate frames returned from a DSP
+ instance or codec translator. The reasoning for these changes are
+ the same as what I wrote in the commit message for rev 222878.
+
+2009-10-20 22:07 +0000 [r224855] Tilghman Lesher <tlesher@digium.com>
+
+ * main/audiohook.c: Pay attention to the return value of the
+ manipulate function. While this looks like an optimization, it
+ prevents a crash from occurring when used with certain audiohook
+ callbacks (diagnosed with SVN trunk, backported to 1.4 to keep
+ the source consistent across versions).
+
+2009-10-20 17:46 +0000 [r224773] Joshua Colp <jcolp@digium.com>
+
+ * res/res_features.c: Add support for relaying early media in the
+ features attended transfer option. (closes issue #14828) Reported
+ by: licedey
+
+2009-10-19 23:44 +0000 [r224670] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/rtp.c: Correct timestamp calculations when RTP sample rates
+ over 8kHz are used. While testing some endpoints that support
+ 16kHz and 32kHz sample rates, some log messages were generated
+ due to calc_rxstamp() computing timestamps in a way that produced
+ odd results, so this patch sanitizes the result of the
+ computations.
+
+2009-10-19 19:47 +0000 [r224565] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c: Do not attempt early media bridging (ie: direct
+ RTP setup) if options are enabled that should prevent it. (closes
+ issue #14763) Reported by: cupotka
+
+2009-10-17 01:32 +0000 [r224330] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Fix stale caller id data from being
+ reported in AMI NewChannel event The problem here is that
+ chan_dahdi is designed in such a way to set certain values in the
+ dahdi_pvt only once. One of those such values is the configured
+ caller id data in chan_dahdi.conf. For PRI, the configured caller
+ id data could be overwritten during a call. Instead of saving the
+ data and restoring, it was decided that for all non-analog
+ channels it was simply best to not set the configured caller id
+ in the first place and also clear it at the end of the call.
+ (closes issue #15883) Reported by: jsmith
+
+2009-10-16 20:25 +0000 [r224260] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Never released PRI channels when using
+ Busy() or Congestion() dialplan apps. When the Busy() or
+ Congestion() application is used towards ISDN (an ISDN progress
+ is sent), the responding ISDN Disconnect or Release may contain
+ the ISDN cause user busy or one of the congestion causes. In
+ chan_dahdi.c these causes will only set the needbusy or
+ needcongestion flags and not activate the softhangup procedure.
+ Unfortunately only the latter can interrupt the endless wait loop
+ of Busy()/Congestion(). Result: PRI channels staying in state
+ busy for the rest of asterisk life or until the other end times
+ out and forces the call to clear. (in issue 0014292) Reported by:
+ tomaso Patches: disc_rel_userbusy.patch uploaded by tomaso
+ (license 564) (This patch is unrelated to the issue.)
+
+2009-10-13 20:58 +0000 [r223955] Jean Galarneau <jgalarneau@digium.com>
+
+ * channels/chan_dahdi.c: Fix PRI timer T309 operation
+
+2009-10-12 23:12 +0000 [r223804] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_dial.c: Ensure ringing continues for branched calls
+ after progress is received While waiting for an answer, don't
+ send progress for branched calls for which ringing was sent.
+ (closes issue #15028) Reported by: fnordian
+
+2009-10-12 15:30 +0000 [r223692] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: Remove automatic switching from T.38 to
+ voice mode in chan_sip. chan_sip has some code to automatically
+ switch from T.38 mode to voice mode when a voice frame is written
+ to the channel while it is in T.38 mode; this was intended to
+ handle the situation when a FAX transmission has ended and the
+ channel is not yet hung up, but is causing problems at the
+ beginning of FAX sessions as well when there are still voice
+ frames 'in flight' at the time the T.38 negotiation completes.
+ This patch removes the automatic switchover. (issue #16025)
+ Reported by: jamicque
+
+2009-10-11 18:34 +0000 [r223485-223550] Russell Bryant <russell@digium.com>
+
+ * apps/app_queue.c: Remove a duplicate ao2_iterator_destroy().
+
+ * main/autoservice.c: Remove some unnecessary code.
+
+ * main/autoservice.c: Don't use data outside of its scope. The
+ purpose of this code was to have a hangup frame put on the list
+ of deferred frames. However, the code that read the hangup frame
+ was outside of the scope of where the hangup frame was declared.
+
+2009-10-09 18:20 +0000 [r223225] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c: Signal timeouts by returning AST_CONTROL_RINGING
+ when originating calls. (closes issue #15104) Reported by:
+ nblasgen Patches: manager-timeout1.diff uploaded by mnicholson
+ (license 96) Tested by: nblasgen, mnicholson
+
+2009-10-09 18:17 +0000 [r223213] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c: Fix potential memory leak in app_dial.c
+
+2009-10-09 17:52 +0000 [r223142-223205] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes sip registration using authuser in
+ user.conf (closes issue #14954) Reported by: tornblad Tested by:
+ mmichelson, tornblad, dvossel
+
+ * channels/chan_sip.c: 'auth=' did not parse md5 secret correctly
+ (closes issue https://issues.asterisk.org/view.php?id=15949)
+ Reported by: ebroad Patches: authparsefix.patch uploaded by
+ ebroad (license 878) 15949_trunk.diff uploaded by dvossel
+ (license 671) Tested by: ebroad
+
+2009-10-08 19:45 +0000 [r222878] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/file.h, main/frame.c, main/file.c,
+ include/asterisk/frame.h: Make filestream frame handling safer by
+ isolating frames before returning them. This patch is related to
+ a number of issues on the bug tracker that show crashes related
+ to freeing frames that came from a filestream. A number of fixes
+ have been made over time while trying to figure out these
+ problems, but there re still people seeing the crash. (Note that
+ some of these bug reports include information about other
+ problems. I am specifically addressing the filestream frame crash
+ here.) I'm still not clear on what the exact problem is. However,
+ what is _very_ clear is that we have seen quite a few problems
+ over time related to unexpected behavior when we try to use
+ embedded frames as an optimization. In some cases, this
+ optimization doesn't really provide much due to improvements made
+ in other areas. In this case, the patch modifies filestream
+ handling such that the embedded frame will not be returned.
+ ast_frisolate() is used to ensure that we end up with a
+ completely mallocd frame. In reality, though, we will not
+ actually have to malloc every time. For filestreams, the frame
+ will almost always be allocated and freed in the same thread.
+ That means that the thread local frame cache will be used. So,
+ going this route doesn't hurt. With this patch in place, some
+ people have reported success in not seeing the crash anymore.
+ (SWP-150) (AST-208) (ABE-1834) (issue #15609) Reported by: aragon
+ Patches: filestream_frisolate-1.4.diff2.txt uploaded by russell
+ (license 2) Tested by: aragon, russell (closes issue #15817)
+ Reported by: zerohalo Tested by: zerohalo (closes issue #15845)
+ Reported by: marhbere Review:
+ https://reviewboard.asterisk.org/r/386/
+
+2009-10-08 19:45 +0000 [r222877] David Vossel <dvossel@digium.com>
+
+ * main/netsock.c, include/asterisk/netsock.h: fixes an
+ ast_netsock_list memory leak. ABE-1998 Review:
+ https://reviewboard.asterisk.org/r/395/
+
+2009-10-08 16:33 +0000 [r222691-222797] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn_config.c: Fix memory leak if chan_misdn config
+ parameter is repeated. Memory leak when the same config option is
+ set more than once in an misdn.conf section. Why must this be
+ considered? Templates! Defining a template with default port
+ options and later adding to or overriding some of them. Patches:
+ memleak-misdn.patch JIRA ABE-1998
+
+ * channels/chan_misdn.c: chan_misdn.c:process_ast_dsp() memory leak
+ misdn.conf: astdtmf must be set to "yes". With "no", buffer loss
+ does not occur. The translated frame "f2" when passing through
+ ast_dsp_process() is not freed whenever it is not used further in
+ process_ast_dsp(). Then in the end it is never ever freed.
+ Patches: translate.patch JIRA ABE-1993
+
+2009-10-07 17:41 +0000 [r222542] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: crash on transfer handle_invite_replaces()
+ attempts to uplock a pvt's owner channel without first verifing
+ that it exists. (issue #16027)
+
+2009-10-06 23:51 +0000 [r222393-222462] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Add missing unlock(s) in dahdi_read (two
+ cases in trunk) (closes issue #15683) Reported by: alecdavis
+
+ * channels/chan_dahdi.c: Fix potential crash when entire span
+ request is received. The variable index used in this scenario for
+ accessing the dahdi_pvts was wrong and was most likely copied
+ from the several other places it is used correctly. (closes issue
+ #15998) Reported by: tsearle Patches: dahdi_reset_crash.patch
+ uploaded by tsearle (license 373)
+
+2009-10-06 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.4.27-rc2
+
+2009-10-06 01:16 +0000 [r222152] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/astobj2.c, include/asterisk/astobj2.h,
+ res/res_musiconhold.c, apps/app_queue.c, channels/chan_iax2.c:
+ Fix ao2_iterator API to hold references to containers being
+ iterated. See Mantis issue for details of what prompted this
+ change. Additional notes: This patch changes the ao2_iterator API
+ in two ways: F_AO2I_DONTLOCK has become an enum instead of a
+ macro, with a name that fits our naming policy; also, it is now
+ necessary to call ao2_iterator_destroy() on any iterator that has
+ been created. Currently this only releases the reference to the
+ container being iterated, but in the future this could also
+ release other resources used by the iterator, if the iterator
+ implementation changes to use additional resources. (closes issue
+ #15987) Reported by: kpfleming Review:
+ https://reviewboard.asterisk.org/r/383/
+
+2009-10-02 17:32 +0000 [r222026] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: Removes unnecessary unlock, clarifies a
+ memcpy.
+
+2009-10-02 16:58 +0000 [r221776-221970] Tilghman Lesher <tlesher@digium.com>
+
+ * main/astobj2.c: Ensure the result of the hash function is
+ positive. Negative array offsets suck.
+
+ * main/asterisk.c, main/rtp.c, main/say.c: Fix a bunch of
+ off-by-one errors
+
+2009-10-01 23:18 +0000 [r221769] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h:
+ Occasionally losing use of B channels in chan_misdn. I have not
+ been able to reproduce the problem of losing channels. However, I
+ have seen in the code a reentrancy problem that might give these
+ symptoms. The reentrancy patch does several things: 1) Guards B
+ channel and B channel structure allocation. 2) Makes the B
+ channel structure find routines more precise in locating records.
+ 3) Never leave a B channel allocated if we received cause 44. The
+ last item may cause temporary outgoing call problems, but they
+ should clear when the line becomes idle. (closes issue #15490)
+ Reported by: slutec18 Patches:
+ issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett
+ (license 664) Tested by: rmudgett, slutec18 (closes issue #15458)
+ Reported by: FabienToune Patches:
+ issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett
+ (license 664) Tested by: FabienToune, rmudgett, slutec18
+
+2009-10-01 15:24 +0000 [r221360-221588] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Use unsigned ints for portinuri flags.
+
+ * channels/chan_sip.c: Make portinuri a bitfield.
+
+ * channels/chan_sip.c, configs/sip.conf.sample: Fix SRV lookup and
+ Request-URI generation in chan_sip. This patch adds a new field
+ "portinuri" to the sip dialog struct and the sip peer struct.
+ That field is used during RURI generation to determine if the
+ port should be included in the RURI. It is also used in some
+ places to determine if an SRV lookup should occur. (closes issue
+ #14418) Reported by: klaus3000 Tested by: klaus3000, mnicholson
+ Review: https://reviewboard.asterisk.org/r/369/
+
+2009-09-30 19:02 +0000 [r221303] Matthias Nick <mnick@digium.com>
+
+ * funcs/func_strings.c: changed the prototype definition of
+ csv_quote
+
+2009-09-30 16:55 +0000 [r221200] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c: Avoid a potential NULL dereference. (closes issue
+ #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt
+ uploaded by tilghman (license 14) Tested by: kobaz
+
+2009-09-30 15:41 +0000 [r221153-221157] Matthias Nick <mnick@digium.com>
+
+ * configs/cdr_custom.conf.sample, funcs/func_strings.c: added a new
+ dialplan function 'CSV_QUOTE' and changed the
+ cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr
+ Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by:
+ mnick
+
+ * funcs/func_strings.c: check bounds - prevents for buffer overflow
+
+2009-09-30 14:49 +0000 [r221086] Terry Wilson <twilson@digium.com>
+
+ * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h,
+ configs/sip.conf.sample: Change the SSRC by default when our
+ media stream changes Be default, change SSRC when doing an audio
+ stream changes Asterisk doesn't honor marker bit when reinvited
+ to already-bridged RTP streams,resulting in far-end stack
+ discarding packets with "old" timestamps that areactually part of
+ a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever
+ there is a reinvite, unless the 'constantssrc' is set to true in
+ sip.conf. The original issue reported to Digium support detailed
+ the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based
+ Application Server Call comes in fromITSP, Asterisk dials the app
+ server which sends a re-invite back toAsterisk--not to negotiate
+ to send media directly to the ITSP, but to indicatethat it's
+ changing the stream it's sending to Asterisk. The app
+ servergenerates a new SSRC, sequence numbers, timestamps, and
+ sets the marker bit on the new stream. Asterisk passes through
+ the teimstamp of the new stream, butdoes not reset the SSRC,
+ sequence numbers, or set the marker bit. When the timestamp on
+ the new stream is older than the timestamp on the originalstream,
+ the ITSP (which doesn't know there has been any change) discards
+ the newframes because it thinks they are too old. This patch
+ addresses this by changing the SSRC on a stream update unless
+ constantssrc=true is set in sip.conf. Review:
+ https://reviewboard.asterisk.org/r/374/
+
+2009-09-29 20:14 +0000 [r220907] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_chanspy.c: Avoid a deadlock in chanspy, just in case the
+ spyee is masqueraded and chanspy_ds_chan_fixup() is called with
+ the channel locked. (closes issue #15965) Reported by: atis
+ Patches: chanspy-deadlock-fix1.diff uploaded by mnicholson
+ (license 96) Tested by: atis
+
+2009-09-29 17:59 +0000 [r220873] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Reduce CPU usage related to building a peer
+ merely for devicestates. This fixes a 100% CPU problem in the SIP
+ driver, found by profiling the driver while the problem was
+ occurring. (closes issue #14309) Reported by: pkempgen Patches:
+ 20090924__issue14309.diff.txt uploaded by tilghman (license 14)
+ Tested by: pkempgen, vrban
+
+2009-09-28 19:09 +0000 [r220717] Sean Bright <sean@malleable.com>
+
+ * Makefile.rules: When selecting DONT_OPTIMIZE in menuselect,
+ explicitly pass -O0 to the compiler so we override any default
+ optimization levels for a particular install.
+
+2009-09-24 19:39 +0000 [r220288] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_playback.c, main/pbx.c, apps/app_disa.c: Implicitly
+ sending a progress signal breaks some applications. Call
+ Progress() in your dialplan if you explicitly want progress to be
+ sent. (Reverts change 216430, closes issue #15957) Reported by:
+ Pavel Troller on the Asterisk-Dev mailing list
+ http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
+
+2009-09-24 18:18 +0000 [r220099-220213] Sean Bright <sean@malleable.com>
+
+ * Makefile: Resolve parallel build warnings. Reported by Klaus
+ Darilion on the asterisk-dev mailing list.
+
+ * Makefile, build_tools/mkpkgconfig: Remove the remaining bashisms
+ in the Makefile/mkpkgconfig
+
+2009-09-24 08:33 +0000 [r220027] Michiel van Baak <michiel@vanbaak.info>
+
+ * build_tools/mkpkgconfig: mkpkgconfig does not need bash so make
+ it use /bin/sh This fixes building on all systems that don't have
+ bash at /bin/bash Reported by _ys on #asterisk-dev Tested by _ys
+ on #asterisk-dev
+
+2009-09-22 21:37 +0000 [r219816] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: When IMAP variables were changed during a
+ reload, Voicemail did not use the new values. This change
+ introduces a configuration version variable, which ensures that
+ connections with the old values are not reused but are allowed to
+ expire normally. (closes issue #15934) Reported by:
+ viniciusfontes Patches: 20090922__issue15934.diff.txt uploaded by
+ tilghman (license 14) Tested by: viniciusfontes
+
+2009-09-21 16:55 +0000 [r219720] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: Reverting merge 219520. This change was not
+ necessary.
+
+2009-09-20 17:52 +0000 [r219653] Tilghman Lesher <tlesher@digium.com>
+
+ * main/file.c: Really stop the stream, when ast_closestream() is
+ called. (closes issue #15129) Reported by: bmh Patches:
+ 20090918__issue15129.diff.txt uploaded by tilghman (license 14)
+ Review: https://reviewboard.asterisk.org/r/372/
+
+2009-09-19 02:51 +0000 [r219586] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Make sure the iax_pvt exists before
+ dereferencing it. This fixes the latest crash posted on issue
+ 15609. (issue #15609)
+
+2009-09-18 23:19 +0000 [r219450-219519] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: iax2 frame double free The iax frame's
+ retrans sched id was written over right before iax2_frame_free
+ was called. In iax2_frame_free that retrans id is used to delete
+ the sched item. By writing over the retrans field before the
+ sched item could be deleted, it was possible for a retransmit to
+ occur on a freed frame.
+
+ * channels/chan_sip.c: via-header branches not updated correctly on
+ INVITE INVITE requests must always contain a new unique branch
+ id. When a new branch id is created for an INVITE, the dialog's
+ invite_branch variable must be updated so CANCEL requests use the
+ correct branch id. (closes issue #15262) Reported by: maniax
+ Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety
+ (license 608) invite_new_branch_trunk.diff uploaded by dvossel
+ (license 671) Tested by: maniax, dvossel
+
+2009-09-17 22:20 +0000 [r219320] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Send a 100 Trying response when we detect a
+ spiral. This was problematic during spiral tests at SIPit...
+ along with some other things as well.
+
+2009-09-17 21:29 +0000 [r219303] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: INVITE w/Replaces deadlock fix This patch
+ cleans up the locking logic in chan_sip.c's
+ handle_invite_replaces() function as well as making use of
+ ast_do_masquerade() rather than forcing the masquerade on an
+ ast_read(). The code had several redundant unlocks that would
+ result in 'freed more times than we've locked!' errors. I cleaned
+ these up as well as moving all the unlock logic to the end of the
+ function. This patch should also resolve the issue people were
+ having with the replacecall channel never being unlocked with one
+ legged calls. (closes issue #15151) Reported by: irroot Patches:
+ invite_w_replaces_1.4.diff uploaded by dvossel (license 671)
+ Tested by: irroot, dvossel Review:
+ https://reviewboard.asterisk.org/r/371/
+
+2009-09-17 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.4.27-rc1
+
+2009-09-17 14:58 +0000 [r219136] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, include/asterisk/cdr.h,
+ include/asterisk/channel.h: Prevent a potential race condition
+ and crash when hanging up a channel by removing the channel from
+ the channel list before begining channel tear down. This fix may
+ potentially cause problems with CDR backends that access the
+ channel a CDR is associated with via the channel list. This fix
+ makes the channel unavabile at the time when the CDR backend is
+ invoked. This has been documented in include/asterisk/cdr.h.
+ (closes issue #15316) Reported by: vmarrone Tested by: mnicholson
+ Review: https://reviewboard.asterisk.org/r/362/
+
+2009-09-16 23:21 +0000 [r219023] Tilghman Lesher <tlesher@digium.com>
+
+ * main/config.c, configs/extensions.conf.sample: Properly deal with
+ quotes in the arguments of '#exec' includes. (closes issue
+ #15583) Reported by: pkempgen Patches:
+ 20090726__issue15583.diff.txt uploaded by tilghman (license 14)
+ 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license
+ 169) Tested by: pkempgen
+
+2009-09-16 18:00 +0000 [r218867] David Brooks <dbrooks@digium.com>
+
+ * main/pbx.c: Fixes CID pattern matching behavior to mirror that of
+ extension pattern matching. Pattern matching for extensions uses
+ a type of scoring system, giving values for specificity to each
+ character in the pattern. Unfortunately, this is done character
+ by character, in order. This does lead to some less specific
+ patterns being first in line for matching, but it will usually
+ get the job done. This patch merely brings CID matching to the
+ same level as extension matching. This patch does not attempt to
+ tackle the problem shared by extension matching. (closes issue
+ #14708) Reported by: klaus3000
+
+2009-09-16 13:33 +0000 [r218798] Russell Bryant <russell@digium.com>
+
+ * contrib/firmware/iax/iaxy.bin (removed), UPGRADE.txt: Remove the
+ IAXy firmware from Asterisk. The firmware can now be found on
+ downloads.digium.com, where the rest of our binary downloads
+ live. This was the last part of our Asterisk tarballs that was
+ considered non-free by Debian. :-) (closes issue #15838) Reported
+ by: paravoid
+
+2009-09-15 22:27 +0000 [r218730] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: If the user enters the same password as
+ before, don't signal an error when the change does nothing.
+ (closes issue #15492) Reported by: cbbs70a Patches:
+ 20090713__issue15492.diff.txt uploaded by tilghman (license 14)
+
+2009-09-15 16:29 +0000 [r218623] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Fix small memory leak in handle_init_event
+ by always destroying the pthread attr before returning.
+
+2009-09-15 16:03 +0000 [r218578] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Send request contact header field with
+ response to registrer queries instead of the address of record.
+ (closes issue #14438) Reported by: ravindrad Patches:
+ regquerypatch uploaded by ravindrad (license 684) Tested by:
+ ravindrad
+
+2009-09-15 16:01 +0000 [r218577] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_followme.c: Ensure FollowMe sets language in channels it
+ creates. Also, not in the original bug report, but related fields
+ are accountcode and musicclass, and the inheritance of
+ datastores. (closes issue #15372) Reported by: Romik Patches:
+ 20090828__issue15372.diff.txt uploaded by tilghman (license 14)
+ Tested by: cervajs
+
+2009-09-15 14:57 +0000 [r218497-218498] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: revert accidental commit
+
+ * channels/chan_sip.c, sounds/Makefile: Use proper hostname for
+ downloading sound files.
+
+2009-09-14 21:47 +0000 [r218401] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Fix handling of DAHDI_EVENT_REMOVED event
+ to prevent crash in do_monitor. After talking to rmudgett about
+ some of his recent iflist locking changes, it was determined that
+ the only place that would destroy a channel without being
+ explicitly to do so was in handle_init_event. The loop to walk
+ the interface list has been modified to wait to destroy the
+ channel until the dahdi_pvt of the channel to be destroyed is no
+ longer needed. (closes issue #15378) Reported by: samy
+
+2009-09-14 19:16 +0000 [r218331] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, sounds/Makefile: Don't say "Please try
+ again" if we don't give the user another chance to try again.
+ (issue #15055, SWP-129) Reported by: jthurman
+
+2009-09-14 14:53 +0000 [r218223] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_directed_pickup.c: Ensure we don't pickup ourselves when
+ doing pickup by exten. (closes issue #15100) Reported by:
+ lmsteffan Patches: (modified) pickup.patch uploaded by lmsteffan
+ (license 779)
+
+2009-09-10 23:52 +0000 [r217917-217989] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c: Don't ring another channel, if there's not
+ enough time for a queue member to answer. (Fixes AST-228)
+
+ * contrib/scripts/iax-friends.sql, channels/chan_sip.c,
+ channels/chan_iax2.c: Backport realtime fix to 1.4
+
+2009-09-10 21:06 +0000 [r217806] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: IAX2 encryption regression The IAX2 Call
+ Token security patch inadvertently broke the use of encryption
+ due to the reorganization of code in the socket_process()
+ function. When encryption is used, an incoming full frame must
+ first be decrypted before the information elements can be parsed.
+ The security release mistakenly moved IE parsing before
+ decryption in order to process the new Call Token IE. To resolve
+ this, decryption of full frames is once again done before looking
+ into the frame. This involves searching for an existing callno,
+ checking the pvt to see if encryption is turned on, and
+ decrypting the packet before the internal fields of the full
+ frame are accessed. associated with AST-2009-006 (closes issue
+ #15834) Reported by: karesmakro Patches:
+ iax2_encryption_fix_1.4.diff uploaded by dvossel (license 671)
+ Tested by: dvossel, karesmakro Review:
+ https://reviewboard.asterisk.org/r/355/
+
+2009-09-10 19:52 +0000 [r217668-217735] Olle Johansson <oej@edvina.net>
+
+ * utils/Makefile: Reinstate muted that was removed by mistake.
+ muted doesn't compile any more on os/x, so I have to disable it
+ in order to testcompile other code...
+
+ * utils/Makefile, channels/chan_sip.c: Remove harmful code that
+ causes endless loops. Remove code that causes loops in
+ registrations. We have agreed that the patch that this code was
+ part of was bad. I am ripping out the code that causes the issue.
+ putnopvut needs to check the rest of the patch, if it needs to be
+ changed as well. This solves the issue reported in #15540, but
+ needs more work before we close it (as described above).
+
+2009-09-08 20:01 +0000 [r217156] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c: When MOH is playing on the channel,
+ announcements sent through the conference are not heard. (closes
+ issue #14588) Reported by: voipas Patches:
+ 20090716__issue14588__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: lmadsen, twisted, tilghman
+
+2009-09-04 13:56 +0000 [r216432-216435] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/utils.c, include/asterisk/lock.h: make asterisk compile
+ under devmode with DEBUG_THREADS enabled on OpenBSD
+
+ * channels/chan_sip.c: make chan_sip compile under devmode again
+
+2009-09-04 13:45 +0000 [r216430] Olle Johansson <oej@edvina.net>
+
+ * apps/app_playback.c, main/pbx.c, channels/chan_sip.c,
+ apps/app_disa.c, configs/sip.conf.sample: Make apps send PROGRESS
+ control frame for early media and fix too early media issue in
+ SIP The issue at hand is that some legacy (dying) PBX systems
+ send empty media frames on PRI links *before* any call progress.
+ The SIP channel receives these frames and by default signals 183
+ Session progress and starts sending media. This will cause phones
+ to play silence and ignore the later 180 ringing message. A bad
+ user experience. The fix is twofold: - We discovered that
+ asterisk apps that support early media ("noanswer") did not send
+ any PROGRESS frame to indicate early media. Fixed. - We introduce
+ a setting in chan_sip so that users can disable any relay of
+ media frames before the outbound channel actually indicates any
+ sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be
+ disabled for backward compatibility. In later versions of
+ Asterisk, this will be enabled. We don't assume that it will
+ change your Asterisk phone experience - only for the better. We
+ encourage third-party application developers to make sure that if
+ they have applications that wants to send early media, add a
+ PROGRESS control frame transmission to make sure that all channel
+ drivers actually will start sending early media. This has not
+ been the default in Asterisk previous to this patch, so if you
+ got inspiration from our code, you need to update accordingly.
+ Sorry for the trouble and thanks for your support. This code has
+ been running for a few months in a large scale installation (over
+ 250 servers with PRI and/or BRI links to old PBX systems). That's
+ no proof that this is an excellent patch, but, well, it's tested
+ :-)
+
+2009-09-04 13:16 +0000 [r216369] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/astobj2.c: Make sure 'start' is always initialized. This is
+ the same as rev 216222 in trunk but 1.4 is affected as well
+
+2009-09-04 10:48 +0000 [r216008-216263] Russell Bryant <russell@digium.com>
+
+ * doc/IAX2-security.txt (added), /: Merged revisions 216262 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04 Sep 2009)
+ | 2 lines Add a plain text version of the IAX2 security document.
+ ........
+
+ * /, UPGRADE.txt: Merged revisions 216080 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03 Sep 2009)
+ | 2 lines Add a note about IAX2 to UPGRADE.txt. ........
+
+ * /, doc/IAX2-security.pdf (added): Merged revisions 216005 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03 Sep 2009)
+ | 2 lines Add IAX2 security document related to AST-2009-006.
+ ........
+
+2009-09-03 18:32 +0000 [r216000] David Vossel <dvossel@digium.com>
+
+ * channels/iax2-parser.c, main/astobj2.c, configs/iax.conf.sample,
+ include/asterisk/acl.h, channels/iax2-parser.h,
+ include/asterisk/astobj2.h, channels/iax2.h, main/acl.c,
+ channels/chan_iax2.c: Merge code associated with AST-2009-006
+ (closes issue #12912) Reported by: rathaus Tested by: tilghman,
+ russell, dvossel, dbrooks
+
+2009-09-02 21:41 +0000 [r215682] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Re-send non-100 provisional responses to
+ prevent cancellation From section 13.3.1.1 of RFC 3261: If the
+ UAS desires an extended period of time to answer the INVITE, it
+ will need to ask for an "extension" in order to prevent proxies
+ from canceling the transaction. A proxy has the option of
+ canceling a transaction when there is a gap of 3 minutes between
+ responses in a transaction. To prevent cancellation, the UAS MUST
+ send a non-100 provisional response at every minute, to handle
+ the possibility of lost provisional responses. (closes issue
+ #11157) Reported by: rjain Tested by: twilson Review:
+ https://reviewboard.asterisk.org/r/315/
+
+2009-09-01 23:04 +0000 [r215270] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * apps/app_softhangup.c: Use strrchr() so SoftHangup will correctly
+ truncate multi-hyphen channel names In general channel names are
+ in the form Foo/Bar-Z, but the channel name could have multiple
+ hyphens and look like Foo/B-a-r-Z. Use strrchr to truncate the
+ channel name at the last hyphen. (closes issue #15810) Reported
+ by: dhubbard Patches: dw-softhangup-1.4.patch uploaded by
+ dhubbard (license 733)
+
+2009-08-31 16:16 +0000 [r214940] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c: Also unlock the "other" channel, when
+ returning, due to glare. (closes issue #15787) Reported by:
+ tim_ringenbach Patches: chan_local.diff uploaded by tim
+ ringenbach (license 540) Tested by: tim_ringenbach
+
+2009-08-28 20:13 +0000 [r214357-214701] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c: Modify comment to be a bit more accurate. We have
+ kept this comment around long enough, that it's pretty clear that
+ we're keeping the code, because changing the code would require a
+ pretty fundamental architectural shift. We've also taken
+ criticism in some quarters, because it was believed that it was
+ referring to the code being nasty. No, the code isn't nasty, just
+ the operation itself is rather odd. Fixed for eternity (probably
+ not).
+
+ * autoconf/libcurl.m4 (added), configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Use autoconf to
+ detect libcurl, as this enables cross-compilation checks,
+ something we didn't allow before. (closes issue #15714) Reported
+ by: pprindeville Patches: 20090813__issue15714.diff.txt uploaded
+ by tilghman (license 14) Tested by: pprindeville
+
+ * autoconf/ast_ext_lib.m4, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: One more build
+ system change, to make the descriptions look better, if we have
+ better information.
+
+ * autoconf/ast_ext_lib.m4, configure,
+ include/asterisk/autoconfig.h.in: Make autoheader descriptions
+ render correctly in our autoconfig.h file. (Figured out while
+ working with issue #14906)
+
+2009-08-26 16:36 +0000 [r214194] David Vossel <dvossel@digium.com>
+
+ * main/channel.c: ast_write() ignores ast_audiohook_write() results
+ In ast_write(), if a channel has a list of audiohooks, those
+ lists are written to and the resulting frame is what ast_write()
+ should continue with. The problem was the returned audiohook
+ frame was not being handled at all, and the original frame passed
+ into it did not contain the mixed audio, so essentially audio was
+ being lost. One result of this was chan_spy's whisper mode no
+ longer worked. To complicate the issue, frames passed into
+ ast_write may either be a single frame, or a list of frames. So,
+ as the list of frames is processed in the audiohook_write, the
+ returned frames had to be added to a new list. (closes issue
+ #15660) Reported by: corruptor Tested by: dvossel
+
+2009-08-25 19:28 +0000 [r213899-214069] Tilghman Lesher <tlesher@digium.com>
+
+ * main/say.c: I should always compile before committing...
+
+ * main/say.c: Fix pronunciation of German dates. (closes issue
+ #15273) Reported by: Benjamin Kluck Patches: say_c.patch uploaded
+ by Benjamin Kluck (license 803)
+
+ * main/pbx.c: Improve error message by informing user exactly which
+ function is missing a parethesis. (closes issue #15242) Reported
+ by: Nick_Lewis Patches: pbx.c-funcparenthesis.patch2 uploaded by
+ dbrooks (license 790) pbx.c-funcparenthesis-1.4.diff uploaded by
+ loloski (license 68)
+
+ * Makefile: Use the default runlevels for Debian derivatives,
+ instead of making up our own. (closes issue #14730) Reported by:
+ pkempgen
+
+2009-08-21 20:23 +0000 [r213631] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: Ensure that T.38 INVITEs generated by
+ Asterisk properly result in T.38 being enabled. (closes issue
+ #15373) Reported by: dcolombo Patches: chan_sip.patch uploaded by
+ mbrancaleoni (license 342) Tested by: dcolombo, mbrancaleoni
+
+2009-08-21 16:52 +0000 [r213559] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk.h: Permit DEBUG_FD_LEAKS to be used with C++
+ source files. (closes issue #15698) Reported by: slavon Patches:
+ 20090817__issue15698.diff.txt uploaded by tilghman (license 14)
+ Tested by: slavon, tilghman
+
+2009-08-21 16:03 +0000 [r213493] Jason Parker <jparker@digium.com>
+
+ * configs/queues.conf.sample: Clarify queues.conf comments to
+ specify that variables should be set in the dialplan. (closes
+ issue #15755) Reported by: trendboy
+
+2009-08-20 20:33 +0000 [r213339] Matthew Nicholson <mnicholson@digium.com>
+
+ * res/res_features.c: Fix a crash by checking the proper pointer
+ for validity before deferencing it. (closes issue #15751)
+ Reported by: atis Patches: ast_bridge_call_peer_cdr.patch
+ uploaded by atis (license 242)
+
+2009-08-20 19:53 +0000 [r213283] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.exports (added): Make all the symbols for the
+ C-client callbacks global
+
+2009-08-19 21:18 +0000 [r213103] David Vossel <dvossel@digium.com>
+
+ * apps/app_mixmonitor.c: Fixes memory leak caused by incorrectly
+ freeing mixmonitor (closes issue #15699) Reported by: edantie
+ Patches: mixmonitor.patch uploaded by edantie (license 862)
+
+2009-08-18 20:26 +0000 [r212913] Kevin P. Fleming <kpfleming@digium.com>
+
+ * doc/musiconhold-opsound.txt (added), CREDITS, /, UPGRADE.txt,
+ sounds/sounds.xml, build_tools/prep_tarball,
+ doc/musiconhold-fpm.txt (removed), doc/00README.1st,
+ sounds/Makefile: Convert this branch to Opsound music-on-hold.
+ For more details:
+ http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/
+
+2009-08-18 16:36 +0000 [r212763] Sean Bright <sean@malleable.com>
+
+ * main/manager.c: Delay the creation of temporary files until we
+ have a valid manager command to handle. Without this patch,
+ asterisk creates a temporary file before determining if the
+ specified command is valid. If invalid, we weren't properly
+ cleaning up the file. (closes issue #15730) Reported by: zmehmood
+ Patches: M15730.diff uploaded by junky (license 177) Tested by:
+ zmehmood
+
+2009-08-18 16:00 +0000 [r212727] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c: Removed some deadwood and added some
+ doxygen comments.
+
+2009-08-17 16:34 +0000 [r212498] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/misdn_config.c: Fix segfault when reloading chan_misdn.
+ If more ports were specified than configured in misdn.conf a
+ reload would crash asterisk. The problem was the unconfigured
+ port was using data from the previously configured port. When the
+ data for an unconfigured port was freed a crash would result from
+ the double free. (closes issue #12113) Reported by: agupta
+ Patches: bug12113.patch uploaded by jpeeler (license 325)
+
+2009-08-17 15:36 +0000 [r212430] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Fix uninitialized variable.
+
+2009-08-12 23:04 +0000 [r211953] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_queue.c: This patch adds additional checking when
+ generating queue log TRANSFER events. The additional checks
+ prevent generation of false TRANSFER events in certain
+ situations. (closes issue #14536) Reported by: aragon Patches:
+ queue-log-xfer-fix1.diff uploaded by mnicholson (license 96)
+ Tested by: aragon, mnicholson
+
+2009-08-12 18:46 +0000 [r211807] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Backport fix so that outbound CANCEL
+ requests have same branch as challenged INVITEs. There already
+ was code present to be sure that a CANCEL will contain the same
+ branch-id as the INVITE it is cancelling. However, for INVITES
+ which are challenged downstream, this mechanism did not work
+ properly. Now this is taken care of. This is a backport of a fix
+ already present in all 1.6.X branches and in trunk. It also fixes
+ ABE-1907.
+
+2009-08-10 19:48 +0000 [r211528-211583] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/CODING-GUIDELINES: Conversion specifiers, not format
+ specifiers
+
+ * main/indications.c, main/cli.c, pbx/pbx_loopback.c,
+ channels/chan_dahdi.c, res/res_smdi.c, pbx/pbx_spool.c,
+ channels/chan_skinny.c, pbx/pbx_ael.c, apps/app_dial.c,
+ main/pbx.c, apps/app_privacy.c, codecs/codec_speex.c,
+ funcs/func_math.c, channels/chan_agent.c, apps/app_morsecode.c,
+ apps/app_disa.c, channels/iax2-provision.c, funcs/func_cut.c,
+ pbx/dundi-parser.c, apps/app_talkdetect.c, channels/chan_misdn.c,
+ apps/app_waitforring.c, funcs/func_channel.c, apps/app_macro.c,
+ pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_chanspy.c,
+ main/asterisk.c, res/res_odbc.c, apps/app_voicemail.c,
+ doc/CODING-GUIDELINES, utils/muted.c, apps/app_meetme.c,
+ main/utils.c, apps/app_followme.c, utils/frame.c,
+ channels/misdn_config.c, main/cdr.c, main/channel.c,
+ channels/chan_phone.c, main/manager.c, apps/app_osplookup.c,
+ apps/app_setcallerid.c, res/res_agi.c, apps/app_rpt.c,
+ channels/chan_mgcp.c, apps/app_adsiprog.c, main/dnsmgr.c,
+ channels/chan_sip.c, apps/app_waitforsilence.c, agi/eagi-test.c,
+ main/acl.c, apps/app_queue.c, channels/chan_oss.c,
+ agi/eagi-sphinx-test.c, channels/chan_h323.c, pbx/pbx_dundi.c,
+ apps/app_sms.c, apps/app_verbose.c, apps/app_dahdibarge.c,
+ funcs/func_rand.c, apps/app_readfile.c, main/frame.c, /,
+ res/res_features.c, apps/app_record.c, funcs/func_strings.c,
+ apps/app_random.c, apps/app_alarmreceiver.c,
+ channels/chan_iax2.c: AST-2009-005
+
+2009-08-09 15:41 +0000 [r211274] Tilghman Lesher <tlesher@digium.com>
+
+ * main/astfd.c: Small oops. Clear the flags which have been
+ checked.
+
+2009-08-07 20:11 +0000 [r211112] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c: Resolve a deadlock involving app_chanspy and
+ masquerades. (ABE-1936)
+
+2009-08-07 18:16 +0000 [r210913-211038] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c: QUEUE_MEMBER_LIST _really_ wants the interface
+ name, not the membername. This is a partial revert of revision
+ 82590, which was an attempted cleanup, but in reality, it broke
+ QUEUE_MEMBER_LIST, which has always been intended as a method by
+ which component interfaces could be queried from the queue.
+ Membername isn't useful here, because that field cannot be used
+ to obtain further information about the member. See the
+ documentation on QUEUE_MEMBER_LIST, RemoveQueueMember,
+ QUEUE_MEMBER_PENALTY, and the various AMI commands which take a
+ member argument for further justification. (closes issue #15664)
+ Reported by: rain Patches: app_queue-queue_member_list.diff
+ uploaded by rain (license 327)
+
+ * main/channel.c: Because channel information can be accessed
+ outside of the channel thread, we must lock the channel prior to
+ modifying it. (closes issue #15397) Reported by: caspy Patches:
+ 20090714__issue15397.diff.txt uploaded by tilghman (license 14)
+ Tested by: caspy
+
+2009-08-05 19:18 +0000 [r210575] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Dialplan starts execution before the
+ channel setup is complete. * Issue 15655: For the case where
+ dialing is complete for an incoming call, dahdi_new() was asked
+ to start the PBX and then the code set more channel variables. If
+ the dialplan hungup before these channel variables got set,
+ asterisk would likely crash. * Fixed potential for overlap
+ incoming call to erroneously set channel variables as global
+ dialplan variables if the ast_channel structure failed to get
+ allocated. * Added missing set of CALLINGSUBADDR in the dialing
+ is complete case. (closes issue #15655) Reported by: alecdavis
+
+2009-08-05 18:46 +0000 [r210563] Leif Madsen <lmadsen@digium.com>
+
+ * doc/imapstorage.txt: Update imapstorage.txt documentation.
+ Updated the imapstorage.txt documentation to reflect that issues
+ with c-client versions older than 2007 seem to cause crashing
+ issues that are not seen with more recent versions. Documentation
+ has been updated to reflect this. (closes issue #14496) Reported
+ by: vbcrlfuser Patches: __20090727-imap-documentation-patch.txt
+ uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson,
+ dbrooks
+
+2009-08-04 14:51 +0000 [r210237] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile: Eliminate spurious compiler warnings from system
+ headers on *BSD platforms. Ensure that system headers located in
+ /usr/local/include are actually treated as system headers by the
+ compiler, and not as local headers which are subject to warnings
+ from the -Wundef compiler option and others. (closes issue
+ #15606) Reported by: mvanbaak
+
+2009-08-03 16:15 +0000 [r210067] David Brooks <dbrooks@digium.com>
+
+ * channels/chan_dahdi.c: Fixes dialplan wildcard extension taking
+ precedence over call pickup code. Prior to this patch, a wildcard
+ extension in the dialplan (for example, _*.) would take
+ precedence over picking up a call in the channel's pickup group.
+ This patch simply moves the block of code handling pickup group
+ matching to above the extension matching code. (closes issue
+ #14735) Reported by: stevedavies Review:
+ https://reviewboard.asterisk.org/r/319/
+
+2009-08-03 16:11 +0000 [r210064-210066] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_oss.c, apps/app_playback.c, main/asterisk.exports,
+ configure, include/asterisk/autoconfig.h.in,
+ include/asterisk/compat.h, main/strcompat.c, configure.ac,
+ funcs/func_cut.c: Reverting index() fix, applying a different
+ methodology, based upon developer discussions. (related to issue
+ #15639)
+
+ * main/asterisk.exports, include/asterisk/compat.h: Helps if we
+ export the index() function. (Related to issue #15639)
+
+ * configure, include/asterisk/autoconfig.h.in, main/strcompat.c,
+ configure.ac: Apparently, some platforms don't have the index()
+ function. (closes issue #15639) Reported by: nmav
+
+2009-08-01 11:27 +0000 [r209838-209879] Russell Bryant <russell@digium.com>
+
+ * main/db1-ast/mpool/mpool.c: Resolve a valgrind warning about a
+ read from uninitialized memory. (issue #15396) Reported by:
+ aragon
+
+ * apps/app_milliwatt.c: Modify how Playtones() is used in
+ Milliwatt() to resolve gain issue. When Milliwatt() was changed
+ internally to use Playtones() so that the proper tone was used,
+ it introduced a drop in gain in the output signal. So, use the
+ playtones API directly and specify a volume argument such that
+ the output matches the gain of the original Milliwatt() code.
+ (closes issue #15386) Reported by: rue_mohr Patches:
+ issue_15386.rev2.diff uploaded by russell (license 2) Tested by:
+ rue_mohr
+
+2009-08-01 00:52 +0000 [r209759] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/misdn/isdn_lib.c, utils/frame.c, main/Makefile,
+ channels/misdn/ie.c: Minor changes inspired by testing with
+ latest GCC. The latest GCC (what will become 4.5.x) has a few new
+ warnings, that in these cases found some either downright buggy
+ code, or at least seriously poorly designed code that could be
+ improved.
+
+2009-07-28 00:12 +0000 [r209315] Tilghman Lesher <tlesher@digium.com>
+
+ * sounds/sounds.xml: Publish French extra sounds
+
+2009-07-27 17:44 +0000 [r209131] Mark Michelson <mmichelson@digium.com>
+
+ * main/udptl.c, configs/udptl.conf.sample: Allow for UDPTL to use
+ only even-numbered ports if desired. There are some VoIP
+ providers out there that will not accept SDP offers with odd
+ numbered UDPTL ports. While it is my personal opinion that these
+ VoIP providers are misinterpreting RFC 2327, it really is not a
+ big deal to play along with their silly little games. Of course,
+ since restricting UDPTL ports to only even numbers reduces the
+ range of available ports by half, so the option to use only even
+ port numbers is off by default. A user can enable the behavior by
+ setting use_even_ports=yes in udptl.conf. (closes issue #15182)
+ Reported by: CGMChris Patches: 15182.patch uploaded by mmichelson
+ (license 60) Tested by: CGMChris
+
+2009-07-27 09:56 +0000 [r208990] Michiel van Baak <michiel@vanbaak.info>
+
+ * res/res_crypto.c: backport rev 205532 from trunk: pthread_self
+ returns a pthread_t which is not an unsigned int on all pthread
+ implementations. Casting it to an unsigned int fixes compiler
+ warnings.
+
+2009-07-27 01:18 +0000 [r208923] Jeff Peeler <jpeeler@digium.com>
+
+ * main/translate.c, channels/chan_iax2.c: Fix logic errors from
+ 208746
+
+2009-07-25 06:19 +0000 [r208746] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_skinny.c, main/translate.c, channels/chan_iax2.c:
+ Fix compiling under dev-mode with gcc 4.4.0. Mostly trivial
+ changes, but I did not know of any other way to fix the
+ "dereferencing type-punned pointer will break strict-aliasing
+ rules" error without creating a tmp variable in chan_skinny.
+
+2009-07-24 19:24 +0000 [r208622] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Don't impose an arbitrary limit on member lines
+ in queues.conf I know what some of you are thinking: "UGH! Mark,
+ why are you using ast_strdup and ast_free for the string when you
+ can just use ast_strdupa and let the memory free itself?! Have
+ the bats been chewing on your brain again?" Based on past
+ experiences, I don't like using ast_strdupa inside a loop. It's a
+ good way to potentially exhaust stack space. Also, since this
+ only happens when reloading queues, I don't think that heap
+ allocations and frees are going to be a huge problem. (closes
+ issue #15559) Reported by: amorsen
+
+2009-07-24 18:38 +0000 [r208592] Russell Bryant <russell@digium.com>
+
+ * apps/app_dial.c: Do not log an ERROR if autoservice_stop()
+ returns -1. This does not indicate an error. A return of -1 just
+ means that the channel has been hung up. (reported in
+ #asterisk-dev)
+
+2009-07-24 18:26 +0000 [r208386-208587] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Only send a BYE when hanging up a channel
+ that is up. For cases where Asterisk sends an INVITE and receives
+ a non 2XX final response, Asterisk would follow the INVITE
+ transaction by immediately sending a BYE, which was unnecessary.
+ (closes issue #14575) Reported by: chris-mac
+
+ * channels/chan_sip.c: Fix a problem where a 491 response could be
+ sent out of dialog. This generalizes the fix for issue 13849. The
+ initial fix corrected the problem that Asterisk would reply with
+ a 491 if a reinvite were received from an endpoint and we had not
+ yet received an ACK from that endpoint for the initial INVITE it
+ had sent us. This expansion also allows Asterisk to appropriately
+ handle an INVITE with authorization credentials if Asterisk had
+ not received an ACK from the previous transaction in which
+ Asterisk had responded to an unauthorized INVITE with a 407.
+ (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch
+ uploaded by mmichelson (license 60) Tested by: klaus3000
+
+2009-07-23 19:19 +0000 [r208380] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Only set the priindication setting when
+ not performing a reload (closes issue #14696) Reported by:
+ fdecher
+
+2009-07-23 16:29 +0000 [r208262-208312] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Remove inaccurate XXX comment.
+
+ * channels/chan_sip.c: Properly handle 183 responses which do not
+ contain an SDP. (closes issue #15442) Reported by: ffloimair
+ Patches: 15442.patch uploaded by mmichelson (license 60) Tested
+ by: tkarl, ffloimair
+
+2009-07-22 20:23 +0000 [r207945-208083] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.exports, include/asterisk/compat.h: Export symbols
+ for functions included in our compatibility headers. (closes
+ issue #15556) Reported by: smw1218
+
+ * funcs/func_strings.c: Force an error if a blank is passed to
+ QUOTE (because the documentation states the argument is not
+ optional). This change makes URIENCODE and QUOTE behave
+ similarly, since the documentation states that the argument is
+ not optional, for both. (closes issue #15439) Reported by:
+ pkempgen Patches: 20090706__issue15439.diff.txt uploaded by
+ tilghman (license 14)
+
+2009-07-21 20:16 +0000 [r207786-207827] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Wait for wink before dialing when using
+ E&M wink signaling There was already code for other signaling
+ types in dahdi_handle_event to handle dialing if a dial operation
+ dial string was present. Simply add SIG_EMWINK to the list.
+ (closes issue #14434) Reported by: araasch
+
+ * channels/chan_dahdi.c: Revert r207573, this approach could
+ potentially block for an unacceptable amount of time.
+
+2009-07-21 14:26 +0000 [r207714] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c: Document default timeout for AMI originations.
+ AST-224
+
+2009-07-21 13:04 +0000 [r207647] Kevin P. Fleming <kpfleming@digium.com>
+
+ * codecs/lpc10/Makefile, main/db1-ast/Makefile, Makefile,
+ agi/Makefile, codecs/Makefile, utils/Makefile, main/Makefile,
+ codecs/gsm/Makefile, Makefile.moddir_rules, Makefile.rules,
+ pbx/Makefile, res/Makefile, channels/Makefile: Ensure that
+ user-provided CFLAGS and LDFLAGS are honored. This commit changes
+ the build system so that user-provided flags (in ASTCFLAGS and
+ ASTLDFLAGS) are supplied to the compiler/linker *after* all flags
+ provided by the build system itself, so that the user can
+ effectively override the build system's flags if desired. In
+ addition, ASTCFLAGS and ASTLDFLAGS can now be provided *either*
+ in the environment before running 'make', or as variable
+ assignments on the 'make' command line. As a result, the use of
+ COPTS and LDOPTS is no longer necessary, so they are no longer
+ documented, but are still supported so as not to break existing
+ build systems that supply them when building Asterisk.
+
+2009-07-20 23:23 +0000 [r207573] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Wait for wink before dialing when using
+ E&M wink signaling This patch adds a new dahdi_wait function to
+ specifically wait for the wink event. If the wink is not
+ eventually received the channel is hung up. (closes issue #14434)
+ Reported by: araasch Patches: emwinkmod uploaded by araasch
+ (license 693)
+
+2009-07-20 19:39 +0000 [r207423] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Answer video SDP offers properly when
+ videosupport is not enabled. Copied from Review board: In issue
+ 12434, the reporter describes a situation in which audio and
+ video is offered on the call, but because videosupport is
+ disabled in sip.conf, Asterisk gives no response at all to the
+ video offer. According to RFC 3264, all media offers should have
+ a corresponding answer. For offers we do not intend to actually
+ reply to with meaningful values, we should still reply with the
+ port for the media stream set to 0. In this patch, we take note
+ of what types of media have been offered and save the information
+ on the sip_pvt. The SDP in the response will take into account
+ whether media was offered. If we are not otherwise going to
+ answer a media offer, we will insert an appropriate m= line with
+ the port set to 0. It is important to note that this patch is
+ pretty much a bandage being applied to a broken bone. The patch
+ *only* helps for situations where video is offered but
+ videosupport is disabled and when udptl_pt is disabled but T.38
+ is offered. Asterisk is not guaranteed to respond to every media
+ offer. Notable cases are when multiple streams of the same type
+ are offered. The 2 media stream limit is still present with this
+ patch, too. In trunk and the 1.6.X branches, things will be a bit
+ different since Asterisk also supports text in SDPs as well.
+ (closes issue #12434) Reported by: mnnojd Review:
+ https://reviewboard.asterisk.org/r/311 Review:
+ https://reviewboard.asterisk.org/r/313
+
+2009-07-20 16:26 +0000 [r207360] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Only do the chan->fdno check in ast_read() in a
+ developer build. I changed this check to only happen in a
+ dev-mode build. I also added a comment explaining what is going
+ on. I also made it so that detection of this situation does not
+ affect ast_read() operation. (closes issue #14723) Reported by:
+ seadweller
+
+2009-07-20 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.4.26
+
+2009-07-13 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.4.26-rc6
+
+2009-07-13 15:12 +0000 [r206126] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c: Print CID match in "show dialplan". (closes issue
+ #14702) Reported by: klaus3000 Patches:
+ patch_asterisk_1.4.23_CID_matching.txt uploaded by klaus3000
+ (license 65)
+
+2009-07-10 17:39 +0000 [r205877] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Properly ACK 487 responses to canceled
+ INVITEs. From the review board request: The fix from review 298
+ has exposed a new bug in chan_sip. When we hang up an outgoing
+ call, we first will dump all the outstanding packets on the
+ sip_pvt using __sip_pretend_ack. Then, if we can, we send a
+ CANCEL. The problem with this is that since destroyed all the
+ outstanding packets on the dialog, we cannot match the incoming
+ 487 response to our INVITE. Because we cannot match the response,
+ we do not send an ACK. To correct this, instead of using
+ __sip_pretend_ack, I have changed the code to loop through the
+ list of packets and call __sip_semi_ack on each one instead. This
+ causes us to stop retransmitting the requests, but we still have
+ them around in case we get responses for them. When discussing
+ this earlier today with Josh Colp, we both agreed that in the
+ majority of cases, this would be enough of a fix. However, we
+ also agreed that we should have a safety net in place in case we
+ never receive a response to our initial INVITE. To handle this, I
+ have modified __sip_autodestruct to behave similar to the way it
+ does in Asterisk 1.4. If there are outstanding packets on the
+ sip_pvt, the needdestroy flag is not set, and the last request on
+ the dialog was either a CANCEL or BYE, then we set the
+ needdestroy flag and reschedule destruction for 10 seconds in the
+ future. If, though, the needdestroy flag is set, then we use
+ __sip_pretend_ack to kill the remaining outstanding packets so
+ that the monitor thread can destroy the sip_pvt. I ran two
+ separate tests. First, I placed a call from my Aastra phone to my
+ Polycom phone. I hung up the Aastra before the Polycom answered.
+ I verified through sip debug output that Asterisk properly ACKed
+ the 487 received from the Polycom. For my second test, I set up a
+ SIPp UAS scenario so that it would send a 200 OK in response to a
+ CANCEL but would not send a 487 for the original INVITE. I
+ verified that after about 40 seconds, Asterisk properly cleans up
+ the outgoing sip_pvt for the call. Review:
+ https://reviewboard.asterisk.org/r/308
+
+2009-07-10 16:23 +0000 [r205804] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: SIP registration auth loop caused by stale
+ nonce If an endpoint sends two registration requests in a very
+ short period of time with the same nonce, both receive 401
+ responses from Asterisk, each with a different nonce (the second
+ 401 containing the current nonce and the first one being stale).
+ If the endpoint responds to the first 401, it does not match the
+ current nonce so Asterisk sends a third 401 with a newly
+ generated nonce (which updates the current nonce)... Now if the
+ endpoint responds to the second 401, it does not match the
+ current nonce either and Asterisk sends a fourth 401 with a newly
+ generated nonce... This loop goes on and on. There appears to be
+ a simple fix for this. If the nonce from the request does not
+ match our nonce, but is a good response to a previous nonce,
+ instead of sending a 401 with a newly generated nonce, use the
+ current one instead. This breaks the loop as the nonce is not
+ updated until a response is received. Additional logic has been
+ added to make sure no nonce can be responded to twice though.
+ (closes issue #15102) Reported by: Jamuel Patches:
+ patch-bug_0015102 uploaded by Jamuel (license 809) nonce_sip.diff
+ uploaded by dvossel (license 671) Tested by: Jamuel Review:
+ https://reviewboard.asterisk.org/r/289/
+
+2009-07-10 15:51 +0000 [r205775] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Ensure that outbound NOTIFY requests are
+ properly routed through stateful proxies. With this change, we
+ make note of Record-Route headers present in any SUBSCRIBE
+ request that we receive so that our outbound NOTIFY requests will
+ have the proper Route headers in them. (closes issue #14725)
+ Reported by: ibc
+
+2009-07-09 23:37 +0000 [r205728] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: No audio on calls from Asterisk to various
+ ISDN devices until DTMF sent by caller. Add missing clearing of
+ the dialing flag when the ISDN call is CONNECTED. (i.e. When
+ libpri generates the event PRI_EVENT_ANSWER.) (closes issue
+ #15420) Reported by: scottbmilne Patches:
+ bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585)
+ Tested by: scottbmilne, alecdavis (closes issue #15416) Reported
+ by: avinoash (closes issue #15389) Reported by: alecdavis This
+ patch should also fix the following issue: (issue #15205)
+ Reported by: vinsik
+
+2009-07-09 16:18 +0000 [r205409-205599] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/time.h: Changing ast_samp2tv to not use floating
+ point.
+
+ * main/rtp.c, channels/chan_iax2.c, include/asterisk/frame.h: Fixes
+ 8khz assumptions Many calculations assume 8khz is the codec rate.
+ This is not always the case. This patch only addresses chan_iax.c
+ and res_rtp_asterisk.c, but I am sure there are other areas that
+ make this assumption as well. Review:
+ https://reviewboard.asterisk.org/r/306/
+
+ * include/asterisk/devicestate.h, main/pbx.c, main/devicestate.c,
+ include/asterisk/pbx.h: moving ast_devstate_to_extenstate to
+ pbx.c from devicestate.c ast_devstate_to_extenstate belongs in
+ pbx.c. This change fixes a compile time error with chan_vpb as
+ well.
+
+2009-07-08 19:26 +0000 [r205349] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Prevent phantom calls to queue members. If a
+ caller were to hang up while a periodic announcement or position
+ were being said, the return value for those functions would
+ incorrectly indicate that the caller was still in the queue. With
+ these changes, the problem does not occur. (closes issue #14631)
+ Reported by: latinsud Patches: queue_announce_ghost_call2.diff
+ uploaded by latinsud (license 745) (with small modification from
+ me)
+
+2009-07-08 18:19 +0000 [r205288] Jason Parker <jparker@digium.com>
+
+ * config.guess, config.sub: Update config.guess and config.sub from
+ the savannah.gnu.org git repo.
+
+2009-07-08 16:53 +0000 [r205215] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/time.h: ast_samp2tv needs floating point for
+ 16khz audio In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate
+ is 16000. The .5 is currently stripped off because we don't
+ calculate using floating points. This causes madness with 16khz
+ audio. (issue ABE-1899) Review:
+ https://reviewboard.asterisk.org/r/305/
+
+2009-07-08 16:26 +0000 [r205188] Tilghman Lesher <tlesher@digium.com>
+
+ * main/say.c: Add redirection warnings for the invalid language
+ codes previously removed.
+
+2009-07-08 15:54 +0000 [r205149] Russell Bryant <russell@digium.com>
+
+ * res/res_crypto.c: Make OpenSSL usage thread-safe. OpenSSL is not
+ thread-safe by default. However, making it thread safe is very
+ easy. We just have to provide a couple of callbacks. One callback
+ returns a thread ID. The other handles locking. For more
+ information, start with the "Is OpenSSL thread-safe?" question on
+ the FAQ page of openssl.org.
+
+2009-07-02 21:59 +0000 [r204834] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: Removed confusing warning message "Got
+ Busy in Connected State" If an incoming mISDN call is answered
+ with the Answer application and a subsequent Dial gets a busy
+ endpoint then it is valid for that already connected channel to
+ get the busy indication. Asterisk will play the busy tones until
+ the dialplan plays something else or hangs up the call. (closes
+ issue #11974) Reported by: fvdb
+
+2009-07-02 18:15 +0000 [r204755] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/devicestate.h, main/pbx.c, main/devicestate.c,
+ include/asterisk/pbx.h: moving device state functions from pbx.h
+ to devicestate.h to sync with other branches
+
+2009-07-02 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.4.26-rc5
+
+2009-07-02 15:05 +0000 [r204681] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/devicestate.h, main/pbx.c, main/devicestate.c,
+ include/asterisk/pbx.h: Improved mapping of extension states from
+ combined device states. This fixes a few issues with incorrect
+ extension states and adds a cli command, core show
+ device2extenstate, to display all possible state mappings.
+ (closes issue #15413) Reported by: legart Patches:
+ exten_helper.diff uploaded by dvossel (license 671) Tested by:
+ dvossel, legart, amilcar Review:
+ https://reviewboard.asterisk.org/r/301/
+
+2009-06-30 20:23 +0000 [r204556] Tilghman Lesher <tlesher@digium.com>
+
+ * main/say.c, UPGRADE.txt: More incorrect language codes, plus
+ ensuring that regionalizations use the specified language, and
+ not English for grammar. (closes issue #15022) Reported by:
+ greenfieldtech Patches: 20090519__issue15022.diff.txt uploaded by
+ tilghman (license 14)
+
+2009-06-30 18:47 +0000 [r204474] Jason Parker <jparker@digium.com>
+
+ * main/say.c: Fix ast_say_counted_noun to correctly handle Polish.
+ Fix a comment typo in passing.
+
+2009-06-30 18:23 +0000 [r204469] Tilghman Lesher <tlesher@digium.com>
+
+ * main/say.c, UPGRADE.txt: "tw" is the language specification for
+ Twi (from Ghana) not Taiwanese. (closes issue #15346) Reported
+ by: volivier Patches: 20090617__issue15346__1.4.diff.txt uploaded
+ by tilghman (license 14) 20090617__issue15346__trunk.diff.txt
+ uploaded by tilghman (license 14)
+ 20090617__issue15346__1.6.0.diff.txt uploaded by tilghman
+ (license 14) 20090617__issue15346__1.6.1.diff.txt uploaded by
+ tilghman (license 14) 20090617__issue15346__1.6.2.diff.txt
+ uploaded by tilghman (license 14) Tested by: volivier
+
+2009-06-29 22:45 +0000 [r204243-204300] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Add error message so that it is clear why a
+ SIP peer was not processed when a DNS lookup fails on a host or
+ outboundproxy. (closes issue #13432) Reported by: p_lindheimer
+ Patches: outboundproxy.patch uploaded by p (license 558)
+
+ * channels/chan_sip.c: Fix build oops.
+
+ * channels/chan_sip.c: Fix a problem where chan_sip would ignore
+ "old" but valid responses. chan_sip has had a problem for quite a
+ long time that would manifest when Asterisk would send multiple
+ SIP responses on the same dialog before receiving a response. The
+ problem occurred because chan_sip only kept track of the highest
+ outgoing sequence number used on the dialog. If Asterisk sent two
+ requests out, and a response arrived for the first request sent,
+ then Asterisk would ignore the response. The result was that
+ Asterisk would continue retransmitting the requests and ignoring
+ the responses until the maximum number of retransmissions had
+ been reached. The fix here is to rearrange the code a bit so that
+ instead of simply comparing the sequence number of the response
+ to our latest outgoing sequence number, we walk our list of
+ outstanding packets and determine if there is a match. If there
+ is, we continue. If not, then we ignore the response. In doing
+ this, I found a few completely useless variables that I have now
+ removed. (closes issue #11231) Reported by: flefoll Review:
+ https://reviewboard.asterisk.org/r/298
+
+2009-06-29 19:36 +0000 [r204170] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_odbc.c, funcs/func_strings.c: Revision 189537 was
+ supposed to make 1.4 more correct. Instead, it broke func_odbc.
+ Reverting. (closes issue #15317, issue #14614)
+
+2009-06-29 17:04 +0000 [r204067] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: segfault after SPINLOCK schedule delete
+ Using the SPINLOCK schedule delete macro can result in the
+ iax_pvt lock being given up. This makes it possible for the
+ iax_pvt to dissappear when we thought we held the mutex the
+ entire time. To resolve this, the iax_pvt's ref count is
+ incremented. (closes issue #15377) Reported by: aragon Patches:
+ iax_spin_issue_1.4.diff uploaded by dvossel (license 671) Tested
+ by: aragon, dvossel
+
+2009-06-29 15:04 +0000 [r204012] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_mixmonitor.c: Place unlock of mutex in an else block so
+ that it does not get unlocked twice. (closes issue #15400)
+ Reported by: aragon
+
+2009-06-27 00:55 +0000 [r203908] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: The ISDN CPE side should not exclusively
+ pick B channels normally. Before this patch, Asterisk
+ unconditionally picked B channels exclusively on the CPE side and
+ normally allowed alternative B channels on the network side. Now
+ Asterisk does the opposite. Reasons for the CPE side to normally
+ not pick B channels exclusively: * For CPE point-to-multipoint
+ mode (i.e. phone side), the CPE side does not have enough
+ information to exclusively pick B channels. (There may be other
+ devices on the line.) * Q.931 gives preference to the network
+ side picking B channels. * Some telcos require the CPE side to
+ not pick B channels exclusively. (closes issue #14383) Reported
+ by: mbrancaleoni
+
+2009-06-26 22:09 +0000 [r203848] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Make sure to recreate the dahdi pseudo
+ channel after dahdi restart (closes issue #14477) Reported by:
+ timking
+
+2009-06-26 21:16 +0000 [r203785] Russell Bryant <russell@digium.com>
+
+ * main/file.c: Don't fast forward past the end of a message. This
+ is nice change for users of the voicemail application. If someone
+ gets a little carried away with fast forwarding through a
+ message, they can easily get to the end and accidentally exit the
+ voicemail application by hitting the fast forward key during the
+ following prompt. This adds some safety by not allowing a fast
+ forward past the end of a message. (closes issue #14554) Reported
+ by: lacoursj Patches: 21761.patch uploaded by lacoursj (license
+ 707) Tested by: lacoursj
+
+2009-06-26 20:03 +0000 [r203719] David Brooks <dbrooks@digium.com>
+
+ * apps/app_voicemail.c: Fixing voicemail's error in checking max
+ silence vs min message length Max silence was represented in
+ milliseconds, yet vmminsecs (minmessage) was represented as
+ seconds. Also, the inequality was reversed. The warning, if
+ triggered, was "Max silence should be less than minmessage or you
+ may get empty messages", which should have been logged if max
+ silence was greater than minmessage, but the check was for less
+ than. Also, conforming if statement to coding guidelines. closes
+ issue #15331) Reported by: markd Review:
+ https://reviewboard.asterisk.org/r/293/
+
+2009-06-25 21:13 +0000 [r203380] Terry Wilson <twilson@digium.com>
+
+ * main/cli.c: I didn't see that Mark already fixed the underlying
+ issue! Yay for removing useless code.
+
+2009-06-25 21:02 +0000 [r203375] Russell Bryant <russell@digium.com>
+
+ * res/res_features.c: Fix a case where CDR answer time could be
+ before the start time involving parking. (closes issue #13794)
+ Reported by: davidw Patches: 13794.patch uploaded by murf
+ (license 17) 13794.patch.160 uploaded by murf (license 17) Tested
+ by: murf, dbrooks
+
+2009-06-25 20:09 +0000 [r203311] Terry Wilson <twilson@digium.com>
+
+ * main/cli.c: Don't try to free NULL
+
+2009-06-25 18:52 +0000 [r203230] Mark Michelson <mmichelson@digium.com>
+
+ * main/astmm.c: Prevent false positives when freeing a NULL pointer
+ with MALLOC_DEBUG enabled.
+
+2009-06-25 16:02 +0000 [r203115] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Resolve a crash related to a T.38 reinvite
+ race condition. This change resolves a crash observed locally
+ during some T.38 testing. A call was set up using a call file,
+ and when the T.38 reinvite came in, the channel state was still
+ AST_STATE_DOWN. The reason is explained by a comment in the code
+ that previously lived in the handling of AST_STATE_RINGING. This
+ change modifies the logic to handle the same race condition for
+ any channel state that is not UP. (closes ABE-1895)
+
+2009-06-24 21:01 +0000 [r203036] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Improved chan_dahdi.conf pritimer error
+ checking. Valid format is: pritimer=timer_name,timer_value *
+ Fixed segfault if the ',' is missing. * Completely check the
+ range returned by pri_timer2idx() to prevent possible access
+ outside array bounds.
+
+2009-06-24 18:28 +0000 [r202966] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Use the handy UNLINK macro instead of
+ hand-coding the same thing in-line.
+
+2009-06-24 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.4.26-rc4
+
+2009-06-23 16:28 +0000 [r202671] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: MWI NOTIFY contains a wrong URI if Asterisk
+ listens to non-standard port and transport (closes issue #14659)
+ Reported by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt
+ uploaded by klaus3000 (license 65) mwi_port-transport_trunk.diff
+ uploaded by dvossel (license 671) Tested by: dvossel, klaus3000
+ Review: https://reviewboard.asterisk.org/r/288/
+
+2009-06-23 15:22 +0000 [r202572-202601] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix more memory leaks that may result if rtp
+ is not successfully allocated.
+
+ * channels/chan_sip.c: Fix potential memory leak in chan_sip when
+ video rtp is not allocated properly.
+
+2009-06-22 20:08 +0000 [r202414-202496] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Report CallerID change during a masquerade.
+ Reported by: markster
+
+ * channels/chan_sip.c: Make Polycom subscription type override
+ check more explicit.
+
+2009-06-22 14:44 +0000 [r202336-202342] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Remove an extra debug line left from
+ previous commit.
+
+ * channels/chan_sip.c: Fix a situation in which Asterisk would not
+ stop retransmitting 487s. If a CANCEL were received by Asterisk,
+ we would send a 487 in response to the original INVITE and a 200
+ OK for the CANCEL. If there were a network hiccup which caused
+ the 200 OK and the 487 to be lost, then the UA communicating with
+ Asterisk may try to retransmit its CANCEL. Asterisk's response to
+ this used to be to try sending another 487 to the canceled INVITE
+ and another 200 OK to the CANCEL. The problem here is that the
+ originally-sent 487 was sent "reliably" meaning that it will be
+ retransmitted until it is received properly. So when we receive
+ the second CANCEL it is likely that the first batch of 487s we
+ sent is still going strong and reaches the UA. The result was
+ that the second set of 487s would be retransmitted constantly
+ until the maximum number of retries had been reached. The fix for
+ this is that if we receive a second CANCEL for an INVITE, then we
+ cancel the retransmission of the first set of 487s and start a
+ second set. This causes the dialog to be terminated reasonably.
+ (closes issue #14584) Reported by: klaus3000 Patches:
+ 14584_v2.patch uploaded by mmichelson (license 60) Tested by:
+ klaus3000
+
+ * channels/chan_sip.c: Fix a possible infinite loop in SDP parsing
+ during glare situation. There was a while loop in
+ get_ip_and_port_from_sdp which was controlled by a call to
+ get_sdp_iterate. The loop would exit either if what we were
+ searching for was found or if the return was NULL. The problem is
+ that get_sdp_iterate never returns NULL. This means that if what
+ we were searching for was not present, the loop would run
+ infinitely. This modification of the loop fixes the problem.
+ (closes issue #15213) Reported by: schmidts (closes issue #15349)
+ Reported by: samy (closes issue #14464) Reported by: pj (closes
+ issue #15345) Reported by: aragon Patches: sip_inf_loop.patch
+ uploaded by mmichelson (license 60) Tested by: aragon
+
+2009-06-20 17:51 +0000 [r202153] Sean Bright <sean@malleable.com>
+
+ * channels/chan_sip.c: Since we don't have sip_pvt_lock() in 1.4,
+ we need to use ast_mutex_* directly. (closes issue #15366)
+ Reported by: loloski
+
+2009-06-19 21:21 +0000 [r202022] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Added deadlock protection to
+ try_suggested_sip_codec in chan_sip.c. Review:
+ https://reviewboard.asterisk.org/r/287/
+
+2009-06-19 20:22 +0000 [r201993] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: timestamp was being converted to host order
+ as a short rather than a long (closes issue #15361) Reported by:
+ ffloimair Patches: ts_issue.diff uploaded by dvossel (license
+ 671)
+
+2009-06-19 00:40 +0000 [r201828] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_features.c: If the "h" extension fails, give it another
+ chance in main/pbx.c. If the "h" extension fails, give it another
+ chance in main/pbx.c, when it returns from the bridge code. Fixes
+ an issue where the "h" extension may occasionally not fire, when
+ a Dial is executed from a Macro. Debugged in #asterisk with user
+ tompaw.
+
+2009-06-18 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.4.26-rc3
+
+2009-06-18 15:24 +0000 [r201600] Russell Bryant <russell@digium.com>
+
+ * res/res_musiconhold.c: Fix memory corruption and leakage related
+ reloads of non files mode MoH classes. For Music on Hold classes
+ that are not files mode, meaning that we are executing an
+ application that will feed us audio data, we use a thread to
+ monitor the external application and read audio from it. This
+ thread also makes use of the MoH class object. In the MoH class
+ destructor, we used pthread_cancel() to ask the thread to exit.
+ Unfortunately, the code did not wait to ensure that the thread
+ actually went away. What needed to be done is a pthread_join() to
+ ensure that the thread fully cleans up before we proceed. By
+ adding this one line, we resolve two significant problems: 1)
+ Since the thread was never joined, it never fully goes away. So,
+ on every reload of non-files mode MoH, an unused thread was
+ sticking around. 2) There was a race condition here where the
+ application monitoring thread could still try to access the MoH
+ class, even though the thread executing the MoH reload has
+ already destroyed it. (issue #15109) Reported by: jvandal (issue
+ #15123) Reported by: axisinternet (issue #15195) Reported by:
+ amorsen (issue AST-208)
+
+2009-06-17 19:59 +0000 [r201450] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c: Change the datastore traversal in
+ ast_do_masquerade to use a safe list traversal. It is possible
+ for datastore fixup functions to remove the datastore from the
+ list and free it. In particular, the queue_transfer_fixup in
+ app_queue does this. While I don't yet know of this causing any
+ crashes, it certainly could. Found while discussing a separate
+ issue with Brian Degenhardt.
+
+2009-06-17 19:28 +0000 [r201423] David Vossel <dvossel@digium.com>
+
+ * apps/app_mixmonitor.c: StopMixMonitor race condition (not giving
+ up file immediately) StopMixMonitor only indicates to the
+ MixMonitor thread to stop writing to the file. It does not
+ guarantee that the recording's file handle is available to the
+ dialplan immediately after execution. This results in a race
+ condition. To resolve this, the filestream pointer is placed in a
+ datastore on the channel. When StopMixMonitor is called, the
+ datastore is retrieved from the channel and the filestream is
+ closed immediately before returning to the dialplan.
+ Documentation indicating the use of StopMixMonitor to free files
+ has been updated as well. (closes issue #15259) Reported by:
+ travisghansen Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/283/
+
+2009-06-17 18:45 +0000 [r201380] David Brooks <dbrooks@digium.com>
+
+ * channels/chan_sip.c: Checks for NULL sip_pvt pointer in
+ chan_sip.c->acf_channel_read() Zombie channels could be passed,
+ and chan_sip.c wasn't checking for it. Could crash Asterisk. Now
+ checking for NULL pointer. (closes issue #15330) Reported by:
+ okrief Tested by: dbrooks
+
+2009-06-17 12:03 +0000 [r200991-201261] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/linkedlists.h: Correct AST_LIST_APPEND_LIST
+ behavior when list to be appended is empty. When the list to be
+ appended is empty, and the list to be appended to is *not*,
+ AST_LIST_APPEND_LIST would actually cause the target list to
+ become broken, and no longer have a pointer to its last entry.
+ This patch fixes the problem. (reported by Stanislaw Pitucha on
+ the asterisk-dev mailing list)
+
+ * apps/app_chanspy.c, apps/app_mixmonitor.c, main/channel.c,
+ build_tools/cflags-devmode.xml, main/autoservice.c, main/frame.c,
+ apps/app_meetme.c, main/slinfactory.c,
+ include/asterisk/linkedlists.h, main/file.c,
+ include/asterisk/channel.h, include/asterisk/frame.h: Improve
+ support for media paths that can generate multiple frames at
+ once. There are various media paths in Asterisk (codec
+ translators and UDPTL, primarily) that can generate more than one
+ frame to be generated when the application calling them expects
+ only a single frame. This patch addresses a number of those
+ cases, at least the primary ones to solve the known problems. In
+ addition it removes the broken TRACE_FRAMES support, fixes a
+ number of bugs in various frame-related API functions, and cleans
+ up various code paths affected by these changes.
+ https://reviewboard.asterisk.org/r/175/
+
+2009-06-16 13:25 +0000 [r200875] Eliel C. Sardanons <eliels@gmail.com>
+
+ * res/res_smdi.c: Show the interface name on error, if it is not
+ found. If the smdiport specified is not found, show the interface
+ name instead of '(null)'.
+
+2009-06-15 15:21 +0000 [r200513] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Add INFO to our allowed methods so that
+ endpoints know they may send it to us. AST-223
+
+2009-06-12 19:06 +0000 [r200360] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c: Suppress a warning message and give a better
+ return code when generating inband ringing after a call is
+ answered. (closes issue #15158) Reported by: madkins Patches:
+ 15158.patch uploaded by mmichelson (license 60) Tested by:
+ madkins
+
+2009-06-11 22:20 +0000 [r200185] Sean Bright <sean.bright@gmail.com>
+
+ * Makefile: Backport fix for parallel build warnings from trunk
+ r199781.
+
+2009-06-11 12:12 +0000 [r200037] Leif Madsen <lmadsen@digium.com>
+
+ * build_tools/make_version_h: Fix path for .flavor and .version.
+ (issue #14737) Reported by: davidw Patches: flavor.patch uploaded
+ by davidw (license 780) Tested by: davidw
+
+2009-06-10 16:08 +0000 [r199856] Sean Bright <sean.bright@gmail.com>
+
+ * include/asterisk/utils.h: __WORDSIZE is not available on all
+ platforms, so use sizeof(void *) instead.
+
+2009-06-09 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.4.26-rc2
+
+2009-06-08 19:28 +0000 [r199626-199628] Sean Bright <sean.bright@gmail.com>
+
+ * include/asterisk/utils.h: Fix a typo in the stack size
+ calculation just introduced.
+
+ * include/asterisk/utils.h: Increase the size of our thread stack
+ on 64 bit processors. We were setting the stack size for each
+ thread to 240KB regardless of architecture, which meant that in
+ some scenarios we actually had less available stack space on 64
+ bit processors (pointers use 8 bytes instead of 4). So now we
+ calculate the stack size we reserve based on the platform's
+ __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128
+ bit -> 1008KB (that's right, we're ready for 128 bit processors)
+ Patch typed by me but written by several members of
+ #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes
+ issue #14932) Reported by: jpiszcz Patches:
+ 06052009_issue14932.patch uploaded by seanbright (license 71)
+ Tested by: seanbright
+
+2009-06-05 21:19 +0000 [r199297] David Vossel <dvossel@digium.com>
+
+ * main/pbx.c: Fixes issue with hints giving unexpected results.
+ Hints with two or more devices that include ONHOLD gave
+ unexpected results. (closes issue #15057) Reported by:
+ p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel
+ (license 671) pbx.c.1.4.patch uploaded by p (license 558)
+ devicestate.c.trunk.patch uploaded by p (license 671) Tested by:
+ p_lindheimer, dvossel Review:
+ https://reviewboard.asterisk.org/r/254/
+
+2009-06-04 19:00 +0000 [r199138] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: Additional updates to AST-2009-001
+
+2009-06-04 14:14 +0000 [r198957-199022] Sean Bright <sean.bright@gmail.com>
+
+ * main/asterisk.c, main/loader.c, include/asterisk.h: Safely handle
+ AMI connections/reload requests that occur during startup. During
+ asterisk startup, a lock on the list of modules is obtained by
+ the primary thread while each module is initialized. Issue 13778
+ pointed out a problem with this approach, however. Because the
+ AMI is loaded before other modules, it is possible for a module
+ reload to be issued by a connected client (via Action: Command),
+ causing a deadlock. The resolution for 13778 was to move
+ initialization of the manager to happen after the other modules
+ had already been lodaded. While this fixed this particular issue,
+ it caused a problem for users (like FreePBX) who call AMI scripts
+ via an #exec in a configuration file (See issue 15189). The
+ solution I have come up with is to defer any reload requests that
+ come in until after the server is fully booted. When a call comes
+ in to ast_module_reload (from wherever) before we are fully
+ booted, the request is added to a queue of pending requests. Once
+ we are done booting up, we then execute these deferred requests
+ in turn. Note that I have tried to make this a bit more
+ intelligent in that it will not queue up more than 1 request for
+ the same module to be reloaded, and if a general reload request
+ comes in ('module reload') the queue is flushed and we only issue
+ a single deferred reload for the entire system. As for how this
+ will impact existing installations - Before 13778, a reload
+ issued before module initialization was completed would result in
+ a deadlock. After 13778, you simply couldn't connect to the
+ manager during startup (which causes problems with
+ #exec-that-calls-AMI configuration files). I believe this is a
+ good general purpose solution that won't negatively impact
+ existing installations. (closes issue #15189) (closes issue
+ #13778) Reported by: p_lindheimer Patches:
+ 06032009_15189_deferred_reloads.diff uploaded by seanbright
+ (license 71) Tested by: p_lindheimer, seanbright Review:
+ https://reviewboard.asterisk.org/r/272/
+
+ * pbx/pbx_spool.c: Fix a possible crash in pbx_spool. We were
+ trying to reference members of a struct that had previously been
+ freed. This patch makes sure that we free the struct after it has
+ been removed from the spooler queue. (closes issue #15072)
+ Reported by: garlew Patches: spool.diff uploaded by garlew
+ (license 376)
+
+2009-06-03 15:49 +0000 [r198891] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, res/res_features.c, include/asterisk/channel.h:
+ Generic call forward api, ast_call_forward() The function
+ ast_call_forward() forwards a call to an extension specified in
+ an ast_channel's call_forward string. After an ast_channel is
+ called, if the channel's call_forward string is set this function
+ can be used to forward the call to a new channel and terminate
+ the original one. I have included this api call in both
+ channel.c's ast_request_and_dial() and res_feature.c's
+ feature_request_and_dial(). App_dial and app_queue already
+ contain call forward logic specific for their application and
+ options. (closes issue #13630) Reported by: festr Review:
+ https://reviewboard.asterisk.org/r/271/
+
+2009-06-01 20:07 +0000 [r198665] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_musiconhold.c: If using the old deprecated format, a
+ reload would cause the class to disappear. (closes issue #14759)
+ Reported by: lidocaineus Patches: 20090518__issue14759.diff.txt
+ uploaded by tilghman (license 14) Tested by: lmadsen
+
+2009-05-30 19:36 +0000 [r198370] Sean Bright <sean.bright@gmail.com>
+
+ * res/res_jabber.c: Properly terminate AMI JabberSend response
+ messages. The response message (either Error or Success) needs an
+ extra trailing \r\n after the fields to inform the client that
+ the message is complete. (closes issue #14876) Reported by: srt
+ Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright
+ (license 71) asterisk_14876.patch uploaded by srt (license 378)
+ trunk-14876-2.diff uploaded by phsultan (license 73)
+
+2009-05-30 03:42 +0000 [r198311] Russell Bryant <russell@digium.com>
+
+ * res/res_smdi.c: Fix a crash that occurred when MWI SMDI messages
+ expired. (closes issue #14561) Reported by: cmoss28
+
+2009-05-30 02:46 +0000 [r198251] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_dial.c: Treat an empty FORWARD_CONTEXT the same way we
+ treat a missing one. (closes issue #15056) Reported by:
+ p_lindheimer Patches: 05292009_bug15056.diff uploaded by
+ seanbright (license 71) Tested by: p_lindheimer
+
+2009-05-29 18:53 +0000 [r198068] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/cdr.c, main/channel.c, res/res_features.c,
+ include/asterisk/cdr.h: Use AST_CDR_NOANSWER instead of
+ AST_CDR_NULL as the default CDR disposition. This change also
+ involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is
+ used on originated channels to distinguish: them from dialed
+ channels. (closes issue #12946) Reported by: meral Patches:
+ null-cdr2.diff uploaded by mnicholson (license 96) Tested by:
+ mnicholson, dbrooks (closes issue #15122) Reported by: sum Tested
+ by: sum
+
+2009-05-29 18:14 +0000 [r197998] Sean Bright <sean.bright@gmail.com>
+
+ * Makefile: Fix 'make config' target for Slackware. There was a
+ missing semi-colon after the echo statement in the Makefile that
+ was causing problems for some users. Fix suggested by reporter.
+ (closes issue #15225) Reported by: pdavis
+
+2009-05-28 23:57 +0000 [r197895] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_mixmonitor.c: Update MixMonitor documentation. Updated
+ the MixMonitor documentation for the 'b' option so that it is
+ more obvious that you must not optimize awat the Local channel
+ when using this option. (issue #14829)
+
+2009-05-28 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.4.26-rc1
+
+2009-05-28 15:51 +0000 [r197620] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: 'iax show peer blah' now outputs whether or
+ not peer 'blah' is in trunk mode or not.
+
+2009-05-28 15:27 +0000 [r197588] Mark Michelson <mmichelson@digium.com>
+
+ * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Allow
+ for media to arrive from an alternate source when responding to a
+ reinvite with 491. When we receive a SIP reinvite, it is possible
+ that we may not be able to process the reinvite immediately since
+ we have also sent a reinvite out ourselves. The problem is that
+ whoever sent us the reinvite may have also sent a reinvite out to
+ another party, and that reinvite may have succeeded. As a result,
+ even though we are not going to accept the reinvite we just
+ received, it is important for us to not have problems if we
+ suddenly start receiving RTP from a new source. The fix for this
+ is to grab the media source information from the SDP of the
+ reinvite that we receive. This information is passed to the RTP
+ layer so that it will know about the alternate source for media.
+ Review: https://reviewboard.asterisk.org/r/252
+
+2009-05-28 15:21 +0000 [r197562] Eliel C. Sardanons <eliels@gmail.com>
+
+ * channels/chan_sip.c: Use the address we already know when
+ reloading a peer with nat=yes. If we already have an address for
+ a peer, and we are reloading the sip configuration, try to use
+ that address to contact the peer, instead of getting it from the
+ Contact. (closes issue #15194) Reported by: ibc Patches:
+ sip.patch uploaded by eliel (license 64) Tested by: manwe
+
+2009-05-28 14:49 +0000 [r197537] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c, include/asterisk/audiohook.h,
+ main/audiohook.c: Add flags to chanspy audiohook so that audio
+ stays in sync. There are two flags being added to the chanspy
+ audiohook here. One is the pre-existing
+ AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that
+ the read and write slinfactories on the audiohook do not skew
+ beyond a certain tolerance. In addition, there is a new audiohook
+ flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set,
+ we do not allow for a slinfactory to build up a substantial
+ amount of audio before flushing it. For this particular issue,
+ this means that the person spying on the call will hear the
+ conversations in real time with very little delay in the audio.
+ (closes issue #13745) Reported by: geoffs Patches: 13745.patch
+ uploaded by mmichelson (license 60) Tested by: snblitz
+
+2009-05-28 13:44 +0000 [r197466] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix a bug where the flag indicating the
+ presence of rport would get overwritten by the nat setting. The
+ presence of rport is now stored as a separate flag. Once the
+ dialog is setup and authenticated (or it passes through
+ unauthenticated) the proper nat flag is set. (closes issue
+ #13823) Reported by: dimas
+
+2009-05-27 20:12 +0000 [r197264] Sean Bright <sean.bright@gmail.com>
+
+ * Makefile: Use bash explicitly when calling
+ build_tools/mkpkgconfig from the Makefile. Since we use bashisms
+ in build_tools/mkpkgconfig, we should call on bash explicitly
+ when running from the Makefile, otherwise we get errors during a
+ 'make install.' (closes issue #15209) Reported by: seandarcy
+
+2009-05-27 20:07 +0000 [r197259] Olle Johansson <oej@edvina.net>
+
+ * doc/asterisk-conf.txt: Typo fix
+
+2009-05-27 19:09 +0000 [r197194] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_cut.c: Use a different determinator on whether to
+ print the delimiter, since leading fields may be blank. (closes
+ issue #15208) Reported by: ramonpeek Patch by me, though inspired
+ in part by a patch from ramonpeek
+
+2009-05-27 16:49 +0000 [r197124] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, include/asterisk/channel.h: Fix broken attended
+ transfers The bridge was terminating immediately after the
+ attended transfer was completed. The problem was because upon
+ reentering ast_channel_bridge nexteventts was checked to see if
+ it was set and if so could possibly return AST_BRIDGE_COMPLETE.
+ (closes issue #15183) Reported by: andrebarbosa Tested by:
+ andrebarbosa, tootai, loloski
+
+2009-05-27 13:54 +0000 [r197024] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_queue.c: Fix handling of the 'state_interface' option of
+ the 'queue add member' CLI command. This change relates to
+ r184980, which was a backport of the state interface changes to
+ app_queue from trunk. trunk and all of the 1.6.x branches are not
+ affected. 'queue add member' allows for specifying an interface
+ to use for device state when adding a queue member via CLI, but
+ the validation code was not properly updated to reflect this
+ optional argument. (closes issue #15198) Reported by: loloski
+ Patches: 05272009_app_queue.diff uploaded by seanbright (license
+ 71) Tested by: loloski
+
+2009-05-26 18:14 +0000 [r196826] Russell Bryant <russell@digium.com>
+
+ * res/res_convert.c: Resolve a file handle leak. The frames here
+ should have always been freed. However, out of luck, there was
+ never any memory leaked. However, after file streams became
+ reference counted, this code would leak the file stream for the
+ file being read. (closes issue #15181) Reported by: jkroon
+
+2009-05-26 13:06 +0000 [r196657] Joshua Colp <jcolp@digium.com>
+
+ * contrib/scripts/safe_asterisk: Remove some bash specific stuff
+ from safe_asterisk. (closes issue #10812) Reported by: paravoid
+ Patches: safe_asterisk_bashism.diff uploaded by tzafrir (license
+ 46)
+
+2009-05-22 13:54 +0000 [r196116] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_misdn.c: Fix a bug where using immediate with mISDN
+ caused a cause code of 16 to get sent back instead of 1 if the
+ 's' extension did not exist. (closes issue #12286) Reported by:
+ lmamane
+
+2009-05-21 19:04 +0000 [r195991] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: Sign problem calculating timestamp for iax
+ frame leads to no audio on the receiving peer. There are rare
+ cases in which a frame's delivery timestamp is slightly less than
+ the iax2_pvt's offset. This causes the pvt's timestamp to be a
+ small negative number, but since the timestamp value is unsigned
+ it looks like a huge positive number. This patch checks for this
+ negative case and sets the ms to zero. A similar check is already
+ done right below this one in the 'else' statement. (closes issue
+ #15032) Reported by: guillecabeza Patches:
+ chan_iax2.c.patch_timestamp uploaded by guillecabeza (license
+ 380) Tested by: guillecabeza (closes issue #14216) Reported by:
+ Andrey Sofronov
+
+2009-05-21 15:25 +0000 [r195881] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/cdr.c, res/res_features.c, include/asterisk/cdr.h: This
+ commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and
+ AST_CDR_FLAG_LOCKED from being updated in certain cases. This is
+ accomplished by adding two functions to update the answer time
+ and disposition of calls that checks for the proper lock flags.
+ These functions are used in the ast_bridge_call() function so
+ that ForkCDR(A) calls are respected. This patch also modifies the
+ way ast_bridge_call() chooses the cdr record to base the
+ bridged_cdr on. Previously the first unlocked cdr record would be
+ chosen, now instead the first cdr record is chosen and forked cdr
+ records are moved to the bridge_cdr. This allows the original cdr
+ record and any forked cdr records to be properly updated with
+ answer and end times. (closes issue #13797) Reported by: sh0t
+ Tested by: sh0t (closes issue #14744) Reported by: deepesh
+
+2009-05-21 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.4.25
+
+2009-05-13 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.4.25-rc1
+
+2009-05-13 13:38 +0000 [r194208] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: Fix RFC2833 issues with DTMF getting duplicated and
+ with duration wrapping over. (closes issue #14815) Reported by:
+ geoff2010 Patches: v1-14815.patch uploaded by dimas (license 88)
+ Tested by: geoff2010, file, dimas, ZX81, moliveras (closes issue
+ #14460) Reported by: moliveras Tested by: moliveras
+
+2009-05-13 00:52 +0000 [r194137] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Fix logic for how to proceed with a single digit
+ extension. (closes issue #15091) Reported by: andrew Patches:
+ 20090512__issue15091.diff.txt uploaded by tilghman (license 14)
+ Tested by: andrew
+
+2009-05-12 22:15 +0000 [r194028] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_queue.c: This change modifies app_queue to properly
+ generate CDR records in failure situations. This involves setting
+ a proper cdr disposition coresponding to the given failure
+ condition and ensuring the proper information is stored in the
+ cdr record. (closes issue #13691) Reported by: dferrer Tested by:
+ mnicholson (closes issue #13637) Reported by: atis Tested by:
+ atis
+
+2009-05-12 20:39 +0000 [r193955] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Avoid initializing routines if the
+ authentication fails. Fixes a crash (RR) issue. (closes issue
+ #14508) Reported by: tiziano Patches:
+ 20090221_2_wrongmailbox.diff.txt uploaded by tiziano (license
+ 377)
+
+2009-05-12 18:18 +0000 [r193880] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Set the invitestate to INV_CANCELLED only if
+ we are actually sending a SIP CANCEL. The problem was that the
+ hangup code was setting the invitestate too early. The result of
+ this was that we would always send a CANCEL request, even if it
+ was not an appropriate time to do so (e.g. we have not yet
+ received a provisional response for our INVITE). Note that this
+ same fix had been applied to trunk and the 1.6.X branches
+ starting with revision 155467. This is why you will see this
+ revision being blocked from those places. AST-216
+
+2009-05-11 22:48 +0000 [r193755] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Move 300 bytes around on the stack, to make
+ more room for an extension buffer. This allows more concurrent
+ extensions to be copied for a single voicemail, without creating
+ a possibility of upsetting existing users, where a dialplan could
+ run out of stack space where it had run fine before.
+ Alternatively, we could have allocated off the heap, but that is
+ a larger change and would have increased the chance for
+ instability introduced by this change. This is really solved
+ starting in 1.6.0.11, as the use of an ast_str buffer allows an
+ unlimited number of extensions (up to available memory). We
+ additionally create a new warning message when the buffer length
+ is exceeded, permitting administrators to see an issue after the
+ fact, whereas previously the list was silently truncated. (closes
+ issue #14739) Reported by: p_lindheimer Patches:
+ 20090417__bug14739.diff.txt uploaded by tilghman (license 14)
+ Tested by: p_lindheimer
+
+2009-05-11 19:09 +0000 [r193613] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: Sent wrong message to clear a call we
+ started if the other end has not responed yet. In the state
+ MISDN_CALLING (i.e. SETUP was sent but no answer has arrived
+ yet), it is not allowed to clear the call with RELEASE_COMPLETE.
+ It must be cleared with DISCONNECT. A RELEASE_COMPLETE is only
+ allowed as an answer to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a,
+ 5.3.2.b) Patches: chan-misdn-ccstate7.patch uploaded by customer.
+ JIRA ABE-1862
+
+2009-05-11 17:35 +0000 [r193544] Leif Madsen <lmadsen@digium.com>
+
+ * funcs/func_channel.c: Document CHANNEL(transfercapability) in CLI
+ documentation. (issue #15073) Reported by: pkempgen Patches:
+ 20090511__issue15073.diff.txt uploaded by tilghman (license 14)
+
+2009-05-08 21:01 +0000 [r193391] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c: Set the proper disposition on originated calls.
+ (closes issue #14167) Reported by: jpt Patches:
+ call-file-missing-cdr2.diff uploaded by mnicholson (license 96)
+ Tested by: dlotina, rmartinez, mnicholson
+
+2009-05-08 14:51 +0000 [r193262] David Vossel <dvossel@digium.com>
+
+ * channels/misdn_config.c: "misdn show config" segfaults asterisk,
+ if no MSN lists (closes issue #14976) Reported by: alecdavis
+ Patches: misdn_config.diff.txt uploaded by alecdavis (license
+ 585) Tested by: alecdavis, FabienToune
+
+2009-05-08 14:03 +0000 [r193193] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configs/logger.conf.sample, main/logger.c: Make absolute paths
+ for logger channels work properly (Note: This is not a new
+ feature, it was previously undocumented and broken.) The Asterisk
+ logger has a feature to support absolute pathnames for logger
+ channels, but the code implementing the feature was broken. This
+ has been fixed, and the absolute path feature is now documented
+ in the sample logger.conf.
+
+2009-05-07 23:41 +0000 [r193119] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Fix Background within a Macro for FreePBX. If the
+ single digit DTMF is an extension in the specified context, then
+ go there and signal no DTMF. Otherwise, we should exit with that
+ DTMF. If we're in Macro, we'll exit and seek that DTMF as the
+ beginning of an extension in the Macro's calling context. If
+ we're not in Macro, then we'll simply seek that extension in the
+ calling context. Previously, someone complained about the
+ behavior as it related to the interior of a Gosub routine, and
+ the fix (#14011) inadvertently broke FreePBX (#14940). This
+ change should fix both of these situations, but with the possible
+ incompatibility that if a single digit extension does not exist
+ (but a longer extension COULD have matched), it would have
+ previously gone immediately to the "i" extension, but will now
+ need to wait for a timeout. (closes issue #14940) Reported by:
+ p_lindheimer Patches: 20090420__bug14940.diff.txt uploaded by
+ tilghman (license 14) Tested by: p_lindheimer
+
+2009-05-07 22:17 +0000 [r193050] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: Give a more helpful message when an
+ incoming call's dialed extension does not match. Added the dialed
+ extension and context to the chan_misdn messages warning that the
+ dialed number cannot be matched in the dialplan.
+
+2009-05-07 16:29 +0000 [r192932] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Eliminate repetition of fullcontact during
+ reconstruction. If the fullcontact field appears in both the
+ sippeers and the sipregs table, then during reconstruction of the
+ field, it will otherwise be doubled. (closes issue #14754)
+ Reported by: Alexei Gradinari Patches:
+ 20090506__bug14754.diff.txt uploaded by tilghman (license 14)
+ Tested by: lmadsen
+
+2009-05-06 22:15 +0000 [r192858] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_features.c: Make ParkedCall application stop execution of
+ the dialplan after hang up Just changed park_exec to always
+ return non-zero. I really wasn't entirely sure at first if this
+ was a bug. Decided it was since it would be surprising when not
+ using ParkedCall in the dialplan to hang up and have dialplan
+ execution continue. (closes issue #14555) Reported by:
+ francesco_r
+
+2009-05-06 13:30 +0000 [r192633] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Update some old logic to stop both begin and
+ end DTMF frames from reaching the core if rfc2833 is not enabled.
+ (closes issue #15036) Reported by: dimas Patches: v1-15036.patch
+ uploaded by dimas (license 88)
+
+2009-05-05 19:56 +0000 [r192524] Sean Bright <sean.bright@gmail.com>
+
+ * static-http/astman.js: Fix Javascript error when using astman.js
+ in Internet Explorer. Internet Explorer (tested with 7.0) does
+ not like trailing commas on constructs like object initializers,
+ so get rid of them to avoid some errors. (closes issue #15026)
+ Reported by: rajnishgiri Patches: bug15026.patch uploaded by
+ seanbright (license 71) Tested by: seanbright
+
+2009-05-05 18:22 +0000 [r192429-192454] Joshua Colp <jcolp@digium.com>
+
+ * res/res_features.c: Fix an incorrect assumption that certain
+ values on the channel will always exist when they may not. The
+ CDR code involved with bridges wrongly assumed that the currently
+ executing application and data values will always exist. It is
+ possible for this to be false when call forwarding is involved.
+ (closes issue #14984) Reported by: gincantalupo
+
+ * apps/app_followme.c: Fix a bug where the followme application
+ would continue trying numbers after the caller hung up. (closes
+ issue #13624) Reported by: sgenyuk
+
+2009-05-04 22:37 +0000 [r192213] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: global mohinterpret setting is ignored
+ mohinterpret and mohsuggest global variables were not copied over
+ during build_users and build_peers. (closes issue #14728)
+ Reported by: dimas Patches: v1-14728.patch uploaded by dimas
+ (license 88) Tested by: dimas, dvossel
+
+2009-05-02 18:48 +0000 [r191628-191778] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Fix a bug which resulted from the Hebrew
+ voicemail commit. This fixes a case where a certain message could
+ get played twice. (closes issue #13155) Reported by:
+ greenfieldtech Patches: app_voicemail.c.multi-lang-patch uploaded
+ by greenfieldtech (license 369) Tested by: greenfieldtech
+
+ * apps/app_chanspy.c: Kevin has informed me that thi sort of thing
+ is not necessary.
+
+ * apps/app_chanspy.c: Move static buffers to outside for loops in
+ app_chanspy. Similar to seanbright's commit 191422, this moves
+ some static buffers to be defined outside of for loops since it
+ is undefined if memory will be re-used or if the stack will grow
+ with each iteration of the loop.
+
+2009-05-01 20:00 +0000 [r191559] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: SIP Response 410 maps to cause code 22 (or
+ 23), not 1. (closes issue #14993) Reported by: BigJimmy Patches:
+ causepatch uploaded by BigJimmy (license 371)
+
+2009-05-01 17:40 +0000 [r191488] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c: Fix DTMF not being sent to other side after a
+ partial feature match This fixes a regression from commit 176701.
+ The issue was that ast_generic_bridge never exited after the
+ feature digit timeout had elapsed, which prevented the queued
+ DTMF from being sent to the other side. This issue was reported
+ to me directly.
+
+2009-05-01 15:42 +0000 [r191422] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_queue.c: Move the defintion of the a couple arrays out
+ of loops. According to Kevin, it is unspecified as to whether a
+ variable defined inside a block is allocated once by the compiler
+ or for each pass through the block (loops being the only
+ interesting case), so just define these before we get into our
+ loop to be sure.
+
+2009-04-29 23:10 +0000 [r191220] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/h323/ast_h323.cxx, channels/chan_h323.c: Allow H.323 to
+ compile with FDLEAK checking enabled.
+
+2009-04-29 18:07 +0000 [r191096] David Brooks <dbrooks@digium.com>
+
+ * pbx/pbx_config.c: Patch to fix tab-completion crash on "remove
+ extension" This patch simply removes some old code back before
+ Asterisk used editline. This fixes the crash that occurred when
+ tab-completing "remove extension". (closes issue #14689) Reported
+ by: isaacgal
+
+2009-04-29 15:23 +0000 [r191041] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_queue.c: Fix a crash in app_queue with very long member
+ lists. A user reported via #asterisk that with very long lists of
+ members, a crash occurs in ast_strdupa, so just use a single
+ buffer and ast_copy_string instead of stack allocating copys of
+ each interface name.
+
+2009-04-27 19:29 +0000 [r190721] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in: Fix 'inconsistent
+ line endings' when autoconf 2.63 is used Attempt to make
+ configure script regeneration 'safe' using autoconf 2.63, which
+ embeds a bare CR into the script, thus making Subversion complain
+ about inconsistent line endings This commit changes the MIME type
+ of the configure script to be 'binary' thus making Subversion no
+ longer inspect line endings, and as a bonus 'svn diff' will no
+ longer try to generate diff output for it, which is not generally
+ useful anyway.
+
+2009-04-27 19:03 +0000 [r190661-190662] Russell Bryant <russell@digium.com>
+
+ * res/res_smdi.c: Fix a typo from 190661.
+
+ * res/res_smdi.c: Resolve a crash in res_smdi when used with
+ chan_dahdi. When chan_dahdi goes to get an SMDI message, it
+ provides no search criteria. It just grabs the next message that
+ arrives. This code was written with the SMDI dialplan functions
+ in mind, since that is now the preferred method of using SMDI.
+ However, this broke support of it being used from chan_dahdi.
+ (closes AST-212)
+
+2009-04-23 21:07 +0000 [r190356] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Remove a bogus ast_channel_unlock().
+
+2009-04-23 19:13 +0000 [r190286] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_local.c: Fix a bug in chan_local glare hangup
+ detection. If both sides of a Local channel were hung up at
+ around the same time it was possible for one thread to destroy
+ the local private structure and have the other thread immediately
+ try to remove the already freed structure from the local channel
+ list.
+
+2009-04-23 10:07 +0000 [r190187] Olle Johansson <oej@edvina.net>
+
+ * include/asterisk/lock.h: unistd.h is required for usleep() on
+ Darwin. It will not hurt to include it always on other platforms
+ either.
+
+2009-04-22 21:35 +0000 [r190092] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/lock.h: Detect availability of
+ pthread_rwlock_timedwrlock() before using it. (closes issue
+ #14930) Reported by: tilghman Patches:
+ 20090420__bug14930.diff.txt uploaded by tilghman (license 14)
+ Tested by: mvanbaak, tilghman
+
+2009-04-22 19:20 +0000 [r189991] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/h323/ast_h323.cxx, channels/chan_h323.c,
+ channels/h323/chan_h323.h: Make chan_h323 respect packetization
+ settings Previously, packetization settings were ignored and now
+ they are not. A new config option 'autoframing' has been added to
+ mirror the way chan_sip handles it. Turning on the autoframing
+ option (available both as a global option or per peer) overrides
+ the local settings with the remote packetization settings.
+ Testing was performed with varying packetization levels with the
+ following codecs: ulaw, alaw, gsm, and g729. (closes issue
+ #12415) Reported by: pj Patches:
+ 2009012200_h323packetization.diff.txt uploaded by mvanbaak
+ (license 7), modified by me
+
+2009-04-22 14:29 +0000 [r189849] Michiel van Baak <michiel@vanbaak.info>
+
+ * contrib/scripts/get_ilbc_source.sh: replace sed with tr to remove
+ \r from downloaded file On some systems, sed does not recognize
+ \r in the pattern the way it was used here. Use tr instead
+ because this works the same across systems. (closes issue #14936)
+ Reported by: leobrown Patches: 2009042201_14936.diff.txt uploaded
+ by mvanbaak (license 7) Tested by: leobrown, mvanbaak
+
+2009-04-21 15:52 +0000 [r189601-189664] Doug Bailey <dbailey@digium.com>
+
+ * utils/muted.c: Remove daemon call on systems that do not support
+ forking.
+
+ * main/config.c, configure, include/asterisk/autoconfig.h.in,
+ include/asterisk/compat.h, configure.ac: Add check in configure
+ script to check for GLOB_NOMAGIC and GLOB_BRACE in glob.h This
+ allows config.c to compile when linked against uclibc that does
+ not support these parameters
+
+2009-04-20 22:02 +0000 [r189537] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_odbc.c, funcs/func_strings.c: Add a workaround for
+ func_odbc/ARRAY() for problems that occur with certain special
+ characters. In certain cases, due to the way Set() works in 1.4,
+ values may not get set properly. This is a workaround for 1.4
+ only that corrects for these issues, without making func_odbc
+ more difficult to use properly. (closes issue #14614) Reported
+ by: wdoekes Patches: 20090309__bug14614__2.diff.txt uploaded by
+ tilghman (license 14)
+ double_set_unescape_workaround_for_func_odbc.osso-and-tilghman-1.diff
+ uploaded by wdoekes (license 717) Tested by: wdoekes, tilghman
+
+2009-04-20 21:10 +0000 [r189463-189465] Terry Wilson <twilson@digium.com>
+
+ * apps/app_dial.c: Update CDR appropriately when
+ AST_CAUSE_NO_ANSWER is set
+
+ * apps/app_dial.c: Don't treat a NOANSWER like a CHANUNAVAIL
+
+2009-04-20 20:58 +0000 [r189462] Sean Bright <sean.bright@gmail.com>
+
+ * pbx/ael/ael.tab.c, pbx/ael/ael.y: Properly handle @s within hints
+ in AEL. AEL was not handling the case of a device hint containing
+ an @ symbol, which caused parking hints (e.g.
+ hint(park:exten@context)) to error out the parser. This patch
+ makes AEL treat the @ the same way it treats colon and ampersand
+ now, meaning the characters are included in verbatim. (closes
+ issue #14941) Reported by: bpgoldsb Patches: bug14941.patch
+ uploaded by seanbright (license 71) Tested by: bpgoldsb
+
+2009-04-20 19:10 +0000 [r189391] Doug Bailey <dbailey@digium.com>
+
+ * main/manager.c, main/db1-ast/recno/rec_open.c,
+ channels/chan_iax2.c: Clean up problem with manager
+ implementation of mmap where it was not testing against
+ MAP_FAILED response. Got rid of shadowed variable used in
+ processign the mmap results. Change test of mmap results to
+ compare against MAP_FAILED
+
+2009-04-20 14:04 +0000 [r189277] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c: Move the check for chan->fdno == -1 to after the
+ zombie/hangup check. Many users were finding that their hung up
+ channels were staying up and causing 100% CPU usage. (issue
+ #14723) Reported by: seadweller Patches: 14723_1-4-tip.patch
+ uploaded by mmichelson (license 60) Tested by: falves11, bamby
+
+2009-04-18 01:27 +0000 [r189203] David Vossel <dvossel@digium.com>
+
+ * channels/chan_agent.c: Fixed autologoff in agents.conf not
+ working when agent logs in via AgentLogin app An agent logs in by
+ calling an extension that calls the AgentLogin app. In
+ agents.conf ackcall=always is set, so when they get a call they
+ have the choice to either acknowledge it or ignore it.
+ autologoff=10 is set as well, so if the agent ignores the call
+ over 10sec one may assume that the agent should be logged out
+ (and in this case hungup on as well), but this was not happening.
+ (closes issue #14091) Reported by: evandro Patches:
+ autologoff.diff uploaded by dvossel (license 671) Review:
+ http://reviewboard.digium.com/r/225/
+
+2009-04-17 21:27 +0000 [r189134] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c: Modifed/added some debug messages.
+ JIRA ABE-1835
+
+2009-04-17 15:43 +0000 [r189009] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c: Make Busy() application set the CDR disposition to
+ BUSY. (closes issue #14306) Reported by: cristiandimache
+
+2009-04-17 14:41 +0000 [r188937-188946] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix a bug where a value used to create the
+ channel name was bogus. This commit fixes the scenario where an
+ incoming call is authenticated using a peer entry. Previously the
+ channel name was created using either the username setting from
+ the sip.conf entry or the IP address that the call came from. Now
+ the channel name will be created using the peer name itself. This
+ commit will not change the way the channel name is generated for
+ users or friends. (closes issue #14256) Reported by: Nick_Lewis
+ Patches: chan_sip.c-chname.patch uploaded by Nick (license 657)
+ Tested by: Nick_Lewis, file
+
+ * channels/chan_dahdi.c: Fix a situation where the DAHDI channel
+ private structure lock was not unlocked when it should have been.
+ (issue AST-210)
+
+2009-04-16 21:41 +0000 [r188835] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Only update realtime, if global option
+ rtupdate != false (closes issue #14885) Reported by: deepesh
+ Patches: 20090413__bug14885.diff.txt uploaded by tilghman
+ (license 14) Tested by: deepesh
+
+2009-04-16 21:37 +0000 [r188833] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: Only disable mISDN DSP if Asterisk DSP is
+ enabled. Leave jitter setting alone. JIRA ABE-1835
+
+2009-04-16 21:02 +0000 [r188773] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Umask should not be exported into global
+ namespace. (closes issue #14912) Reported by: jcapp
+
+2009-04-15 22:08 +0000 [r188646] David Vossel <dvossel@digium.com>
+
+ * channels/chan_dahdi.c: National prefix inserted even when caller
+ ID not available When the caller ID is restricted, the expected
+ behavior is for the caller id to be blank. In chan_dahdi, the
+ national prefix is placed onto the callers number even if its
+ restricted (empty) causing the caller id to be the national
+ prefix rather than blank. (closes issue #13207) Reported by:
+ shawkris Patches: national_prefix.diff uploaded by dvossel
+ (license 671) Review: http://reviewboard.digium.com/r/220/
+
+2009-04-15 20:04 +0000 [r188582] Mark Michelson <mmichelson@digium.com>
+
+ * main/file.c: Update ast_readvideo_callback to match
+ ast_readaudio_callback. This fixes potential refcount errors that
+ may occur on ast_filestreams. AST-208
+
+2009-04-14 15:02 +0000 [r188287] David Vossel <dvossel@digium.com>
+
+ * main/audiohook.c: audio_audiohook_write_list() does not correctly
+ update sample size after ast_translate.
+ audio_audiohook_write_list() does not take into account that the
+ sample size may change after translation depending on if the
+ original frame is is 8khz or 16khz. While no 16kz codecs are
+ supported in 1.4 at the moment, this will save headaches in the
+ future if they ever are. the sample size is now updated after
+ translating to reflect this possibility. Thanks to jcolp and
+ mmichelson for helping me work this out. (issue AST-197)
+
+2009-04-13 23:04 +0000 [r188149] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_odbc.c: If fileconfig limit exceeds our maximum, then set
+ the limit to the maximum. (Closes issue #14888) Reported by:
+ falves11
+
+2009-04-10 22:16 +0000 [r187962] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/Makefile: Fix module embedding for chan_h323. Include
+ libchanh323.a in the modules.link file so that all the symbols
+ can be resolved at link time. (closes issue #11966) Reported by:
+ dome
+
+2009-04-10 19:26 +0000 [r187865] Russell Bryant <russell@digium.com>
+
+ * channels/chan_dahdi.c: Support "signaling" in addition to
+ "signalling". The sample configuration file has references to
+ both spellings.
+
+2009-04-10 17:28 +0000 [r187763] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/realtime_pgsql.sql,
+ contrib/scripts/sip-friends.sql: Add lastms column to the
+ contributed table designs
+
+2009-04-09 18:51 +0000 [r187484] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Handle a SIP race condition (reinvite before
+ an ACK) properly. RFC 5047 explains the proper course of action
+ to take if a reINVITE is received before the ACK from a previous
+ invite transaction. What we are to do is to treat the reINVITE as
+ if it were both an ACK and a reINVITE and process it normally.
+ Later, when we receive the ACK we had been expecting, we will
+ ignore it since its CSeq is less than the current iseqno of the
+ sip_pvt representing this dialog. (closes issue #13849) Reported
+ by: klaus3000 Patches: 13849_v2.patch uploaded by mmichelson
+ (license 60) Tested by: mmichelson, klaus3000
+
+2009-04-09 18:39 +0000 [r187209-187482] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/lock.h: Oops, typo
+
+ * main/manager.c, include/asterisk/lock.h: Race condition between
+ ast_cli_command() and 'module unload' could cause a deadlock. Add
+ lock timeouts to avoid this potential deadlock. (closes issue
+ #14705) Reported by: jamessan Patches:
+ 20090320__bug14705.diff.txt uploaded by tilghman (license 14)
+ Tested by: jamessan
+
+ * channels/chan_sip.c, apps/app_sendtext.c: Permit zero-length text
+ messages in SIP. (Related to an issue posted to the -users list,
+ subject "AEL2, BASE64_DECODE and hexadecimal")
+
+ * main/astfd.c (added): Oops, missed this file in the last commit.
+
+ * main/asterisk.c, agi/Makefile, build_tools/cflags.xml,
+ utils/Makefile, include/asterisk.h, main/Makefile, main/file.c:
+ Add debugging mode for diagnosing file descriptor leaks. (Related
+ to issue #14625)
+
+ * main/manager.c: Backport resolution for file descriptor leak in
+ 1.6.0 to 1.4. This fixes short reads in http manager sessions,
+ such as those done by the ast-gui branch. (Fixes AST-198)
+
+2009-04-08 19:16 +0000 [r186832-187135] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c: Fix a crash due to too few arguments to
+ RetryDial. (closes issue #14852) Reported by: junky Patches:
+ retry_fix.diff uploaded by junky (license 177)
+
+ * res/res_musiconhold.c: Fix a small logical error when loading moh
+ classes. We were unconditionally incrementing the number of
+ mohclasses registered. However, we should actually only increment
+ if the call to moh_register was successful. While this probably
+ has never caused problems, I noticed it and decided to fix it
+ anyway.
+
+ * main/channel.c: Make a couple of changes with regards to a new
+ message printed in ast_read(). "ast_read() called with no
+ recorded file descriptor" is a new message added after a bug was
+ discovered. Unfortunately, it seems there are a bunch of places
+ that potentially make such calls to ast_read() and trigger this
+ error message to be displayed. This commit does two things to
+ help to make this message appear less. First, the message has
+ been downgraded to a debug level message if dev mode is not
+ enabled. The message means a lot more to developers than it does
+ to end users, and so developers should take an effort to be sure
+ to call ast_read only when a channel is ready to be read from.
+ However, since this doesn't actually cause an error in operation
+ and is not something a user can easily fix, we should not spam
+ their console with these messages. Second, the message has been
+ moved to after the check for any pending masquerades. ast_read()
+ being called with no recorded file descriptor should not
+ interfere with a masquerade taking place. This could be seen as a
+ simple way of resolving issue #14723. However, I still want to
+ try to clear out the existing ways of triggering this message,
+ since I feel that would be a better resolution for the issue.
+
+ * formats/format_wav.c, formats/format_wav_gsm.c: Fix a few typos
+ of the word "frequency." (closes issue #14842) Reported by:
+ jvandal Patches: frequency-typo.diff uploaded by jvandal (license
+ 413)
+
+ * main/channel.c: Set the AST_FEATURE_WARNING_ACTIVE flag when a
+ p2p bridge returns AST_BRIDGE_RETRY. Without this flag set,
+ warning sounds will not be properly played to either party of the
+ bridge. (closes issue #14845) Reported by: adomjan
+
+2009-04-07 22:16 +0000 [r186775] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_macro.c: Fix Macro documentation to match current (and
+ intended) behavior. (See -dev mailing list)
+
+2009-04-07 20:43 +0000 [r186719] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c: Ensure that \r\n is printed after the ActionID in
+ an OriginateResponse. (closes issue #14847) Reported by: kobaz
+
+2009-04-06 13:54 +0000 [r186565] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Revert commit 186445 because it causes the
+ build to fail when IMAP_STORAGE is used.
+
+2009-04-03 20:19 +0000 [r186458] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: Fix a bug where DAHDI/Zaptel channels
+ would not properly switch formats when requested Don't offer
+ AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could
+ provide a slight performance benefit, the translation core in
+ Asterisk has some flaws when a channel driver offers multiple raw
+ formats. this fix is much simpler than fixing the translation
+ core to solve that issue (although that will be done later).
+
+2009-04-03 19:56 +0000 [r186415-186445] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Found a conflict in the last commit, due to
+ multiple targets
+
+ * apps/app_voicemail.c, configs/voicemail.conf.sample: Distinguish
+ in a sent email between simple sends and forwards. (closes issue
+ #11678) Reported by: jamessan Patches:
+ 20090330__bug11678.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman, lmadsen
+
+2009-04-03 15:48 +0000 [r186320] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/crypto.h: Fix a problem with the crypto variable
+ definitions not actually being defined properly. (closes issue
+ #14804) Reported by: jvandal
+
+2009-04-03 01:57 +0000 [r186229] Russell Bryant <russell@digium.com>
+
+ * cdr/cdr_radius.c: Fix a memory leak in cdr_radius. I came across
+ this while doing some testing of my ast_channel_ao2 branch. After
+ running a test overnight that generated over 5 million calls,
+ Asterisk had taken up about 1 GB of my system memory. So, I
+ re-ran the test with MALLOC_DEBUG turned on. However, it showed
+ no leaks in Asterisk during the test, even though Asterisk was
+ still consuming it somehow. Instead, I turned to valgrind, which
+ when run with --leak-check=full, told me exactly where the leak
+ came from, which was from allocations inside the radiusclient-ng
+ library. This explains why MALLOC_DEBUG did not report it. After
+ a bit of analysis, I found that we were leaking a little bit of
+ memory every time a CDR record was passed to cdr_radius. I don't
+ actually have a radius server set up to receive CDR records.
+ However, I always have my development systems compile and install
+ all modules. In addition to making sure there are not build
+ errors across modules, always loading modules helps find bugs
+ like this, too, so it is strongly recommend for all developers.
+
+2009-04-02 21:55 +0000 [r186174] Mark Michelson <mmichelson@digium.com>
+
+ * configs/features.conf.sample: Fix instructions in one-step
+ parking comment to make more sense. Changed a capital K to a
+ lowercase k.
+
+2009-04-02 17:21 +0000 [r186081] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: ensure that the buffer passed to
+ DAHDI_SET_BUFINFO is fully initialized
+
+2009-04-02 17:09 +0000 [r186057-186059] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+ 186056 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009)
+ | 2 lines Fix for AST-2009-003 ........
+
+ * channels/chan_sip.c: Avoid multiple warning messages in SIP, due
+ to this column not existing
+
+2009-04-02 13:43 +0000 [r185952] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: the DAHDI_GETCONF, DAHDI_SETCONF and
+ DAHDI_GET_PARAMS ioctls were recently corrected to show that they
+ do, in fact, read data from userspace as part of their work. due
+ to this fix, valgrind now reports a number of cases where
+ chan_dahdi passed an uninitialized (or partially) buffer to these
+ ioctls, which could lead to unexpected behavior. this patch
+ corrects chan_dahdi to ensure that buffers passed to these ioctls
+ are always fully initialized.
+
+2009-04-01 19:02 +0000 [r185845] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Fixes issue with dropped calles due to
+ re-Invite glare and re-Invites never executing after a 491
+ Acknowledgement for 491 responses were never being processed
+ because it didn't match our pending invite's seqno. Since the ACK
+ was never processed, the 491 frame would continue to be
+ retransmitted until eventually the call was dropped due to max
+ retries. Now during a pending invite, if we receive another
+ invite, we send an 491 and hold on to that glare invite's seqno
+ in the "glareinvite" variable for that sip_pvt struct. When ACK's
+ are received, we first check to see if it is in response to our
+ pending invite, if not we check to see if it is in response to a
+ glare invite. In this case, it is in response to the glare invite
+ and must be dealt with or the call is dropped. I've changed the
+ wait time for resending the re-Invite after receving a 491
+ response to comply with RFC 3261. Before this patch the scheduled
+ re-Invite would only change a flag indicating that the re-Invite
+ should be sent out, now it actually sends it out as well. (closes
+ issue #12013) Reported by: alx Review:
+ http://reviewboard.digium.com/r/213/
+
+2009-04-01 13:47 +0000 [r185771] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Fix a case where DTMF could bypass audiohooks.
+ This change fixes a situation where an audiohook that wants DTMF
+ would not actually get it. This is in the code path where we end
+ DTMF digit length emulation while handling a NULL frame.
+
+2009-03-31 22:00 +0000 [r185468-185599] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix crash that would occur if an empty member
+ was specified in queues.conf. (closes issue #14796) Reported by:
+ pida
+
+ * channels/chan_sip.c: Use AST_SCHED_DEL_SPINLOCK instead of
+ manually using the logic.
+
+ * apps/app_voicemail.c: Fix Russian voicemail intro to say the word
+ "messages" properly. (closes issue #14736) Reported by: chappell
+ Patches: voicemail_no_messages.diff uploaded by chappell (license
+ 8)
+
+2009-03-31 16:37 +0000 [r185362] David Brooks <dbrooks@digium.com>
+
+ * channels/chan_gtalk.c: Fix incorrect parsing in chan_gtalk when
+ xmpp contains extra whitespaces To drill into the xmpp to find
+ the capabilities between channels, chan_gtalk calls iks_child()
+ and iks_next(). iks_child() and iks_next() are functions in the
+ iksemel xml parsing library that traverse xml nodes. The bug here
+ is that both iks_child() and iks_next() will return the next
+ iks_struct node *regardless* of type. chan_gtalk expects the next
+ node to be of type IKS_TAG, which in most cases, it is, but in
+ this case (a call being made from the Empathy IM client), there
+ exists iks_struct nodes which are not IKS_TAG data (they are
+ extraneous whitespaces), and chan_gtalk doesn't handle that case,
+ so capabilities don't match, and a call cannot be made.
+ iks_first_tag() and iks_next_tag(), on the other hand, will not
+ return the very next iks_struct, but will check to see if the
+ next iks_struct is of type IKS_TAG. If it isn't, it will be
+ skipped, and the next struct of type IKS_TAG it finds will be
+ returned. This assures that chan_gtalk will find the iks_struct
+ it is looking for. This fix simply changes all calls to
+ iks_child() and iks_next() to become calls to iks_first_tag() and
+ iks_next_tag(), which resolves the capability matching. The
+ following is a payload listing from Empathy, which, due to the
+ extraneous whitespace, will not be parsed correctly by iksemel:
+ <iq from='dbrooksjab@235-22-24-10/Telepathy'
+ to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'>
+ <session xmlns='http://www.google.com/session'
+ initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate'
+ id='1837267342'> <description
+ xmlns='http://www.google.com/session/phone'> <payload-type
+ clockrate='16000' name='speex' id='96'/> <payload-type
+ clockrate='8000' name='PCMA' id='8'/> <payload-type
+ clockrate='8000' name='PCMU' id='0'/> <payload-type
+ clockrate='90000' name='MPA' id='97'/> <payload-type
+ clockrate='16000' name='SIREN' id='98'/> <payload-type
+ clockrate='8000' name='telephone-event' id='99'/> </description>
+ </session> </iq>
+
+2009-03-31 15:34 +0000 [r185298] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix some state_interface stuff that was in
+ trunk but not in the backport to 1.4. Issue #14359 was fixed
+ between the time that I posted the review of the backport of the
+ state interface change for 1.4. This merges the changes from that
+ issue back into 1.4. (closes issue #14359) Reported by:
+ francesco_r
+
+2009-03-31 14:06 +0000 [r185196] Joshua Colp <jcolp@digium.com>
+
+ * main/audiohook.c: Fix crash when moving audiohooks between
+ channels. Handle the scenario where we are called to move
+ audiohooks between channels and the source channel does not
+ actually have any on it. (closes issue #14734) Reported by:
+ corruptor
+
+2009-03-30 20:40 +0000 [r185120-185121] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn_config.c, configs/misdn.conf.sample: Update the
+ channel allocation method documentation.
+
+ * channels/misdn/isdn_lib.c: Make chan_misdn BRI TE side normally
+ defer channel selection to the NT side. Channel allocation
+ collisions are not handled by chan_misdn very well. This patch
+ simply avoids the problem for BRI only. For PRI, allocation
+ collisions are still possible but less likely since there are
+ simply more channels available and each end could use a different
+ allocation strategy. misdn.conf options available:
+ te_choose_channel - Use to force the TE side to allocate
+ channels. method - Specify the channel allocation strategy.
+ (closes issue #13488) Reported by: Christian_Pinedo Patches:
+ isdn_lib.patch.txt uploaded by crich Tested by: crich, siepkes,
+ festr
+
+2009-03-30 16:17 +0000 [r184980-185031] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix queue weight behavior so that calls in
+ low-weight queues are not inappropriately blocked. (This is
+ copied and pasted from the review request I made for this patch)
+ Asterisk has some odd behavior when queue weights are used. The
+ current logic used when potentially calling a queue member is: If
+ the member we are going to call is part of another queue and
+ _that other queue has any callers in it_ and has a higher weight
+ than the queue we are calling from, then don't try to contact
+ that member. The issue here is what I have marked with
+ underscores. If the higher-weighted queue has any callers in it
+ at all, then the queue member will be unreachable from the
+ lower-weighted queue. This has the potential to be really really
+ bad if using a queue strategy, such as leastrecent or
+ fewestcalls, with the potential to call the same member
+ repeatedly. The fix proposed by garychen on issue 13220 is very
+ simple and, as far as I can see, works well for this situation.
+ With this set of changes, the logic used becomes: If the member
+ we are going to call is part of another queue, the other queue
+ has a higher weight than the queue we are calling from, and the
+ higher weight queue has at least as many callers as available
+ members, then do not try to contact the queue member. If the
+ higher weighted queue has fewer callers than available members,
+ then there is no reason to deny the call to this member since the
+ other queue can afford to spare a member. Since the fix involved
+ writing a generic function for determining the number of
+ available members in the queue, I also modified the is_our_turn
+ function to make use of the new num_available_members function to
+ determine if it is our turn to try calling a member. There is one
+ small behavior change. Before writing this patch, if you had
+ autofill disabled, then if you were the head caller in a queue,
+ you would automatically be told that it was your turn to try
+ calling a member. This did not take into account whether there
+ were actually any queue members available to take the call. Now
+ we actually make sure there is at least one member available to
+ take the call if autofill is disabled. (closes issue #13220)
+ Reported by: garychen Review:
+ http://reviewboard.digium.com/r/202/
+
+ * configs/queues.conf.sample, apps/app_queue.c: Backport state
+ interface changes to app_queue from trunk. After several issues
+ raised on the Asterisk bugtracker against the 1.4 branch were
+ determined to be fixable with the state interface change
+ available in the 1.6.X series, it finally came time to just suck
+ it up and backport the change. For a detailed explanation of what
+ this change entails, the original trunk commit for this feature
+ may be found here:
+ http://svn.digium.com/view/asterisk?view=revision&revision=97203
+ In addition, the details for the use of this change to fix the
+ problems stated in issue #12970 may be found in the review
+ request I made for this change. It is linked below. (closes issue
+ #12970) Reported by: edugs15 Review:
+ http://reviewboard.digium.com/r/116
+
+2009-03-30 14:35 +0000 [r184947] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Improve our handling of T38 in the initial
+ INVITE from a device. We now answer with matching media streams
+ to what is requested. If an INVITE is received with both a T38
+ and RTP media stream this means we answer with both. For any
+ outgoing calls created as a result of this inbound one no T38 is
+ requested in the initial INVITE. Instead if we start receiving
+ udptl packets we trigger a reinvite on the outbound side. (closes
+ issue #12437) Reported by: marsosa Tested by: pinga-fogo, okrief,
+ file, afu Review: http://reviewboard.digium.com/r/208/
+
+2009-03-29 05:51 +0000 [r184842] Russell Bryant <russell@digium.com>
+
+ * apps/app_followme.c: Ensure targs variable is fully initialized.
+ (closes issue #14758) Reported by: tim_ringenbach
+
+2009-03-27 13:06 +0000 [r184565] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix an issue where nat=yes would not always
+ take effect for the RTP session on outgoing calls. If calls were
+ placed using an IP address or hostname the global nat setting was
+ copied over but was not set on the RTP session itself. This
+ caused the RTP stack to not perform symmetric RTP actions.
+ (closes issue #14546) Reported by: acunningham
+
+2009-03-26 22:17 +0000 [r184447] Kevin P. Fleming <kpfleming@digium.com>
+
+ * sounds/Makefile: use new, improved 8kHz prompts
+
+2009-03-26 21:07 +0000 [r184388] David Vossel <dvossel@digium.com>
+
+ * apps/app_test.c: pri loop TestClient/TestServer fails: server
+ SEND DTMF 8 app_test was failing when sending the last DTMF
+ digit, 8, because of the 100ms pause issued after DTMF is sent.
+ During this pause the other side would hang up causing the test
+ to look like it failed. Now the other side waits a second before
+ hanging up. (closes issue #12442) Reported by: tzafrir
+
+2009-03-25 14:12 +0000 [r184188] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/asterisk.c: Avoid destroying the CLI line when moving the
+ cursor backward and trying to autocomplete. When moving the
+ cursor backward and pressing TAB to autocomplete, a NULL is put
+ in the line and we are loosing what we have already wrote after
+ the actual cursor position. (closes issue #14373) Reported by:
+ eliel Patches: asterisk.c.patch uploaded by eliel (license 64)
+ Tested by: lmadsen
+
+2009-03-24 22:34 +0000 [r184078] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_senddtmf.c: Change NULL pointer check to be
+ ast_strlen_zero. The 'digit' variable is guaranteed to be
+ non-NULL, so the if statement could never evaluate true. Changing
+ to ast_strlen_zero makes the logic correct. This was found while
+ reviewing ast_channel_ao2 code review.
+
+2009-03-24 15:25 +0000 [r183913] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/voicemail.conf.sample: Additionally note that the
+ operator option needs an 'o' extension. (Related to issue #14731)
+
+2009-03-23 17:59 +0000 [r183700] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_monitor.c: Fix a memory leak in res_monitor.c The only
+ way that this leak would occur is if Monitor were started using
+ the Manager interface and no File: header were given. Discovered
+ while reviewing the ast_channel_ao2 review request.
+
+2009-03-20 16:53 +0000 [r183559] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Fix a crash in IAX2 registration handling
+ found during load testing with dvossel.
+
+2009-03-19 23:37 +0000 [r183481] Terry Wilson <twilson@digium.com>
+
+ * apps/app_dial.c: Add missing datastore inherit (exists in all
+ other branches)
+
+2009-03-19 19:40 +0000 [r183386] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/features.h, apps/app_dial.c, res/res_features.c:
+ Cleaning up a few things in detect disconnect patch Initialized
+ ast_call_feature in detect_disconnect to avoid accessing
+ uninitialized memory. Cleaned up /param tags in features.h. No
+ longer send dynamic features in ast_feature_detect. issue #11583
+
+2009-03-19 19:21 +0000 [r183319-183342] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c: Reordering, to change prior to unlocking
+
+ * channels/chan_dahdi.c: Delay signalling progress until a PRI
+ channel really signals progress. (closes issue #13034) Reported
+ by: klaus3000 Patches: 20090316__bug13034.diff.txt uploaded by
+ tilghman (license 14) patch_trunk_183progress_klaus3000.txt
+ uploaded by klaus3000 (license 65) Tested by: klaus3000
+
+2009-03-19 18:28 +0000 [r183291] Jason Parker <jparker@digium.com>
+
+ * main/asterisk.exports: Export some more required symbols.
+
+2009-03-19 17:52 +0000 [r183145-183241] Russell Bryant <russell@digium.com>
+
+ * main/loader.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Remove the use of RTLD_NOLOAD, as it is not
+ behaving like expected.
+
+ * main/asterisk.exports: Allow the AES API to work.
+
+ * main/asterisk.exports: Add missing semicolon in exports script.
+
+2009-03-19 16:15 +0000 [r183126] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/features.h, apps/app_dial.c, res/res_features.c,
+ res/res_features.exports: Allow disconnect feature before a call
+ is bridged feature.conf has a disconnect option. By default this
+ option is set to '*', but it could be anything. If a user wishes
+ to disconnect a call before the other side answers, only '*' will
+ work, regardless if the disconnect option is set to something
+ else. This is because features are unavailable until bridging
+ takes place. The default disconnect option, '*', was hardcoded in
+ app_dial, which doesn't make any sense from a user perspective
+ since they may expect it to be something different. This patch
+ allows features to be detected from outside of the bridge, but
+ not operated on. In this case, the disconnect feature can be
+ detected before briding and handled outside of features.c.
+ (closes issue #11583) Reported by: sobomax Patches:
+ patch-apps__app_dial.c uploaded by sobomax (license 359)
+ 11583.latest-patch uploaded by murf (license 17)
+ detect_disconnect.diff uploaded by dvossel (license 671) Tested
+ by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/
+
+2009-03-19 16:13 +0000 [r183123] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.exports: Allow the CallerID API to work again.
+
+2009-03-19 16:04 +0000 [r183115] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix an issue where cancelled outgoing SIP
+ calls would erroneously report the device as "in use." A user was
+ having an issue where if an outgoing SIP call was canceled, the
+ SIP device would remain in use if we had not received any
+ response to the initial INVITE we sent out. The SIP device would
+ remain in use until the autocongestion timer was exhausted. I
+ tracked down the cause of this to be the section of code I am
+ removing here. I asked several people what the purpose of this
+ code was meant to be, but no one could give me any sort of answer
+ as to why this was here. The person who was having this issue has
+ been using this patch for several months and it has stopped the
+ problems they have had. AST-196
+
+2009-03-18 20:02 +0000 [r182963-182965] Jeff Peeler <jpeeler@digium.com>
+
+ * configure, autoconf/ast_check_openh323.m4: fix typo which broke
+ configure
+
+ * channels/h323/compat_h323.cxx, channels/h323/ast_h323.cxx,
+ configure, autoconf/ast_check_openh323.m4,
+ channels/h323/compat_h323.h, channels/chan_h323.c,
+ channels/h323/ast_h323.h, channels/h323/chan_h323.h: Allow H.323
+ Plus library to be used in addition to the OpenH323 library
+ Chan_h323 can now be compiled against both the previously
+ supported versions of OpenH323 as well as the current H.323 Plus
+ (version 1.20.2). The configure script has been modified to look
+ in the default install location of h323 to hopefully help avoid
+ using the environment variables OPENH323DIR and PWLIBDIR. Also,
+ the CLI command "h323 show version" has been added which
+ indicates which version of h323 is in use. (closes issue 0011261)
+ Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch
+ uploaded by jthurman (license 614)
+
+2009-03-18 11:31 +0000 [r182882] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/callerid.h, channels/chan_dahdi.c,
+ main/callerid.c: fix another symbol namespace issue (reported by
+ Andrew on asterisk-dev)
+
+2009-03-18 02:09 +0000 [r182810] Russell Bryant <russell@digium.com>
+
+ * main/poll.c, main/io.c, main/channel.c, main/manager.c,
+ channels/chan_skinny.c, configure, apps/app_mp3.c, res/res_agi.c,
+ include/asterisk/poll-compat.h, channels/chan_alsa.c,
+ main/asterisk.c, apps/app_nbscat.c, main/Makefile,
+ include/asterisk/autoconfig.h.in, configure.ac, main/utils.c,
+ include/asterisk/io.h, include/asterisk/channel.h: Fix cases
+ where the internal poll() was not being used when it needed to
+ be. We have seen a number of problems caused by poll() not
+ working properly on Mac OSX. If you search around, you'll find a
+ number of references to using select() instead of poll() to work
+ around these issues. In Asterisk, we've had poll.c which
+ implements poll() using select() internally. However, we were
+ still getting reports of problems. vadim investigated a bit and
+ realized that at least on his system, even though we were
+ compiling in poll.o, the system poll() was still being used. So,
+ the primary purpose of this patch is to ensure that we're using
+ the internal poll() when we want it to be used. The changes are:
+ 1) Remove logic for when internal poll should be used from the
+ Makefile. Instead, put it in the configure script. The logic in
+ the configure script is the same as it was in the Makefile.
+ Ideally, we would have a functionality test for the problem, but
+ that's not actually possible, since we would have to be able to
+ run an application on the _target_ system to test poll()
+ behavior. 2) Always include poll.o in the build, but it will be
+ empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll()
+ throughout the source tree to ast_poll(). I feel that it is good
+ practice to give the API call a new name when we are changing its
+ behavior and not using the system version directly in all cases.
+ So, normally, ast_poll() is just redefined to poll(). On systems
+ where AST_POLL_COMPAT is defined, ast_poll() is redefined to
+ ast_internal_poll(). 4) Change poll() in main/poll.c to be
+ ast_internal_poll(). It's worth noting that any code that still
+ uses poll() directly will work fine (if they worked fine before).
+ So, for example, out of tree modules that are using poll() will
+ not stop working or anything. However, for modules to work
+ properly on Mac OSX, ast_poll() needs to be used. (closes issue
+ #13404) Reported by: agalbraith Tested by: russell, vadim
+ http://reviewboard.digium.com/r/198/
+
+2009-03-18 01:55 +0000 [r182802-182808] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/astobj2.c, main/asterisk.exports (added),
+ res/res_odbc.exports (added), res/res_speech.exports (added),
+ res/res_config_odbc.c, res/res_features.exports (added),
+ build_tools/strip_nonapi (removed), res/res_adsi.exports (added),
+ res/res_indications.c, default.exports (added), makeopts.in,
+ res/res_jabber.exports (added), res/res_monitor.exports (added),
+ res/res_config_pgsql.c, res/res_snmp.c, main/Makefile,
+ res/res_smdi.exports (added), include/asterisk/astobj2.h,
+ res/res_crypto.c, res/res_agi.exports (added), Makefile.rules,
+ res/res_musiconhold.c: Improve the build system to *properly*
+ remove unnecessary symbols from the runtime global namespace.
+ Along the way, change the prefixes on some internal-only API
+ calls to use a common prefix. With these changes, for a module to
+ export symbols into the global namespace, it must have *both* the
+ AST_MODFLAG_GLOBAL_SYMBOLS flag and a linker script that allows
+ the linker to leave the symbols exposed in the module's .so file
+ (see res_odbc.exports for an example).
+
+ * main/astobj2.c, main/asterisk.exports (removed),
+ res/res_odbc.exports (removed), main/channel.c,
+ res/res_config_odbc.c, res/res_features.exports (removed),
+ default.exports (removed), include/asterisk/frame.h,
+ res/res_jabber.exports (removed), res/res_config_pgsql.c,
+ main/Makefile, res/res_smdi.exports (removed),
+ include/asterisk/astobj2.h, main/slinfactory.c, res/res_crypto.c,
+ res/res_agi.exports (removed), res/res_speech.exports (removed),
+ include/asterisk/linkedlists.h, main/file.c,
+ build_tools/strip_nonapi (added), res/res_adsi.exports (removed),
+ res/res_indications.c, makeopts.in, apps/app_mixmonitor.c,
+ apps/app_chanspy.c, res/res_monitor.exports (removed),
+ main/autoservice.c, build_tools/cflags-devmode.xml, main/frame.c,
+ apps/app_meetme.c, res/res_snmp.c, Makefile.rules,
+ res/res_musiconhold.c: revert commit that included extranous
+ changes
+
+ * /: remove accidentally merged properties
+
+ * main/astobj2.c, main/asterisk.exports (added),
+ res/res_odbc.exports (added), main/channel.c,
+ res/res_config_odbc.c, res/res_features.exports (added),
+ default.exports (added), include/asterisk/frame.h,
+ res/res_jabber.exports (added), res/res_config_pgsql.c,
+ main/Makefile, res/res_smdi.exports (added),
+ include/asterisk/astobj2.h, main/slinfactory.c, res/res_crypto.c,
+ res/res_agi.exports (added), res/res_speech.exports (added),
+ include/asterisk/linkedlists.h, main/file.c,
+ build_tools/strip_nonapi (removed), res/res_adsi.exports (added),
+ res/res_indications.c, makeopts.in, apps/app_mixmonitor.c,
+ apps/app_chanspy.c, res/res_monitor.exports (added),
+ main/autoservice.c, build_tools/cflags-devmode.xml, main/frame.c,
+ /, apps/app_meetme.c, res/res_snmp.c, Makefile.rules,
+ res/res_musiconhold.c: Improve the build system to *properly*
+ remove unnecessary symbols from the runtime global namespace.
+ Along the way, change the prefixes on some internal-only API
+ calls to use a common prefix. With these changes, for a module to
+ export symbols into the global namespace, it must have *both* the
+ AST_MODFLAG_GLOBAL_SYMBOLS flag and a linker script that allows
+ the linker to leave the symbols exposed in the module's .so file
+ (see res_odbc.exports for an example).
+
+2009-03-17 20:13 +0000 [r182652] Jason Parker <jparker@digium.com>
+
+ * channels/chan_dahdi.c, apps/app_flash.c: Allow dahdichanname to
+ work as advertised. (closes issue #14056) Reported by: dsedivec
+ Patches: load_from_zapata_conf.patch uploaded by dsedivec
+ (license 638)
+
+2009-03-17 05:50 +0000 [r182449] Tilghman Lesher <tlesher@digium.com>
+
+ * main/db.c: Fix race in astdb The underlying db1 implementation
+ does not fully isolate the pages retrieved from astdb, so the
+ lock protecting accesses needs to be extended until the copy from
+ the shared memory structure is done. (closes issue #14682)
+ Reported by: makoto
+
+2009-03-16 Leif Madsen <lmadsen@digium.com>
+
+ * Released 1.4.24
+
+2009-03-06 Leif Madsen <lmadsen@digium.com>
+
+ * Released 1.4.24-rc1
+
+2009-03-06 18:23 +0000 [r180567] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Make compilation succeed in dev-mode when
+ IMAP storage is enabled.
+
+2009-03-06 17:19 +0000 [r180532] David Vossel <dvossel@digium.com>
+
+ * main/enum.c: Fix handling of backreferences for ENUM lookups
+ enum.c did not handle regex backtraces correctly. The '\1' in the
+ regex is a backreference that requires a pattern match to be
+ inserted. The way the code used to work is that it would find the
+ backreference and insert the entire input string minus the '+'.
+ This is incorrect. The regexec() function takes in a variable
+ called pmatch which is an array of structs containing the start
+ and end indexes for each backreference substring. The original
+ code actually passed the pmatch array pointer into regexec but
+ never did anything with it. Now when a backtrace is found, the
+ backtrace number is looked up in the pmatch array and the correct
+ substring is inserted. (closes issue #14576) Reported by:
+ chris-mac Review: http://reviewboard.digium.com/r/187/
+
+2009-03-05 23:26 +0000 [r180380-180464] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: [IMAP] Fix message retrieval issues when
+ identical mailbox names were defined in separate contexts. There
+ was a fix put in a while back so that an X-Asterisk-VM-Context
+ message header was added to stored IMAP voicemails. This would
+ allow for us to differentiate if the same mailbox name was used
+ in multiple contexts. The problem still left was that not all
+ places where messages were retrieved actually attempted to use
+ this header for information when retrieving messages. This commit
+ fixes that so that MWI and message retrieval from VoiceMailMain
+ work as expected. (closes issue #13853) Reported by: vicks1
+ Patches: 13853_v2.patch uploaded by mmichelson (license 60)
+ Tested by: lmadsen
+
+ * apps/app_voicemail.c, configs/voicemail.conf.sample: Fix broken
+ mailbox parsing when searchcontexts option is enabled. When using
+ the searchcontexts option in voicemail.conf, the code made the
+ assumption that all mailbox names defined were unique across all
+ contexts. However, the code did nothing to actually enforce this
+ assumption, nor did it do anything to alert a user that he may
+ have created an ambiguity in his voicemail.conf file by defining
+ the same mailbox name in multiple contexts. With this change, we
+ now will issue a nice long warning if searchcontexts is on and we
+ encounter the same mailbox name in multiple contexts and ignore
+ any duplicates after the first box. Whether searchcontexts is
+ enabled or not, if we come across a duplicate mailbox in the same
+ context, then we will issue a warning and ignore the duplicated
+ mailbox. I have also added a small note to voicemail.conf.sample
+ in the explanation for searchcontexts explaining that you cannot
+ define the same mailbox in multiple contexts if you have enabled
+ the option. (closes issue #14599) Reported by: lmadsen Patches:
+ 14599.patch uploaded by mmichelson (license 60) (with slight
+ modification) Tested by: lmadsen
+
+2009-03-05 18:22 +0000 [r180372] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/rtp.c, main/frame.c, include/asterisk/frame.h: Fix problems
+ when RTP packet frame size is changed During some code analysis,
+ I found that calling ast_rtp_codec_setpref() on an ast_rtp
+ session does not work as expected; it does not adjust the
+ smoother that may on the RTP session, in fact it summarily drops
+ it, even if it has data in it, even if the current format's
+ framing size has not changed. This is not good. This patch
+ changes this behavior, so that if the packetization size for the
+ current format changes, any existing smoother is safely updated
+ to use the new size, and if no smoother was present, one is
+ created. A new API call for smoothers,
+ ast_smoother_reconfigure(), was required to implement these
+ changes. Review: http://reviewboard.digium.com/r/184/
+
+2009-03-04 19:22 +0000 [r180194] Joshua Colp <jcolp@digium.com>
+
+ * main/callerid.c: Look for the number in a callerid string
+ starting from the end. This way a value using <> can exist in the
+ name portion. (issue #AST-194)
+
+2009-03-03 23:01 +0000 [r180010] Jason Parker <jparker@digium.com>
+
+ * channels/chan_dahdi.c: Make sure we still support zapchan in
+ users.conf, in addition to dahdichan.
+
+2009-03-03 22:48 +0000 [r180006] Mark Michelson <mmichelson@digium.com>
+
+ * configs/queues.conf.sample, apps/app_queue.c: Clarify some
+ documentation of queues.conf.sample It had always been possible
+ to explicitly specify a "blank" value for a sound file in
+ queues.conf and have no sound played back. The problem with this
+ is that it would result in some ugly CLI warnings from file.c.
+ This commit introduces a check when playing a file in app_queue
+ to see if the name of the file is zero-length and return early if
+ that is the case. Also, the ability to specify the blank sound
+ files in queues.conf is now mentioned more clearly in
+ queues.conf.sample (closes issue #14227) Reported by: caspy
+
+2009-03-03 18:27 +0000 [r179840] Joshua Colp <jcolp@digium.com>
+
+ * res/res_features.c: Do not assume that the bridge_cdr is still
+ attached to the channel when the 'h' exten is finished executing.
+ It is possible for a masquerade operation to occur when the 'h'
+ exten is operating. This operation moves the CDR records around
+ causing the bridge_cdr to no longer exist on the channel where it
+ is expected to. We can not safely modify it afterwards because of
+ this, so don't even try. (closes issue #14564) Reported by: meric
+
+2009-03-03 18:11 +0000 [r179807] Steve Murphy <murf@digium.com>
+
+ * main/ast_expr2.fl, main/ast_expr2.c, utils/Makefile,
+ utils/expr2.testinput, main/ast_expr2.h, main/ast_expr2.y,
+ main/ast_expr2f.c: These changes allow AEL to better check ${}
+ constructs within $[...], that are concatenated with text. I
+ modified and added rules in ast_expr2.fl to better handle the
+ concatenations. I added some default routines to ast_expr2.y so
+ the standalone would compile. It also looks like I haven't run
+ this thru bison since 2.1, so it's good to get this updated. The
+ Makefile has comments added now for check_expr2 and check_expr to
+ explain what they are for, and how to run them. The testexpr2s
+ stuff has been removed, in favor of check_expr2. expr2.testinput
+ has been updated to include the two expressions that inspired
+ these changes (from mcnobody on #asterisk this morning) The
+ regression has been run and all looks well.
+
+2009-03-03 16:45 +0000 [r179741] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Ensure chan->fdno always gets reset to -1 after
+ handling a channel fd event. Since setting fdno to -1 had to be
+ moved, a couple of other code paths that do process an fd event
+ return early and do not pass through the code path where it was
+ moved to. So, set it to -1 in a few other places, too.
+
+2009-03-03 14:38 +0000 [r179671] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c: Move where fdno is set to the default value to
+ *after* the read callback of the channel driver is called. We
+ have to do this as the underlying channel driver may need the
+ fdno value to determine what to read.
+
+2009-03-03 13:53 +0000 [r179608] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Make it easier to detect an improper call to
+ ast_read(). When you call ast_waitfor() on a channel, the index
+ into the channel fds array that holds the file descriptor that
+ poll() determines has input available is stored in fdno. This
+ patch clears out this value after a call to ast_read() and also
+ reports errors if ast_read() is called without an fdno set. From
+ a discussion on the asterisk-dev list.
+
+2009-03-02 23:54 +0000 [r179536] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c: Fix bridging regression from commit 176701 This
+ fixes a bad regression where the bridge would exit after an
+ attended transfer was made. The problem was due to nexteventts
+ getting set after the masquerade which caused the bridge to
+ return AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by:
+ tim_ringenbach
+
+2009-03-02 23:34 +0000 [r179532] Russell Bryant <russell@digium.com>
+
+ * apps/app_meetme.c: Move ast_waitfor() down to avoid the results
+ of the API call becoming stale. This call to ast_waitfor() was
+ being done way too soon in this section of code. Specifically,
+ there was code in between the call to waitfor and the code that
+ uses the result that puts the channel in autoservice. By putting
+ the channel in autoservice, the previous results of ast_waitfor()
+ become meaningless, as the autoservice thread will do it's own
+ ast_waitfor() and ast_read() on the channel. So, when we came
+ back out of autoservice and eventually hit the block of code that
+ calls ast_read() on the channel, there may not actually be any
+ input on the channel available. Even though the previous call to
+ ast_waitfor() in app_meetme said there was input, the autoservice
+ thread has since serviced the channel for some period of time.
+ This bug manifested itself while dvossel was doing some testing
+ of MeetMe in Asterisk trunk. He was using the timerfd timing
+ module. When the code hit ast_read() erroneously, it determined
+ that it must have been called because of input on the timer fd,
+ as chan->fdno was set to AST_TIMING_FD, since that was the cause
+ of the last legitimate call to ast_read() done by autoservice. In
+ this test, an IAX2 channel was calling into the MeetMe
+ conference. It was _much_ more likely to be seen with an IAX2
+ channel because of the way audio is handled. Every audio frame
+ that comes in results in a call to ast_queue_frame(), which then
+ uses ast_timer_enable_continuous() to notify the channel thread
+ that a frame is waiting to be handled. So, the chances of
+ ast_waitfor() indicating that a channel needs servicing due to a
+ timer event on an IAX2 event is very high. Finally, it is
+ interesting to note that if a different timing interface was
+ being used, this bug would probably not be noticed. When
+ ast_read() is called and erroneously thinks that there is a timer
+ event to handle, it calls the ast_timer_ack() function. The
+ pthread and dahdi timing modules handle the ack() function being
+ called when there is no event by simply ignoring it. In the case
+ of the timerfd module, it results in a read() on the timer fd
+ that will block forever, as there is no data to read. This caused
+ Asterisk to lock up very quickly. Thanks to dvossel and
+ mmichelson for the fun debugging session. :-)
+
+2009-03-02 23:09 +0000 [r179468] Tilghman Lesher <tlesher@digium.com>
+
+ * main/app.c: When ending a recording with silence detection,
+ remember to reduce the duration. The end of the recording is
+ correspondingly trimmed, but the duration was not trimmed by the
+ number of seconds trimmed, so the saved duration was necessarily
+ longer than the actual soundfile duration. (closes issue #14406)
+ Reported by: sasargen Patches: 20090226__bug14406.diff.txt
+ uploaded by tilghman (license 14) Tested by: sasargen
+
+2009-03-02 22:58 +0000 [r179461] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Ensure that only one thread is calling
+ ast_settimeout() on a channel at a time. For example, with an
+ IAX2 channel, you can have both the channel thread and the
+ chan_iax2 processing threads calling this function, and doing so
+ twice at the same time is a bad thing. (Found in a debugging
+ session with dvossel and mmichelson)
+
+2009-03-02 20:14 +0000 [r179395] Jason Parker <jparker@digium.com>
+
+ * main/editline/configure, main/editline/np/unvis.c,
+ main/editline/sys.h, main/editline/configure.in: Remove several
+ silly warnings in editline. One about a broken preprocessor
+ directive, and another about strlcpy/strlcat. (closes issue
+ #14264) Reported by: dimas
+
+2009-02-27 19:03 +0000 [r179056] Jason Parker <jparker@digium.com>
+
+ * doc/channelvariables.txt: Update documentation for DIALEDTIME and
+ ANSWEREDTIME variables. (closes issue #14566) Reported by:
+ klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by
+ klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by
+ klaus3000 (license 65)
+
+2009-02-26 21:27 +0000 [r178956] Steve Murphy <murf@digium.com>
+
+ * configs/features.conf.sample, res/res_features.c: This change
+ moves the default feature digit timeout to 1000 ms from the
+ previous default of 500. As per bug 14515, a dev discussion
+ arrived at a "mediated concensus" of a default feature digit
+ timeout of 1.0 sec. Some voted for 1300; ctooley thought 1500 for
+ distracted phone users in phone booths; kpfleming put his foot
+ down at 1.0 sec. Users who found the previous default max delay
+ of 250 msec perfect, are welcome to override the new default.
+ Notice that I said that 250 msec was the default; wait a minute,
+ you might say, the config file said it was 500 msec!; well,
+ because of the bug fix for 14515, we found that 500 msec was
+ actually enforcing a max of 250. The bug fix would restore 500
+ msec, but we felt even that was a bit tight for most users...
+ 2000 msec was pushed earlier by mmichelson, so that reduces to
+ 1000 msec after the bug fix. Enjoy!
+
+2009-02-26 17:24 +0000 [r178838] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: IAX2 prune realtime fix Now prune_users()
+ and prune_peers() are called instead of reload_config() to prune
+ all users/peers that are realtime. These functions remove all
+ users/peers with the rtfriend and delme flags set.
+ iax2_prune_realtime() also lacked the code to properly delete a
+ single friend. For example. if iax2 prune realtime <friend> was
+ called, only the peer instance would be removed. The user would
+ still remain. (closes issue #14479) Reported by: mousepad99
+ Review: http://reviewboard.digium.com/r/176/
+
+2009-02-26 17:09 +0000 [r178640-178804] Steve Murphy <murf@digium.com>
+
+ * res/res_features.c: This patch prevents the feature detection
+ timeout from being cut in half. Because the ast_channel_bridge()
+ call will return 0 and pass a frame pointer for both DTMF_BEGIN
+ and DTMF_END, the feature_timer field in hte config struct is
+ getting decremented twice, which effectively cuts the
+ digittimeout in half. I added conditions to the if statement to
+ only let DTMF_END frames to flow thru, which solved the problem.
+ Also, when the frame pointer is null, let control flow thru--
+ this usually happens on timeouts. I added a comment to the code
+ to explain what's going on and why. Many thanks to sodom for
+ reporting this problem. Personnally, it always seemed like
+ something was wrong with the featuredigittimeout, but I never
+ could quite decide what... and was too busy to investigate. This
+ bug forced the issue, and now we know. Sodom had other issues in
+ 14515, but I couldn't reproduce them. If he still has problems,
+ and wants to get them solved, he is welcome to reopen 14515.
+ (closes issue #14515) Reported by: sodom Patches: 14515.patch
+ uploaded by murf (license 17) Tested by: murf, sodom
+
+ * main/ast_expr2.fl, main/ast_expr2f.c: This patch completes the
+ fixes nec. to make 1.4 asterisk dialplan expressions ($[...])
+ 8-bit transparent While I was updating ast_expr2.fl, I missed one
+ rule that would allow 8-bit chars to be caught in tokens; and in
+ so doing, it absorbs the ${ sequence and messes up the checking
+ of raw exprs by AEL. Trunk already has these changes. (closes
+ issue #14543) Reported by: klaus3000 Patches: patch.14543
+ uploaded by murf (license 17) Tested by: murf
+
+2009-02-25 12:43 +0000 [r178508] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c: Update the copyright year for the main page of
+ the doxygen documentation.
+
+2009-02-24 23:25 +0000 [r178445] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/extensions.conf.sample: Add section about the #exec
+ command in configuration files. (closes issue #14540) Reported
+ by: jtodd Patch by: jtodd, with additional notes by tilghman
+ (license 14)
+
+2009-02-24 20:36 +0000 [r178373] Russell Bryant <russell@digium.com>
+
+ * main/rtp.c: Only set dtmfcount on BEGIN, and ensure it gets reset
+ to 0 properly. (issue #14460) Reported by: moliveras Tested by:
+ russell
+
+2009-02-24 17:02 +0000 [r178266] Terry Wilson <twilson@digium.com>
+
+ * apps/app_dahdiras.c, res/res_musiconhold.c: Change include order
+ to make compile on Centos 5 with DAHDI If BIT_TYPES_DEFINED gets
+ defined before linux/types.h is included, the __s32 type doesn't
+ get defined
+
+2009-02-24 15:16 +0000 [r178205] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Skip check for extension when subscribing
+ for MWI. Since the remote side is not actually subscribing to a
+ specific extension when subscribing for MWI just skip the check
+ to see if the extension exists. They can't use it to specify the
+ mailbox either since we require configuration of that in sip.conf
+ (closes issue #14531) Reported by: festr
+
+2009-02-23 23:09 +0000 [r178141] Russell Bryant <russell@digium.com>
+
+ * main/rtp.c: Fix infinite DTMF when a BEGIN is received without an
+ END. This commit is related to rev 175124 of 1.4 where a previous
+ attempt was made to fix this problem. The problem with the
+ previous patch was that the inserted code needed to go _before_
+ setting the lastrxts to the current timestamp. Because those were
+ the same, the dtmfcount variable was never decremented, and so
+ the END was never sent. In passing, I removed the dtmfsamples
+ variable which was completed unused. I also removed a redundant
+ setting of the lastrxts variable. (closes issue #14460) Reported
+ by: moliveras
+
+2009-02-20 22:59 +0000 [r177701-177786] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Don't print the CR-NL combination when we aren't
+ outputting to the manager. An embedded CR-NL in a CLI command
+ screws up several AMI parsers that don't expect to see that
+ combination in the middle of output. (Closes issue #14305)
+ Reported by: martins Patch by: tilghman
+
+ * include/asterisk/threadstorage.h: This exception does not appear
+ to still be true for Solaris 10, and OpenSolaris definitely needs
+ it to be removed. Fixed for snuff-home on -dev channel.
+
+2009-02-20 20:17 +0000 [r177696] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, include/asterisk/frame.h: Fixes issue with
+ undefined audio codecs in chan_iax2 During iax2 call negotiation,
+ supported codecs are passed in an Information Element containing
+ a 2 byte field where each bit correlates to a specific codec. In
+ 1.4 only audio codec bits 0-12 are defined, leaving bits 13-15
+ undefined. By default all bits are enabled unless specified
+ otherwise. Since its a 2 byte field and 13-15 are not defined,
+ these bits are never turned off. In trunk, bits 13-15 are
+ defined, which means 1.4 is advertising support for codecs it
+ does not have when talking to trunk. I fixed this by adding
+ #define for undefined audio codec bits. These bits are then
+ removed from iax2's full bandwidth capabilities. (closes issue
+ #14283) Reported by: jcovert
+
+2009-02-19 22:51 +0000 [r177540] Steve Murphy <murf@digium.com>
+
+ * main/ast_expr2.fl, main/Makefile, main/ast_expr2f.c: This patch
+ fixes a problem with 8-bit input to the ast_expr2 scanner. The
+ real culprit was the --full argument to flex in the Makefile!
+ This causes a 7-bit scanner to be generated. I reviewed the rules
+ and found one rule where I needed to specifically include 8-bit
+ chars for a token. I tested against the text supplied by ibercom,
+ and all looks very well. This has been there a surprisingly long
+ time! (closes issue #14498) Reported by: ibercom Patches:
+ 14498.patch uploaded by murf (license 17) Tested by: murf
+
+2009-02-19 22:26 +0000 [r177536] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Fix up potential crashes, by reducing the
+ sharing between interactive and non-interactive threads. (closes
+ issue #14253) Reported by: Skavin Patches:
+ 20090219__bug14253.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Skavin
+
+2009-02-19 18:58 +0000 [r177450] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Force a MWI notification after subscribe
+ request. Reported by the Resiprocate dev team. Thanks!
+
+2009-02-19 16:37 +0000 [r177383] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_speech_utils.c: If we are able to create a speech
+ structure unset the ERROR variable in case it was previously set.
+ (issue #LUMENVOX-13)
+
+2009-02-18 22:43 +0000 [r177225] Steve Murphy <murf@digium.com>
+
+ * pbx/ael/ael.tab.c, pbx/ael/ael.y: This patch fixes a regression
+ of sorts that was introduced in rev 24425. It basically fixes
+ AST-190/ABE-1782. What was wrong: the user has 6000 extensions in
+ one context; and then 6000 contexts, one per extension. The
+ parser could only handle about 4893 of the 6000 extens in the
+ single context. This was due to the regression I mentioned. To
+ get rid of shift/reduce conflicts, Luigi set up right-recursive
+ lists for globals, context elements, switch lists, and
+ statements. Right recursive lists got rid of the warnings, but
+ instead, they use up a tremendous amount of stack space when the
+ lists are long. I saw this a few years back, and resolved not to
+ fix it until someone complained. That day has arrived! After the
+ changes were made, I ran the regression test suite, and there
+ were no problems. I took the test case the user provided, and
+ added 100,000 extensions to the single context, that already had
+ 6,000 extens in it. (I'll see your 6, and raise you 100!) It
+ takes a few minutes to read it all in, check it and generate code
+ for it, but no problems. So, I think I can say that
+ fundamentally, there are no longer any limits on the number of
+ items you can place in contexts, statement blocks, switches, or
+ globals, beyond your virt mem constraints.
+
+2009-02-18 20:06 +0000 [r177160] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/h323/cisco-h225.cxx, channels/h323/compat_h323.cxx,
+ autoconf/ast_check_pwlib.m4, channels/h323/cisco-h225.h,
+ channels/h323/caps_h323.cxx, channels/h323/ast_h323.cxx,
+ channels/h323/ast_ptlib.h (added), configure,
+ channels/h323/compat_h323.h, configure.ac,
+ channels/h323/caps_h323.h, autoconf/ast_prog_sed.m4,
+ channels/h323/ast_h323.h, channels/h323/chan_h323.h: Modify h323
+ to build against PTLib as well as the older PWLib Several changes
+ in PTLib have occurred requiring build time detection. Changes
+ accounted for include the library name change, config option
+ change, install location change, and a boolean type change which
+ is handled by ast_ptlib.h. Also, the sed check has been modified
+ to properly work with autoconf >= 2.62. (closes issue #14224)
+ Reported by: bergolth Patches: asterisk-autoconf-sed.patch
+ uploaded by bergolth (license 661) asterisk-pwlib-v3.patch
+ uploaded by bergolth (license 661) Tested by: jpeeler
+
+2009-02-18 18:30 +0000 [r177096] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/config.h: Document the return value of the
+ update method (as requested on -dev list)
+
+2009-02-18 17:41 +0000 [r176945-177039] Doug Bailey <dbailey@digium.com>
+
+ * main/utils.c: Merged revisions 177035 manually from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177035 |
+ dbailey | 2009-02-18 11:24:07 -0600 (Wed, 18 Feb 2009) | 2 lines
+ Fixed error where a check for an zero length, terminated string
+ was needed. ........
+
+ * main/utils.c: Need to take into account the \0 terminator of the
+ old string to determine the amount available.
+
+2009-02-18 00:34 +0000 [r176810] Shaun Ruffell <sruffell@digium.com>
+
+ * codecs/codec_dahdi.c: Several changes to codec_dahdi to play nice
+ with G723. This commit brings in the changes that were living out
+ on the svn/asterisk/team/sruffell/asterisk-1.4-transcoder branch.
+ codec_dahdi.c now always uses signed linear as the simple codec
+ so that a soft g729 codec will not end up being preferred to the
+ hardware codec. There are also changes to allow codec_dahdi.c to
+ feed packets to the hardware in the native sample size of the
+ codec. This solves problems with choppy audio when using G723.
+
+2009-02-17 21:54 +0000 [r176701] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, res/res_features.c, include/asterisk/channel.h:
+ Modify bridging to properly evaluate DTMF after first warning is
+ played The main problem is currently if the Dial flag L is used
+ with a warning sound, DTMF is not evaluated after the first
+ warning sound. To fix this, a flag has been added in
+ ast_generic_bridge for playing the warning which ensures that if
+ a scheduled warning is missed, multiple warrnings are not played
+ back (due to a feature evaluation or waiting for digits).
+ ast_channel_bridge was modified to store the nexteventts in the
+ ast_bridge_config structure as that information was lost every
+ time ast_channel_bridge was reentered, causing a hangup due to
+ incorrect time calculations. (closes issue #14315) Reported by:
+ tim_ringenbach Reviewed on reviewboard:
+ http://reviewboard.digium.com/r/163/
+
+2009-02-17 21:21 +0000 [r176426-176661] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c: Backport change to 1.4: Prior to
+ masquerade, move the group definitions to the channel performing
+ the masq, so that the group count lingers past the bridge.
+ (closes issue #14275) Reported by: kowalma Patches:
+ 20090216__bug14275.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: kowalma
+
+ * channels/chan_sip.c: After a 'sip reload', qualifies for realtime
+ peers weren't immediately restarted, instead waiting until the
+ next registration. We're now caching the qualify across a
+ reload/restart and starting the qualify immediately upon loading
+ the peer. (closes issue #14196) Reported by: pdf Patches:
+ 20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663)
+ Tested by: pdf
+
+2009-02-16 23:30 +0000 [r176354] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: Fixes issue with AST_CONTROL_SRCUPDATE not
+ being relayed correctly during bridging This should have been
+ committed with rev176247, but I missed it. srcupdate frames no
+ longer break out of the native bridge, but are not being sent to
+ the other call leg either. This fixs that. issue #13749
+
+2009-02-16 21:41 +0000 [r176254] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/utils.c: correct a logic error in the last stringfields
+ commit... don't mark additional space as allocated if the string
+ was built using already-allocated space
+
+2009-02-16 21:39 +0000 [r176249-176252] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_meetme.c: Remove unused variable and make dev-mode
+ compilation happy
+
+ * apps/app_meetme.c: Open the DAHDI pseudo device and set it to be
+ nonblocking atomically Apparently on FreeBSD, attempting to set
+ the O_NONBLOCKING flag separately from opening the file was
+ causing an "inappropriate ioctl for device" error. While I cannot
+ fathom why this would be happening, I certainly am not opposed to
+ making the code a bit more compact/efficient if it also fixes a
+ bug. (closes issue #14482) Reported by: ys Patches: meetme.patch
+ uploaded by ys (license 281) Tested by: ys
+
+2009-02-16 21:28 +0000 [r176247] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: Fixes issue with AST_CONTROL_SRCUPDATE
+ breaking out of native bridge In iax2, when a
+ AST_CONTROL_SRCUPDATE is received it breaks from the native
+ bridge, but since there is no code path to handle srcupdate it
+ just goes to be beginning of the loop. This was causing packet
+ storms of srcupdate frames between servers. Now srcupdate frames
+ do not break the native bridge for processing. (closes issue
+ #13749) Reported by: adiemus
+
+2009-02-16 21:10 +0000 [r176216] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/utils.c: fix a flaw in the ast_string_field_build() family
+ of API calls; these functions made no attempt to reuse the space
+ already allocated to a field, so every time the field was written
+ it would allocate new space, leading to what appeared to be a
+ memory leak.
+
+2009-02-16 15:33 +0000 [r176029] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Don't have the Via header stored as a
+ stringfield as it can change often during the lifetime of a
+ dialog. This issue crept up with subscriptions on the AA50. When
+ an outgoing NOTIFY is sent a new branch value is created and the
+ Via header is changed to reflect it. Since this was a stringfield
+ a new spot in the pool was used for the value while the old was
+ left untouched/unused. If the current pool was full a new pool
+ was created. This would cause memory usage to increase steadily.
+ (issue #AA50-2332)
+
+2009-02-15 23:37 +0000 [r175921] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/pbx.c, channels/chan_sip.c, main/devicestate.c,
+ include/asterisk/manager.h: fix mis-spelling of the word
+ registered. Reported by De_Mon on #asterisk-dev.
+
+2009-02-15 20:33 +0000 [r175777-175825] Olle Johansson <oej@edvina.net>
+
+ * formats/format_ilbc.c: format_ilbc does not depend on codec
+ libraries and can therefore always be made. My mistake. Ursäkta!
+
+ * formats/format_ilbc.c: Disable format_ilbc.so by default, like
+ codec_ilbc.so
+
+ * channels/chan_sip.c: Make sure that the debug line is not printed
+ on debug level 0
+
+2009-02-13 21:53 +0000 [r175698] Jason Parker <jparker@digium.com>
+
+ * include/asterisk/dahdi_compat.h: Zaptel is not DAHDI. Rather,
+ Zaptel is actually Zaptel. (in case you're confused, DAHDI is
+ still DAHDI)
+
+2009-02-13 19:47 +0000 [r175407-175590] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Fix a potential crash situation when using
+ IMAP voicemail If calling into VoiceMailMain when using IMAP
+ storage, it was possible to crash Asterisk by hanging up the
+ phone when prompted for a voicemail mailbox. This patch fixes the
+ issue. While it may appear that this patch is superficial, it
+ allows code execution to continue to the failure case just below
+ the IMAP_STORAGE code block where this patch has been applied
+ (closes issue #14473) Reported by: dwpaul Patches:
+ voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license
+ 689)
+
+ * main/file.c: Fix a place where filestreams were not refcounted
+ properly This section was already present in trunk and other
+ branches, but did not exist in 1.4. (closes issue #14395)
+ Reported by: ZX81 Patches: 14395.patch uploaded by putnopvut
+ (license 60) Tested by: ZX81
+
+2009-02-12 21:19 +0000 [r175311] Tilghman Lesher <tlesher@digium.com>
+
+ * main/udptl.c: Fix crashes when receiving certain T.38 packets.
+ Also, increase the maximum size of T.38 packets and warn users
+ when they try to set the limits above those maximums. (closes
+ issue #13050) Reported by: schern Patches:
+ 20090212__bug13050.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: schern
+
+2009-02-12 20:34 +0000 [r175187-175294] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_features.c: Fix ParkedCall event information for From
+ field in the case of a blind transfer If the parker information
+ can not be obtained from the peer, try and see if the
+ BLINDTRANSFER channel variable has been set. Previously, a blind
+ transfer to the ParkAndAnnounce app would return nothing for the
+ From. Closes AST-189
+
+ * res/res_features.c: Fix crash in event of failed attempt to
+ transfer to parking The peer may not necessarily exist, such as
+ in the case of a transfer to ParkAndAnnounce. In this case don't
+ try to play a sound to it.
+
+2009-02-12 16:51 +0000 [r175124] Russell Bryant <russell@digium.com>
+
+ * main/rtp.c: Don't send DTMF for infinite time if we do not
+ receive an END event. I thought that this was going to end up
+ being a pretty gnarly fix, but it turns out that there was
+ actually already a configuration option in rtp.conf, dtmftimeout,
+ that was intended to handle this situation. However, in between
+ Asterisk 1.2 and Asterisk 1.4, the code that processed the option
+ got lost. So, this commit brings it back to life. The default
+ timeout is 3 seconds. However, it is worth noting that having
+ this be configurable at all is not really the recommended
+ behavior in RFC 2833. From Section 3.5 of RFC 2833: Limiting the
+ time period of extending the tone is necessary to avoid that a
+ tone "gets stuck". Regardless of the algorithm used, the tone
+ SHOULD NOT be extended by more than three packet interarrival
+ times. A slight extension of tone durations and shortening of
+ pauses is generally harmless. Three seconds will pretty much
+ _always_ be far more than three packet interarrival times.
+ However, that behavior is not required, so I'm going to leave it
+ with our legacy behavior for now. Code from
+ svn/asterisk/team/russell/issue_14460 (closes issue #14460)
+ Reported by: moliveras
+
+2009-02-12 10:16 +0000 [r175029] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * channels/chan_gtalk.c: Set the initiator attribute to lowercase
+ in our replies when receiving calls. This attribute contains a
+ JID that identifies the initiator of the GoogleTalk voice
+ session. The GoogleTalk client discards Asterisk's replies if the
+ initiator attribute contains uppercase characters. (closes issue
+ #13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded
+ by jcovert (license 551) Tested by: jcovert
+
+2009-02-12 00:19 +0000 [r174997] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: Revert RTP changes for continuation of DTMF. Proxy
+ commit by russell via SMS.
+
+2009-02-12 00:01 +0000 [r174985-174986] Russell Bryant <russell@digium.com>
+
+ * main/rtp.c: Clear out the current event after forcing the end of
+ a digit
+
+ * main/rtp.c: Fixify infinite DTMF in the case that no RFC2833 END
+ event is ever received
+
+2009-02-11 20:54 +0000 [r174885] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, apps/app_macro.c: Restore a behavior that was
+ recently changed, when we fixed issue #13962 and issue #13363
+ (related to issue #6176). When a hangup occurs during a Macro
+ execution in earlier 1.4, the h extension would execute within
+ the Macro context, whereas it was always supposed to execute only
+ within the main context (where Macro was called). So this fix
+ checks for an "h" extension in the deepest macro context where a
+ hangup occurred; if it exists, that "h" extension executes,
+ otherwise the main context "h" is executed. (closes issue #14122)
+ Reported by: wetwired Patches: 20090210__bug14122.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: andrew
+
+2009-02-10 18:50 +0000 [r174644] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Go off hold when we get an empty reinvite
+ telling us to. (closes issue #14448) Reported by: frawd Patches:
+ hold_invite_nosdp.patch uploaded by frawd (license 610)
+
+2009-02-10 17:52 +0000 [r174583] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/jitterbuf.c: Improve behavior of jitterbuffer when
+ maxjitterbuffer is set. This change improves the way the
+ jitterbuffer handles maxjitterbuffer and dramatically reduces the
+ number of frames dropped when maxjitterbuffer is exceeded. In the
+ previous jitterbuffer, when maxjitterbuffer was exceeded, all new
+ frames were dropped until the jitterbuffer is empty. This change
+ modifies the code to only drop frames until maxjitterbuffer is no
+ longer exceeded. Also, previously when maxjitterbuffer was
+ exceeded, dropped frames were not tracked causing stats for
+ dropped frames to be incorrect, this change also addresses that
+ problem. (closes issue #14044) Patches: bug14044-1.diff uploaded
+ by mnicholson (license 96) Tested by: mnicholson Review:
+ http://reviewboard.digium.com/r/144/
+
+2009-02-10 02:27 +0000 [r174369] Steve Murphy <murf@digium.com>
+
+ * apps/app_rpt.c: This patch solves some compiler complaints in
+ both 32 and 64-bit environments.
+
+2009-02-09 17:11 +0000 [r174282] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Don't do an SRV lookup if a port is
+ specified RFC 3263 says to do A record lookups on a hostname if a
+ port has been specified, so that's what we're going to do. See
+ section 4.2. (closes issue #14419) Reported by: klaus3000
+ Patches: patch_chan_sip_nosrvifport_1.4.23.txt uploaded by
+ klaus3000 (license 65)
+
+2009-02-09 14:48 +0000 [r174218] Joshua Colp <jcolp@digium.com>
+
+ * res/res_musiconhold.c: Don't overwrite our pointer to the music
+ class when music on hold stops. We will use this if it starts
+ again to see if we can resume the music where it left off.
+ (closes issue #14407) Reported by: mostyn
+
+2009-02-07 16:15 +0000 [r174148] Russell Bryant <russell@digium.com>
+
+ * res/snmp/agent.c: Fix a race condition that could cause a crash.
+
+2009-02-06 23:36 +0000 [r174082] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * channels/chan_sip.c: check ast_strlen_zero() before calling
+ ast_strdupa() in sip_uri_headers_cmp() and sip_uri_params_cmp()
+ The reporter didn't actually upload a properly-formed patch,
+ instead a modified chan_sip.c file was uploaded. I created a
+ patch to determine the changes, then modified the suggested
+ changes to create a proper fix. The summary above is a complete
+ description of the changes. (closes issue #13547) Reported by:
+ tecnoxarxa Patches: chan_sip.c.gz uploaded by tecnoxarxa (license
+ 258) Tested by: tecnoxarxa
+
+2009-02-06 17:15 +0000 [r173967-173968] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Remove a debug message I put in by accident.
+
+ * channels/chan_sip.c: Some clients do not put the call-id for
+ replaces at the beginning, so support it being anywhere in the
+ string. (closes issue #14350) Reported by: fhackenberger
+
+2009-02-06 16:20 +0000 [r173917] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Limit the addition of the Contact header in
+ SIP responses according to various SIP RFCs. (closes issue
+ #13602) Reported by: hjourdain Tested by: mnicholson
+
+2009-02-06 15:43 +0000 [r173900] Tilghman Lesher <tlesher@digium.com>
+
+ * utils/muted.c: Backport OS X fix from trunk (AGAIN, closes issue
+ #14360)
+
+2009-02-05 23:19 +0000 [r173770] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix logic regarding when to perform an SRV
+ lookup for outgoing REGISTER requests With this fix, we only will
+ perform an SRV lookup at the following times: * The first time we
+ register with a remote registrar * If we send a REGISTER but do
+ not receive a response * If the sendto() function returns an
+ error While I wrote the patch that fixes this issue, a huge
+ amount of credit is due to Brett Bryant, who wrote the initial
+ patch on which I based this one. (closes issue #12312) Reported
+ by: jrast Patches: 12312.patch uploaded by putnopvut (license 60)
+ Tested by: blitzrage Review: http://reviewboard.digium.com/r/132/
+
+2009-02-05 20:47 +0000 [r173696] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c: Add new configuration option to make shared
+ IMAP mailboxes function as expected. The new option is
+ "imapvmshareid" which is an ID to tag multiple mailboxes using
+ the same IMAP storage location to function as one mailbox. This
+ allows all messages to be retrieved for any user in the group.
+ The patch alters the 'X-Asterisk-VM-Extension' header that is
+ responsible for matching voicemails for a given user. (closes
+ issue #13673) Reported by: howardwilkinson
+
+2009-02-05 20:29 +0000 [r173392-173692] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix situations where queue members could be
+ autopaused unexpectedly Specifically, this patch prevents us from
+ autopausing members when we receive a busy or congestion frame
+ from them. (closes issue #14376) Reported by: fiddur Patches:
+ 14376.patch uploaded by putnopvut (license 60) Tested by: fiddur
+
+ * apps/app_mixmonitor.c: Add some missing cleanup to app_mixmonitor
+
+ * apps/app_mixmonitor.c: Fix a problem where a channel pointer
+ becomes invalid due to masquerading or hanging up. app_mixmonitor
+ runs its own thread to monitor the channel's activity and write
+ the mixed audio to a file. Since this thread runs independently
+ of the channel, it is possible that the mixmonitor thread's
+ channel pointer will point to freed memory when the channel
+ either is masqueraded or hangs up (technically, both cases are
+ hangups, but we need to handle the cases slightly differently).
+ The solution for this is to employ a datastore, which has the
+ nice benefit of allowing us to hook into channel masquerades and
+ hangups and update our pointer as necessary. If this looks
+ familiar, this same technique is employed in app_chanspy.
+ app_chanspy is a bit more involved since it does a lot more
+ operations on the channel that is being spied upon.
+ app_mixmonitor does have an extra touch that app_chanspy doesn't
+ have, though. Since there is a thread race between the channel's
+ thread and the mixmonitor thread on a hangup, we em- ploy a
+ condition-and-boolean combination to ensure that the channel
+ thread finishes with our structure before the mixmonitor thread
+ attempts to free it. No crashes! (closes issue #14374) Reported
+ by: aragon Patches: 14374.patch uploaded by putnopvut (license
+ 60) Tested by: aragon, putnopvut
+
+ * apps/app_chanspy.c: Revert my previous change because it was
+ stupid
+
+ * apps/app_chanspy.c: Add a missing unlock. Extremely unlikely to
+ ever matter, but it's needed.
+
+2009-02-03 23:35 +0000 [r173248] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: Fixes issue with IAX2 transfer not handing
+ off calls. Fixes issue with IAX2 transfers not taking place. As
+ it was, a call that was being transfered would never be handed
+ off correctly to the call ends because of how call numbers were
+ stored in a hash table. The hash table, "iax_peercallno_pvt",
+ storing all the current call numbers did not take into account
+ the complications associated with transferring a call, so a
+ separate hash table was required. This second hash table
+ "iax_transfercallno_pvt" handles calls being transfered, once the
+ call transfer is complete the call is removed from the transfer
+ hash table and added to the peer hash table resuming normal
+ operations. Addition functions were created to handle storing,
+ removing, and comparing items in the iax_transfercallno_pvt
+ table. (issue #13468) Review:
+ http://reviewboard.digium.com/r/140/
+
+2009-02-03 21:57 +0000 [r173211] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_features.c: Parking attempts made to one end of a bridge
+ no longer will hang up due to a parking failure. Parking attempts
+ made using either one-touch, or doing either a blind or assisted
+ transfer to the parking extension now keep up the bridge instead
+ of hanging up the attempted parked party. Normal causes for the
+ parking attempt to fail includes the specific specified extension
+ (via PARKINGEXTEN) not being available or if all the parking
+ spaces are currently in use. To avoid having to reverse a
+ masquerade park_space_reserve was made to provide foresight if a
+ parking attempt will succeed and if so reserve the parking space.
+ (closes issue #13494) Reported by: mdu113 Reviewed by Russell:
+ http://reviewboard.digium.com/r/133/
+
+2009-02-03 00:15 +0000 [r173070] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/extensions.conf.sample: Add warning to standard config,
+ that globals may be overridden by other dialplan configuration
+ files. (closes issue #14388) Reported by: macli
+
+2009-02-02 23:48 +0000 [r173066] Terry Wilson <twilson@digium.com>
+
+ * res/res_features.c: Fix a feature inheritance bug I added after
+ code review
+
+2009-02-02 20:28 +0000 [r172962] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample:
+ channels/chan_dahdi.c * Added doxygen comments to the major dahdi
+ structures. * Fixed PRI using an incorrect string value if the
+ extension delimiter is not present in the Dial() function. *
+ Fixed some uninitialized string variables on FXS ports.
+ configs/chan_dahdi.conf.sample * Updated some documentation.
+ These changes are already in trunk -r172400
+
+2009-01-31 00:15 +0000 [r172517-172639] Terry Wilson <twilson@digium.com>
+
+ * configs/features.conf.sample, res/res_features.c: Rename new
+ parkedcallparking option to parkedcallreparking Since this option
+ actually already existed in 1.6.0+, use the same name so as not
+ to confuse people when they upgrade
+
+ * configs/features.conf.sample, apps/app_dial.c,
+ main/global_datastores.c, res/res_features.c,
+ doc/channelvariables.txt, include/asterisk/global_datastores.h,
+ CHANGES: Fix feature inheritance with builtin features When using
+ builtin features like parking and transfers, the AST_FEATURE_*
+ flags would not be set correctly for all instances when either
+ performing a builtin attended transfer, or parking a call and
+ getting the timeout callback. Also, there was no way on a
+ per-call basis to specify what features someone should have on
+ picking up a parked call (since that doesn't involve the Dial()
+ command). There was a global option for setting whether or not
+ all users who pickup a parked call should have
+ AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or
+ PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan
+ variable which can be set either in the dialplan or with setvar
+ in channels that support it. This variable can be set to any
+ combination of 't', 'k', 'w', and 'h' (case insensitive matching
+ of the equivalent dial options), to set what features should be
+ activated on this channel. The patch moves the setting of the
+ features datastores into the bridging code instead of app_dial to
+ help facilitate this. 2) adds global options parkedcallparking,
+ parkedcallhangup, and parkedcallrecording to be similar to the
+ parkedcalltransfers option for globally setting features. 3) has
+ builtin_atxfer call builtin_parkcall if being transfered to the
+ parking extension since tracking everything through multiple
+ masquerades, etc. is difficult and error-prone 4) attempts to fix
+ all cases of return calls from parking and completed builtin
+ transfers not having the correct permissions (closes issue
+ #14274) Reported by: aragon Patches:
+ fix_feature_inheritence.diff.txt uploaded by otherwiseguy
+ (license 396) Tested by: aragon, otherwiseguy Review
+ http://reviewboard.digium.com/r/138/
+
+2009-01-29 22:54 +0000 [r172438] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, apps/app_nbscat.c, autoconf/ast_func_fork.m4,
+ apps/app_festival.c, build_tools/menuselect-deps.in, configure,
+ apps/app_dahdiras.c, apps/app_mp3.c, res/res_agi.c,
+ apps/app_externalivr.c, apps/app_ices.c, res/res_musiconhold.c:
+ Lose the CAP_NET_ADMIN at every fork, instead of at startup.
+ Otherwise, if Asterisk runs as a non-root user and the
+ administrator does a 'restart now', Asterisk loses the ability to
+ set QOS on packets. (closes issue #14004) Reported by: nemo
+ Patches: 20090105__bug14004.diff.txt uploaded by Corydon76
+ (license 14) Tested by: Corydon76
+
+2009-01-29 08:48 +0000 [r172169] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Make sure that we always add the hangupcause
+ headers. In some cases, the owner was disconnected before we
+ checked for the cause. This patch implements a temporary storage
+ in the pvt and use that instead. The code is based on ideas from
+ code from Adomjan in issue #13385 (Add support for Reason:
+ header) Thanks to Klaus Darillion for testing! (closes issue
+ #14294) related to issue #13385 Reported by: klaus3000 and
+ adomjan Patches: bug14294b.diff uploaded by oej (license 306)
+ Based on 20080829_chan_sip.c-q850reason_header.patch uploaded by
+ adomjan (license 487) Tested by: oej, klaus3000
+
+2009-01-28 18:51 +0000 [r172030] Steve Murphy <murf@digium.com>
+
+ * apps/app_channelredirect.c, main/pbx.c, main/manager.c,
+ res/res_features.c, include/asterisk/channel.h: This patch fixes
+ h-exten running misbehavior in manager-redirected situations.
+ What it does: 1. A new Flag value is defined in
+ include/asterisk/channel.h, AST_FLAG_BRIDGE_HANGUP_DONT, which
+ used as a messenge to the bridge hangup exten code not to run the
+ h-exten there (nor publish the bridge cdr there). It will done at
+ the pbx-loop level instead. 2. In the manager Redirect code, I
+ set this flag on the channel if the channel has a non-null pbx
+ pointer. I did the same for the second (chan2) channel, which
+ gets run if name2 is set... and the first succeeds. 3. I restored
+ the ending of the cdr for the pbx loop h-exten running code.
+ Don't know why it was removed in the first place. 4. The first
+ attempt at the fix for this bug was to place code directly in the
+ async_goto routine, which was called from a large number of
+ places, and could affect a large number of cases, so I tested
+ that fix against a fair number of transfer scenarios, both with
+ and without the patch. In the process, I saw that putting the fix
+ in async_goto seemed not to affect any of the blind or attended
+ scenarios, but still, I was was highly concerned that some other
+ scenarios I had not tested might be negatively impacted, so I
+ refined the patch to its current scope, and jmls tested both. In
+ the process, tho, I saw that blind xfers in one situation, when
+ the one-touch blind-xfer feature is used by the peer, we got
+ strange h-exten behavior. So, I inserted code to swap CDRs and to
+ set the HANGUP_DONT field, to get uniform behavior. 5. I added
+ code to the bridge to obey the HANGUP_DONT flag, skipping both
+ publishing the bridge CDR, and running the h-exten; they will be
+ done at the pbx-loop (higher) level instead. 6. I removed all the
+ debug logs from the patch before committing. 7. I moved the
+ AUTOLOOP set/reset in the h-exten code in res_features so it's
+ only done if the h-exten is going to be run. A very minor
+ performance improvement, but technically correct. (closes issue
+ #14241) Reported by: jmls Patches:
+ 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by
+ murf (license 17) Tested by: murf, jmls
+
+2009-01-28 17:25 +0000 [r171963] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c: Clarify log message (suggested by
+ manxpower on #asterisk-dev)
+
+2009-01-28 13:07 +0000 [r171837] Olle Johansson <oej@edvina.net>
+
+ * configs/sip.conf.sample: Add a better explanation of the
+ difference between the device namespace and the dialplan for
+ newbies.
+
+2009-01-27 21:55 +0000 [r171621-171689] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_agent.c: Fix devicestate problems for "always-on"
+ agent channels A revision to chan_agent attempted to "inherit"
+ the device state of the underlying channel in order to report the
+ device state of an agent channel more accurately. The problem
+ with the logic here is that it makes no sense to use this for
+ always-on agents. If the agent is logged in, then to the
+ underlying channel, the agent will always appear to be "in use,"
+ no matter if the agent is on a call or not. The reason is that to
+ the underlying channel, the channel is currently in use on a call
+ to the AgentLogin application. The most common cause that I found
+ for this issue to occur was for a SIP channel to be the
+ underlying channel type for an Agent channel. If the SIP phone
+ re-registers, then the registration will cause the device state
+ core to query the device state of the SIP channel. Since the SIP
+ channel is in use, the Agent channel would also inherit this
+ status. Once the agent channel was set to "in use" there was no
+ way that the device state could change on that channel unless the
+ agent logged out. The solution for this problem is a bit
+ different in 1.4 than it is in the other branches. In 1.4, there
+ will be a one-line fix to make sure that only callback agents
+ will inherit device state from their underlying channel type. For
+ the other branches of Asterisk, since callback support has been
+ removed, there is also no need for device state inheritance in
+ chan_agent, so I will simply be removing it from the code. In
+ addition, the 1.4 source is getting a new comment to help the
+ next person who edits chan_agent.c. I'm adding a comment that a
+ agent_pvt's loginchan field may be used to determine if the agent
+ is a callback agent or not. (closes issue #14173) Reported by:
+ nathan Patches: 14173.patch uploaded by putnopvut (license 60)
+ Tested by: nathan, aramirez
+
+ * main/slinfactory.c: Prevent a crash from occurring when a jitter
+ buffer interpolated frame is removed from a slinfactory
+ slinfactory used the "samples" field of an ast_frame in order to
+ determine the amount of data contained within the frame. In
+ certain cases, such as jitter buffer interpolated frames, the
+ frame would have a non-zero value for "samples" but have NULL
+ "data" This caused a problem when a memcpy call in
+ ast_slinfactory_read would attempt to access invalid memory. The
+ solution in use here is to never feed frames into the slinfactory
+ if they have NULL "data" (closes issue #13116) Reported by:
+ aragon Patches: 13116.diff uploaded by putnopvut (license 60)
+
+2009-01-27 14:33 +0000 [r171527] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Use the same branch tag in CANCEL as in
+ INVITE Originally putnopvut implemented some changes in revision
+ 142079 that according to the bug report seemed to have worked
+ then, but somehow fails now. I guess code, as humans, get old and
+ forget stuff. Anyway, this bug caused CANCEL not to work with
+ picky systems. Thanks Fredrik for pointing out where the bug in
+ the SIP messaging was. (closes issue #14346) Reported by: oej
+ Patches: bug14346.diff uploaded by oej (license 306) Tested by:
+ oej
+
+2009-01-26 21:31 +0000 [r171452] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Resolve some synchronization issues in
+ chan_iax2 scheduler handling. The important changes here are
+ related to the synchronization between threads adding items into
+ the scheduler and the scheduler handling thread. By adjusting the
+ lock and condition handling, we ensure that the scheduler thread
+ sleeps no longer and no less than it is supposed to. We also
+ ensure that it does not wake up more often than it has to. There
+ is no bug report associated with this. It is just something that
+ I found while putting scheduler thread handling into a reusable
+ form (review 129). Review: http://reviewboard.digium.com/r/131/
+
+2009-01-26 12:51 +0000 [r171264] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Don't retransmit 401 on REGISTER requests
+ when alwaysauthreject=yes (closes issue #14284) Reported by:
+ klaus3000 Patches: patch_chan_sip_unreliable_1.4.23_14284.txt
+ uploaded by klaus3000 (license 65) Tested by: klaus3000
+
+2009-01-25 23:44 +0000 [r171120-171187] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_oss.c: Correctly track the hookstate (closes issue
+ #13686) Reported by: itiliti Patches: 20081013__bug13686.diff.txt
+ uploaded by Corydon76 (license 14)
+
+ * res/res_agi.c: Err, yeah.
+
+ * res/res_agi.c: Add thread to kill zombies, when child processes
+ don't die immediately on SIGHUP. (closes issue #13968) Reported
+ by: eldadran Patches: 20090114__bug13968.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: eldadran
+
+2009-01-25 13:33 +0000 [r170979] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_page.c: Resolve a logic error that was causing Page() to
+ crash when more than one channel was specified. (closes issue
+ #14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt
+ uploaded by seanbright (license 71) Tested by: kc0bvu
+
+2009-01-24 13:55 +0000 [r170836] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/res_odbc.conf.sample: Remove superfluous implementation
+ note (closes issue #14319)
+
+2009-01-23 20:55 +0000 [r170671-170719] Mark Michelson <mmichelson@digium.com>
+
+ * configs/res_odbc.conf.sample: Add notes to the idlecheck
+ explanation in res_odbc.conf.sample (closes issue #14319)
+ Reported by: klaus3000 Patches:
+ patch_idlecheck_res_odbc.conf.sample.txt uploaded by klaus3000
+ (license 65)
+
+ * contrib/i18n.testsuite.conf: Update contrib/i18n.testsuite.conf
+ to not use deprecated syntax * Convert Wait,1 to Wait(1) *
+ Convert SetLanguage to Set(CHANNEL(language)) * Use 'n' for all
+ priorities beyond the first Also added test for Chinese numbers,
+ too. (closes issue #14320) Reported by: dant Patches:
+ i18n.testsuite.conf.issue14320.v2.diff uploaded by dant (license
+ 670)
+
+2009-01-23 20:16 +0000 [r170648] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c: When a channel is answered make sure any
+ indications currently playing stop. Usually the phone would do
+ this but if the channel was already answered then they are being
+ generated by Asterisk and we darn well need to stop them. (closes
+ issue #14249) Reported by: RadicAlish
+
+2009-01-23 Tilghman Lesher <tlesher@digium.com>
+
+ * Asterisk 1.4.23.1 released.
+
+ * channels/chan_iax2.c: Regression fix for AST-2009-001 security
+ fix.
+
+2009-01-21 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.4.23 released.
+
+2009-01-20 18:49 -0500 [r169581] Terry Wilson <twilson@digium.com>
+
+ * One-touch parking was calling back the wrong channel on timeout
+
+2009-01-20 13:40 -0500 [r169485] Terry Wilson <twilson@digium.com>
+
+ * Don't play audio to the channel if we've masqueraded (closes
+ issue #14066) Reported by: bluefox Tested by: otherwiseguy,
+ bluefox
+
+2009-01-16 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.23-rc4 released.
+
+2009-01-16 00:19 +0000 [r168745] Steve Murphy <murf@digium.com>
+
+ * pbx/pbx_ael.c: This patch fixes a problem where a goto (or jump,
+ in this case) fails a consistency check because it can't find a
+ matching extension. The problem was a missing instruction to end
+ the range notation in the code where it converts the pattern into
+ a regex and uses the regex code to determine the match. I tested
+ using the AEL code the user supplied, and now, the consistency
+ check passes. (closes issue #14141) Reported by: dimas
+
+2009-01-15 18:43 +0000 [r168721] Olle Johansson <oej@edvina.net>
+
+ * configs/extconfig.conf.sample: Meetme actually has realtime but
+ wasn't documented
+
+2009-01-15 18:22 +0000 [r168716] Terry Wilson <twilson@digium.com>
+
+ * res/res_features.c: Convert call to park_call_full to
+ masq_park_call_announce Since we removed the AST_PBX_KEEPALIVE
+ return value, we need to use masqueraded parking, otherwise we
+ will try to call ast_hangup() in __pbx_run() and in
+ do_parking_thread() and then promptly crash. (closes issue
+ #14215) Reported by: waverly360 Tested by: otherwiseguy (closes
+ issue #14228) Reported by: kobaz Tested by: otherwiseguy
+
+2009-01-15 01:20 +0000 [r168633] Tilghman Lesher <tlesher@digium.com>
+
+ * /: Blocked revision 168632 from /branches/1.2: 1.2 regression on
+ security fix AST-2009-001 (Closes issue #14238)
+
+2009-01-15 00:11 +0000 [r168628] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix some crashes from bad datastore handling in
+ app_queue.c * The queue_transfer_fixup function was searching for
+ and removing the datastore from the incorrect channel, so this
+ was fixed. * Most datastore operations regarding the
+ queue_transfer datastore were being done without the channel
+ locked, so proper channel locking was added, too. (closes issue
+ #14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by
+ putnopvut (license 60) Tested by: ZX81, festr
+
+2009-01-14 21:48 +0000 [r168622] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c: * Fixed create_process() allocation of
+ process ID values. The allocated process IDs could overflow their
+ respective NT and TE fields. Affects outgoing calls.
+
+2009-01-14 20:52 +0000 [r168614] Sean Bright <sean.bright@gmail.com>
+
+ * contrib/scripts/autosupport: Update autosupport script to supply
+ info for both Zaptel and DAHDI in 1.4 and be sure to run
+ dahdi_test in 1.6.x and trunk instead of zttest. (closes issue
+ #14132) Reported by: dsedivec Patches:
+ asterisk-1.4-autosupport.patch uploaded by dsedivec (license 638)
+ asterisk-trunk-autosupport.patch uploaded by dsedivec (license
+ 638)
+
+2009-01-14 19:34 +0000 [r168608] Steve Murphy <murf@digium.com>
+
+ * apps/app_page.c: app_page was failing to compile in dev-mode on
+ my gcc-4.2.4 system. This change gets rid of the warning.
+
+2009-01-14 19:02 +0000 [r168603] Tilghman Lesher <tlesher@digium.com>
+
+ * main/udptl.c: Don't read into a buffer without first checking if
+ a value is beyond the end. (closes issue #13600) Reported by:
+ atis Patches: 20090106__bug13600.diff.txt uploaded by Corydon76
+ (license 14) Tested by: atis
+
+2009-01-14 16:19 +0000 [r168598] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_agent.c: Fix a logic error I found while searching
+ through chan_agent.c I found that the allow_multiple_logins
+ function would never return 0 due to an incorrect comparison
+ being used when traversing the list of agents. While I was
+ modifying this function, I also did a little bit of coding
+ guidelines cleanup, too.
+
+2009-01-14 01:27 +0000 [r168593] Terry Wilson <twilson@digium.com>
+
+ * apps/app_page.c: Don't overflow when paging more than 128
+ extensions The number of available slots for calls in app_page
+ was hardcoded to 128. Proper bounds checking was not in place to
+ enforce this limit, so if more than 128 extensions were passed to
+ the Page() app, Asterisk would crash. This patch instead
+ dynamically allocates memory for the ast_dial structures and
+ removes the (non-functional) arbitrary limit. This issue would
+ have special importance to anyone who is dynamically creating the
+ argument passed to the Page application and allowing more than
+ 128 extensions to be added by an outside user via some external
+ interface. The patch posted by a_villacis was slightly modified
+ for some coding guidelines and other cleanups. Thanks,
+ a_villacis! (closes issue #14217) Reported by: a_villacis
+ Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch
+ uploaded by a (license 660) Tested by: otherwiseguy
+
+2009-01-13 19:13 +0000 [r168561] Russell Bryant <russell@digium.com>
+
+ * main/indications.c, main/channel.c, apps/app_read.c,
+ channels/chan_misdn.c, funcs/func_channel.c,
+ include/asterisk/indications.h, apps/app_disa.c, main/app.c,
+ res/snmp/agent.c, include/asterisk/channel.h,
+ res/res_indications.c: Revert unnecessary indications API change
+ from rev 122314
+
+2009-01-13 18:34 +0000 [r168551] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Don't pass a value with a side effect to a
+ macro (closes issue #14176) Reported by: paraeco Patches:
+ chan_sip.c.diff uploaded by paraeco (license 658)
+
+2009-01-13 17:48 +0000 [r168546] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_logic.c: If either conditional is NULL, don't try
+ copying it. (closes issue #14226) Reported by: caspy Patches:
+ 20090113__bug14226.diff.txt uploaded by Corydon76 (license 14)
+
+2009-01-12 21:42 +0000 [r168507-168516] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_agi.c: (closes issue #13881) Reported by: hoowa Update
+ the app CDR field for AGI commands that are not executing an
+ application via "exec".
+
+ * channels/chan_agent.c: (closes issue #12269) Reported by: IgorG
+ Tested by: denisgalvao This gits rid of the notion of an
+ owning_app allowing the request and hangup to be initiated by
+ different threads. Originating from an active agent channel
+ requires this. The implementation primarily changes __login_exec
+ to wait on a condition variable rather than a lock. Review:
+ http://reviewboard.digium.com/r/35/
+
+2009-01-12 14:58 +0000 [r168482] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: I am reverting the fix made in revision
+ 168128 (and its upward merges) after being contacted by Olle
+ Johansson and being shown how this fix is incorrect. Thanks to
+ Olle for clearing this up for me.
+
+2009-01-12 14:57 +0000 [r168480] Russell Bryant <russell@digium.com>
+
+ * configs/indications.conf.sample: s/ringdance/ringcadence/ for
+ Bulgaria
+
+2009-01-10 20:47 +0000 [r168267-168382] Kevin P. Fleming <kpfleming@digium.com>
+
+ * README: small commit to test new server
+
+ * README: small commit to test new server
+
+ * sounds/Makefile: update to use new sound file packages that
+ include license files
+
+2009-01-09 22:14 +0000 [r168198] Russell Bryant <russell@digium.com>
+
+ * res/res_musiconhold.c: Make this compile for mvanbaak
+
+2009-01-09 21:28 +0000 [r168191] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: * Fix for JIRA AST-175/ABE-1757 *
+ Miscellaneous doxygen comments added.
+
+2009-01-09 20:08 +0000 [r168128] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Add check_via calls to more request handlers
+ INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests were not
+ checking the topmost Via to determine where to send the response.
+ Adding check_via calls to those request handlers solves this.
+ (closes issue #13071) Reported by: baron Patches: check_via.patch
+ uploaded by baron (license 531) Tested by: baron
+
+2009-01-08 22:08 +0000 [r167840] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_agi.c: Don't truncate database results at 255 chars.
+ (closes issue #14069) Reported by: evandro Patches:
+ 20081214__bug14069.diff.txt uploaded by Corydon76 (license 14)
+
+2009-01-08 17:24 +0000 [r167620-167714] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: remove an unnecessary argument to
+ queue_request()
+
+ * channels/chan_sip.c: When a SIP request or response arrives for a
+ dialog with an associated Asterisk channel, and the lock on that
+ channel cannot be obtained because it is held by another thread,
+ instead of dropping the request/response, queue it for later
+ processing when the channel lock becomes available.
+ http://reviewboard.digium.com/r/117/
+
+2009-01-07 22:35 +0000 [r167432-167566] Russell Bryant <russell@digium.com>
+
+ * main/file.c: Fix the last couple of places where free() was
+ improperly used directly.
+
+ * main/file.c: Don't fclose() the file early, the filestream
+ destructor will handle it.
+
+ * main/file.c: Only try to close the file if one was actually
+ opened
+
+ * main/file.c: Don't use free() directly. This caused a crash since
+ ast_filestream is now an ao2 object. Reported by JunK-Y on IRC,
+ #asterisk-dev
+
+ * main/indications.c: Treat an empty string the same way as a NULL
+ country argument. In passing, simplify the handling of returning
+ a default tone zone.
+
+2009-01-06 21:35 +0000 [r167299] Mark Michelson <mmichelson@digium.com>
+
+ * main/db.c: Use the correct variable when creating the format
+ string (closes issue #14177) Reported by: nic_bellamy Patches:
+ asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic
+ (license 299)
+
+2009-01-06 20:48 +0000 [r167260] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 167259 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06
+ Jan 2009) | 2 lines Security fix AST-2009-001. ........
+
+2009-01-05 16:51 +0000 [r167179] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: A couple of changes to T.38 SDP attribute
+ handling There are some boolean attributes for T.38 such as
+ T38FaxFillBitRemoval, T38FaxTranscodingMMR, and
+ T38FaxTranscodingJBIG. By simply being present, we should treat
+ these as a "true" value. The current code, however, was requiring
+ a 1 or 0 as the value of the attribute in order to parse it. This
+ is due to the fact that there are some T.38 endpoints and
+ gateways that also transmit this information incorrectly. This
+ patch follows the "be liberal in what you accept and strict in
+ what you send" philosophy by accepting both the correctly- and
+ incorrectly-formatted attributes, but only sending information as
+ it is supposed to be sent. It was also discovered that a
+ particular type of T.38 gateway sends some non-standard T.38 SDP
+ attributes. Instead of using T38FaxMaxDatagram and T38MaxBitRate,
+ it used T38MaxDatagram and T38FaxMaxRate respectively. We now
+ will properly accept these attributes as well. Note that there
+ are a lot of patches cited in the below commit message template.
+ This is because the person who submitted these patches is an
+ awesome person and wrote 1.4, 1.6.0, and 1.6.1 variants. (closes
+ issue #13976) Reported by: linulin Patches:
+ chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648)
+ chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648)
+ chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648)
+ chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov
+ (license 648) chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded
+ by arcivanov (license 648)
+ chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov
+ (license 648) Tested by: arcivanov
+
+2009-01-01 00:01 +0000 [r166953-167095] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_alsa.c: Repeat attempts to write when we receive
+ -EAGAIN from the driver, as detailed in the ALSA sample code (see
+ http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32)
+ Reported by: Jerry Geis (via the -users list) Fixed by: me
+ (license 14)
+
+ * channels/chan_local.c: Also inherit the musiconhold class.
+ (Closes #14153) Reported by: Jerry Geis, via the users list.
+ Patch by: me (license 14)
+
+2008-12-28 15:13 +0000 [r166772] Russell Bryant <russell@digium.com>
+
+ * channels/misdn_config.c: Use strncat() instead of an sprintf() in
+ which source and target buffers overlap
+ http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html
+
+2008-12-23 15:35 +0000 [r166592] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, channels/chan_iax2.c: Compile, even if both
+ DAHDI and Zaptel are not installed. (Closes issue #14120)
+
+2008-12-23 15:16 +0000 [r166568] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c: Fix a crash resulting from a datastore with
+ inheritance but no duplicate callback The fix for this is to
+ simply set the newly created datastore's data pointer to NULL if
+ it is inherited but has no duplicate callback. (closes issue
+ #14113) Reported by: francesco_r Patches: 14113.patch uploaded by
+ putnopvut (license 60) Tested by: francesco_r
+
+2008-12-23 04:05 +0000 [r166509] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c: Use the integer form of condition for integer
+ comparisons. (closes issue #14127) Reported by: andrew
+
+2008-12-22 20:56 +0000 [r166380] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_dahdi.c: Fix a deadlock relating to channel locks
+ and autoservice It has been discovered that if a channel is
+ locked prior to a call to ast_autoservice_stop, then it is likely
+ that a deadlock will occur. The reason is that the call to
+ ast_autoservice_stop has a check built into it to be sure that
+ the thread running autoservice is not currently trying to
+ manipulate the channel we are about to pull out of autoservice.
+ The autoservice thread, however, cannot advance beyond where it
+ currently is, though, because it is trying to acquire the lock of
+ the channel for which autoservice is attempting to be stopped.
+ The gist of all this is that a channel MUST NOT be locked when
+ attempting to stop autoservice on the channel. In this particular
+ case, the channel was locked by a call to ast_read. A call to
+ ast_exists_extension led to autoservice being started and stopped
+ due to the existence of dialplan switches. It may be that there
+ are future commits which handle the same symptoms but in a
+ different location, but based on my looks through the code, it is
+ very rare to see a construct such as this one. (closes issue
+ #14057) Reported by: rtrauntvein Patches: 14057v3.patch uploaded
+ by putnopvut (license 60) Tested by: rtrauntvein Review:
+ http://reviewboard.digium.com/r/107/
+
+2008-12-22 17:22 +0000 [r166262-166297] Russell Bryant <russell@digium.com>
+
+ * main/utils.c: Fix up timeout handling in ast_carefulwrite().
+
+ * include/asterisk/strings.h, res/res_musiconhold.c: Re-work ref
+ count handling of MoH classes using astobj2 to resolve crashes.
+ (closes issue #13566) Reported by: igorcarneiro Tested by:
+ russell Review: http://reviewboard.digium.com/r/106/
+
+2008-12-19 23:34 +0000 [r166157] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, funcs/func_audiohookinherit.c (added),
+ channels/chan_sip.c, include/asterisk/audiohook.h,
+ main/audiohook.c, CHANGES: Backport of AUDIOHOOK_INHERIT for
+ Asterisk 1.4 (closes issue #13538) Reported by: mbit Patches:
+ 13538.patch uploaded by putnopvut (license 60) Tested by:
+ putnopvut
+
+2008-12-19 22:30 +0000 [r166093] Steve Murphy <murf@digium.com>
+
+ * apps/app_dial.c, res/res_features.c, include/asterisk/pbx.h,
+ apps/app_queue.c: This merges the masqpark branch into 1.4 These
+ changes eliminate the need for (and use of) the KEEPALIVE return
+ code in res_features.c; There are other places that use this
+ result code for similar purposes at a higher level, these appear
+ to be left alone in 1.4, but attacked in trunk. The reason these
+ changes are being made in 1.4, is that parking ends a channel's
+ life, in some situations, and the code in the bridge (and some
+ other places), was not checking the result code properly, and
+ dereferencing the channel pointer, which could lead to memory
+ corruption and crashes. Calling the masq_park function eliminates
+ this danger in higher levels. A series of previous commits have
+ replaced some parking calls with masq_park, but this patch puts
+ them ALL to rest, (except one, purposely left alone because a
+ masquerade is done anyway), and gets rid of the code that tests
+ the KEEPALIVE result, and the NOHANGUP_PEER result codes. While
+ bug 13820 inspired this work, this patch does not solve all the
+ problems mentioned there. I have tested this patch (again) to
+ make sure I have not introduced regressions. Crashes that
+ occurred when a parked party hung up while the parking party was
+ listening to the numbers of the parking stall being assigned, is
+ eliminated. These are the cases where parking code may be
+ activated: 1. Feature one touch (eg. *3) 2. Feature blind xfer to
+ parking lot (eg ##700) 3. Run Park() app from dialplan (eg sip
+ xfer to 700) (eg. dahdi hookflash xfer to 700) 4. Run Park via
+ manager. The interesting testing cases for parking are: I. A
+ calls B, A parks B a. B hangs up while A is getting the numbers
+ announced. b. B hangs up after A gets the announcement, but
+ before the parking time expires c. B waits, time expires, A is
+ redialed, A answers, B and A are connected, after which, B hangs
+ up. d. C picks up B while still in parking lot. II. A calls B, B
+ parks A a. A hangs up while B is getting the numbers announced.
+ b. A hangs up after B gets the announcement, but before the
+ parking time expires c. A waits, time expires, B is redialed, B
+ answers, A and B are connected, after which, A hangs up. d. C
+ picks up A while still in parking lot. Testing this throroughly
+ involves acting all the permutations of I and II, in situations
+ 1,2,3, and 4. Since I added a few more changes (ALL references to
+ KEEPALIVE in the bridge code eliimated (I missed one earlier), I
+ retested most of the above cases, and no crashes. H-extension
+ weirdness. Current h-extension execution is not completely
+ correct for several of the cases. For the case where A calls B,
+ and A parks B, the 'h' exten is run on A's channel as soon as the
+ park is accomplished. This is expected behavior. But when A calls
+ B, and B parks A, this will be current behavior: After B parks A,
+ B is hung up by the system, and the 'h' (hangup) exten gets run,
+ but the channel mentioned will be a derivative of A's... Thus, if
+ A is DAHDI/1, and B is DAHDI/2, the h-extension will be run on
+ channel Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info
+ will be those relating to Channel A. And, in the case where A is
+ reconnected to B after the park time expires, when both parties
+ hang up after the joyful reunion, no h-exten will be run at all.
+ In the case where C picks up A from the parking lot, when either
+ A or C hang up, the h-exten will be run for the C channel. CDR's
+ are a separate issue, and not addressed here. As to WHY this
+ strange behavior occurs, the answer lies in the procedure
+ followed to accomplish handing over the channel to the parking
+ manager thread. This procedure is called masquerading. In the
+ process, a duplicate copy of the channel is created, and most of
+ the active data is given to the new copy. The original channel
+ gets its name changed to XXX<ZOMBIE> and keeps the PBX
+ information for the sake of the original thread (preserving its
+ role as a call originator, if it had this role to begin with),
+ while the new channel is without this info and becomes a call
+ target (a "peer"). In this case, the parking lot manager thread
+ is handed the new (masqueraded) channel. It will not run an
+ h-exten on the channel if it hangs up while in the parking lot.
+ The h exten will be run on the original channel instead, in the
+ original thread, after the bridge completes. See bug 13820 for
+ our intentions as to how to clean up the h exten behavior.
+ Review: http://reviewboard.digium.com/r/29/
+
+2008-12-19 19:48 +0000 [r165991] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/dahdi_compat.h, main/asterisk.c, main/channel.c,
+ apps/app_dahdibarge.c, channels/chan_dahdi.c, apps/app_meetme.c,
+ apps/app_dahdiscan.c, codecs/codec_dahdi.c,
+ res/res_musiconhold.c, channels/chan_iax2.c: (closes issue
+ #13480) Reported by: tzafrir Replace a bunch of if defined checks
+ for Zaptel/DAHDI through several new defines in dahdi_compat.h.
+ This removes a lot of code duplication. Example from bug: #ifdef
+ HAVE_ZAPTEL fd = open("/dev/zap/pseudo", O_RDWR); #else fd =
+ open("/dev/dahdi/pseudo", O_RDWR); #endif is replaced with: fd =
+ open(DAHDI_FILE_PSEUDO, O_RDRW);
+
+2008-12-19 15:03 +0000 [r165796-165889] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c: Ensure that the chanspy datastore is fully
+ initialized. This patch resolved some random crash issues
+ observed by a user on a BSD system (closes issue #14111) Reported
+ by: ys Patches: app_chanspy.c.diff uploaded by ys (license 281)
+
+ * main/utils.c: Make ast_carefulwrite() be more careful. This patch
+ handles some additional cases that could result in partial writes
+ to the file description. This was done to address complaints
+ about partial writes on AMI. (issue #13546) (more changes needed
+ to address potential problems in 1.6) Reported by: srt Tested by:
+ russell Review: http://reviewboard.digium.com/r/99/
+
+2008-12-18 21:14 +0000 [r165767] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Add mutexes around accesses to the IMAP
+ library interface. This prevents certain crashes, especially when
+ shared mailboxes are used. (closes issue #13653) Reported by:
+ howardwilkinson Patches:
+ asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by
+ howardwilkinson (license 590) Tested by: jpeeler
+
+2008-12-18 18:52 +0000 [r165661] Russell Bryant <russell@digium.com>
+
+ * res/res_musiconhold.c: Set the process group ID on the MOH
+ process so that all children will get killed (closes issue
+ #14099) Reported by: caspy Patches:
+ res_musiconhold.c.patch.killpg.try2 uploaded by caspy (license
+ 645)
+
+2008-12-18 17:11 +0000 [r165537-165591] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: Only care about a compatible codec for early bridging
+ if we are actually bridging to another channel. If we are not we
+ actually want to bring the audio back to us. (closes issue
+ #13545) Reported by: davidw
+
+ * apps/app_followme.c: Do not crash if we are not passed in a
+ followme id. (closes issue #14106) Reported by: ys Patches:
+ app_followme.c.2.diff uploaded by ys (license 281)
+
+2008-12-17 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.23-rc3 released.
+
+2008-12-17 21:14 +0000 [r165317] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_macro.c: Reverse the fix from issue #6176 and add proper
+ handling for that issue. (Closes issue #13962, closes issue
+ #13363) Fixed by myself (license 14)
+
+2008-12-17 20:51 +0000 [r164977-165255] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_meetme.c, apps/app_realtime.c, apps/app_directory.c,
+ apps/app_queue.c: Fix some memory leaks found while looking at
+ how realtime configs are handled. Also cleaned up some coding
+ guidelines violations in app_realtime.c, mostly related to
+ spacing
+
+ * channels/chan_sip.c: After looking through SIP registration code
+ most of the day, this is one of the few things I could find that
+ was just plain wrong. Even though it probably isn't possible for
+ it to happen, it seems weird to have code that checks if a
+ pointer is NULL and then immediately dereferences that pointer if
+ it was NULL.
+
+2008-12-16 21:38 +0000 [r164672-164881] Russell Bryant <russell@digium.com>
+
+ * main/utils.c: Fix an issue where DEBUG_THREADS may erroneously
+ report that a thread is exiting while holding a lock. If the last
+ lock attempt was a trylock, and it failed, it will still be in
+ the list of locks so that it can be reported. (closes issue
+ #13219) Reported by: pj
+
+ * apps/app_macro.c: Do not dereference the channel if
+ AST_PBX_KEEPALIVE has been returned. This is a bug I noticed
+ while looking at the code for app_macro. This return code means
+ that another thread has assumed ownership of the channel and it
+ can no longer be touched. (I hate this return code with a
+ passion, by the way.)
+
+ * main/manager.c: Add "restart gracefully" to the AMI blacklist of
+ CLI commands. "module unload" was already identified as a command
+ that can not be used from the AMI. "restart gracefully"
+ effectively unloads all modules, and will run in to the same
+ problems. (closes issue #13894) Reported by: kernelsensei
+
+ * include/asterisk/threadstorage.h, main/threadstorage.c: Fix
+ memory leak and invalid reporting issues with DEBUG_THREADLOCALS.
+ One issue was that the ast_mutex_* API was being used within the
+ context of the thread local data destructors. We would go off and
+ allocate more thread local data while the pthread lib was in the
+ middle of destroying it all. This led to a memory leak. Another
+ issue was an invalid argument being provided to the the
+ object_add API call. (closes issue #13678) Reported by: ys Tested
+ by: Russell
+
+ * channels/chan_sip.c: Fix a memory leak related to the use of the
+ "setvar" configuration option. The problem was that these
+ variables were being appended to the list of vars on the sip_pvt
+ every time a re-registration or re-subscription came in. Since
+ it's just a waste of memory to put them there unless the request
+ was an INVITE, then the fix is to check the request type before
+ copying the vars. (closes issue #14037) Reported by: marvinek
+ Tested by: russell
+
+2008-12-16 15:15 +0000 [r164634] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c: I added a sentence to clarify why - and ' ' are
+ ignored in patterns as per bug 14076. Leif says he'll put some
+ stuff about it in the extensions.conf sample, etc.
+
+2008-12-16 14:28 +0000 [r164605] Russell Bryant <russell@digium.com>
+
+ * res/res_musiconhold.c: Don't try to change working directory if a
+ directory was not configured. (closes issue #14089) Reported by:
+ caspy
+
+2008-12-15 19:53 +0000 [r164416-164422] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/pbx.h: Add the deadlock note to
+ ast_spawn_extension as well
+
+ * include/asterisk/channel.h, include/asterisk/pbx.h: Add notes to
+ autoservice and pbx doxygen regarding a potential deadlock
+ scenario so that it is avoided in the future
+
+2008-12-15 18:11 +0000 [r164204-164350] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Do not try to unlock a non-existant channel
+ if the transfer fails. (closes issue #13800) Reported by: dwagner
+ Patches: asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety
+ (license 608)
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/channel.h: Use autoconf logic to determine
+ whether the system has timersub or not. Do not blindly assume
+ Solaris does not. (closes issue #13838) Reported by: ano
+
+ * apps/app_dial.c: Can we try not to assign an unsigned int to -1?
+ (closes issue #14074) Reported by: wetwired
+
+2008-12-15 14:31 +0000 [r164201] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, res/res_features.c: Handle a case where a call
+ can be bridged to a channel that is still ringing. The issue that
+ was reported was about a case where a RINGING channel got
+ redirected to an extension to pick up a call from parking. Once
+ the parked call got taken out of parking, it heard silence until
+ the other side answered. Ideally, the caller that was parked
+ would get a ringing indication. This patch fixes this case so
+ that the caller receives ringback once it comes out of parking
+ until the other side answers. The fixes are: - Make sure we
+ remember that a channel was an outgoing channel when doing a
+ masquerade. This prevents an erroneous ast_answer() call on the
+ channel, which causes a bogus 200 OK to be sent in the case of
+ SIP. - Add some additional comments to explain related parts of
+ code. - Update the handling of the ast_channel visible_indication
+ field. Storing values that are not stateful is pointless. Control
+ frames that are events or commands should be ignored. - When a
+ bridge first starts, check to see if the peer channel needs to be
+ given ringing indication because the calling side is still
+ ringing. - Rework ast_indicate_data() a bit for the sake of
+ readability. (closes issue #13747) Reported by: davidw Tested by:
+ russell Review: http://reviewboard.digium.com/r/90/
+
+2008-12-13 23:22 +0000 [r164082] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_dial.c: Change the default calldurationlimit from the
+ special value 0 to -1, so we can better detect an exceptional
+ case. This follows on to the changes made in revision 156386.
+ Related to issue #13851. (closes issue #13974) Reported by:
+ paradise Patches: 20081208__bug13974.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: file, blitzrage, ZX81
+
+2008-12-12 22:20 +0000 [r163785] Russell Bryant <russell@digium.com>
+
+ * /: Set the reviewboard:url property on 1.4, as well
+
+2008-12-12 22:03 +0000 [r163761] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, main/editline/read.c: Simple fix for Ctrl-C not
+ immediately exiting Asterisk, but also add a pointer inside
+ editline to look back to asterisk.c, so others don't spend as
+ much time as I did looking (in the wrong place) for the
+ appropriate function. Reported by: ZX81, via the #asterisk-users
+ channel Fixed by: me (license 14)
+
+2008-12-12 14:40 +0000 [r163448-163511] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_dundi.c: Specify uint32_t for variables storing a CRC32
+ so that it is actually 32 bits on 64-bit machines, as well.
+ (inspired by issue #13879)
+
+ * main/channel.c, main/autoservice.c, include/asterisk/channel.h:
+ Resolve issues that could cause DTMF to be processed out of
+ order. These changes come from team/russell/issue_12658 1) Change
+ autoservice to put digits on the head of the channel's frame
+ readq instead of the tail. If there were frames on the readq that
+ autoservice had not yet read, the previous code would have
+ resulted in out of order processing. This required a new API call
+ to queue a frame to the head of the queue instead of the tail. 2)
+ Change up the processing of DTMF in ast_read(). Some of the
+ problems were the result of having two sources of pending DTMF
+ frames. There was the dtmfq and the more generic readq. Both were
+ used for pending DTMF in various scenarios. Simplifying things to
+ only use the frame readq avoids some of the problems. 3) Fix a
+ bug where a DTMF END frame could get passed through when it
+ shouldn't have. If code set END_DTMF_ONLY in the middle of digit
+ emulation, and a digit arrived before emulation was complete,
+ digits would get processed out of order. (closes issue #12658)
+ Reported by: dimas Tested by: russell, file Review:
+ http://reviewboard.digium.com/r/85/
+
+2008-12-11 23:35 +0000 [r163383] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c: When a Ctrl-C or Ctrl-D ends a remote console,
+ on certain shells, the terminal is messed up. By intercepting
+ those events with a signal handler in the remote console, we can
+ avoid those issues. (closes issue #13464) Reported by: tzafrir
+ Patches: 20081110__bug13464.diff.txt uploaded by Corydon76
+ (license 14) Tested by: blitzrage
+
+2008-12-11 22:44 +0000 [r163316] Matt Nicholson <mnicholson@digium.com>
+
+ * pbx/pbx_dundi.c: Clean up the dundi cache every 5 minutes.
+ (closes issue #13819) Reported by: adomjan Patches:
+ pbx_dundi.c-clearcache.patch uploaded by adomjan (license 487)
+ dundi_clearecache3.diff uploaded by mnicholson (license 96)
+ Tested by: adomjan
+
+2008-12-11 21:46 +0000 [r163092-163253] Russell Bryant <russell@digium.com>
+
+ * funcs/func_strings.c, funcs/func_cut.c: Fix some observed
+ slowdowns in dialplan processing. The change is to remove
+ autoservice usage from dialplan functions that do not need it
+ because they do not perform operations that potentially block.
+ (closes issue #13940) Reported by: tbelder
+
+ * res/res_features.c: Fix an issue that made it so you could only
+ have a single caller executing a custom feature at a time. This
+ was especially problematic when custom features ran for any
+ appreciable amount of time. The fix turned out to be quite
+ simple. The dynamic features are now stored in a read/write list
+ instead of a list using a mutex. (closes issue #13478) Reported
+ by: neutrino88 Fix suggested by file
+
+2008-12-11 16:51 +0000 [r163088] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_agi.c: Don't wait forever, if there's a specified
+ recording timeout. (closes issue #13885) Reported by: bamby
+ Patches: res_agi.c.patch uploaded by bamby (license 430)
+
+2008-12-11 16:46 +0000 [r163080-163084] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Revert this cast to long. Using time_t here
+ causes build failures on a FreeBSD 32-bit build.
+
+ * apps/app_queue.c: Fix a potential crash due to unsafe datastore
+ handling. This patch also contains a conversion from using long
+ to time_t for representing times for a queue, as well as some
+ whitespace fixes. (closes issue #14060) Reported by: nivek
+ Patches: datastore_fixup.patch.corrected uploaded by nivek
+ (license 636) with slight modification from me Tested by: nivek
+
+2008-12-10 22:52 +0000 [r162874-162926] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_musiconhold.c: Oops, inverted logic for a strcasecmp
+ check. Pointed out by mmichelson, thanks!
+
+ * res/res_musiconhold.c: (closes issue #13229) Reported by:
+ clegall_proformatique Ensure that moh_generate does not return
+ prematurely before local_ast_moh_stop is called. Also, the sleep
+ in mp3_spawn now only occurs for http locations since it seems to
+ have been added originally only for failing media streams.
+
+2008-12-10 19:01 +0000 [r162738-162804] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix subscription based MWI up a bit. We only
+ want to put sip: at the beginning of the URI if it is not already
+ there and revert code to ignore destination check if subscribing
+ for MWI. (closes issue #12560) Reported by: vsauer Patches:
+ patch001.diff uploaded by ramonpeek (license 266)
+
+ * channels/chan_sip.c: When a SIP peer unregisters set the expiry
+ time back to 0 so that the 200 OK contains an expires of 0.
+ (closes issue #13599) Reported by: hjourdain Patches:
+ chan_sip.c.diff uploaded by hjourdain (license 583)
+
+2008-12-10 16:45 +0000 [r162671] Steve Murphy <murf@digium.com>
+
+ * pbx/ael/ael_lex.c, pbx/ael/ael.flex: (closes issue #14022)
+ Reported by: wetwired Tested by: murf I checked, and I added a
+ mod to the trunk version of Asterisk that would make it 8-bit
+ transparent on 27 Nov 2007, but I made no such updates to 1.4. My
+ best guess is that 1.4 was released, and it was not appropriate
+ to commit an enhancement. But I'm going to add the same fixes to
+ 1.4 now, for the following reasons: 1. wetwired is correct; 1.4
+ is **mostly** 8-bit transparent now. This is because the lexical
+ token forming rules use . in most 'word' state continuances. It's
+ just the beginning of a 'word' that is picky. 2. Accepting 8-bit
+ chars in some places and not others leads to bug reports like
+ this.
+
+2008-12-10 16:44 +0000 [r162659-162670] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/stringfields.h: Update to stringfield handling
+ so that side-effects on parameters are not evaluated multiple
+ times. An example where this caused a problem was in chan_sip.c,
+ with the line ast_string_field_set(p, fromdomain, ++fromdomain);
+ This patch was originally uploaded to issue #13783 by jamessan.
+ While the issue was closed for other reasons, this patch is valid
+ and fixes a separate problem, and is thus being committed.
+
+ * channels/chan_sip.c: Revert fix for issue 13570. It has caused
+ more problems than it helped to fix. (closes issue #13783)
+ Reported by: navkumar (closes issue #14025) Reported by: ffs
+
+ * doc/misdn.txt: Add missing documentation to misdn.txt (closes
+ issue #14052) Reported by: festr Patches: misdn.txt.patch
+ uploaded by festr (license 443)
+
+2008-12-10 16:05 +0000 [r162653] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: Increment the sequence number on the end packets for
+ RFC2833. After reading the RFC some more and doing some testing I
+ agree with this change. (closes issue #12983) Reported by: vt
+ Patches: dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license
+ 520)
+
+2008-12-09 23:08 +0000 [r162463] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Oops, should be "tz", not "zonetag".
+
+2008-12-09 22:17 +0000 [r162413] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c, include/asterisk/utils.h, main/utils.c: Remove
+ the test_for_thread_safety() function completely. The test is not
+ valid. Besides, if we actually suspected that recursive mutexes
+ were not working, we would get a ton of LOG_ERROR messages when
+ DEBUG_THREADS is turned on. (inspired by a discussion on the
+ asterisk-dev list)
+
+2008-12-09 21:53 +0000 [r162348] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: We appear to have documented tz= in the
+ [general] section of voicemail.conf, without actually having
+ implemented it. Oops. (Reported by Olivier on the -users list)
+
+2008-12-09 21:14 +0000 [r162341] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_directed_pickup.c: Add 'down' as a valid state for
+ directed call pickup. This creeps up when we receive session
+ progress when dialing a device and not ringing. (closes issue
+ #14005) Reported by: ddl
+
+2008-12-09 20:57 +0000 [r162286] Russell Bryant <russell@digium.com>
+
+ * apps/app_meetme.c: Fix an issue where callers on an incoming call
+ on an SLA trunk would not hear ringback. We need to make sure
+ that we don't start writing audio to the trunk channel until
+ we're actually ready to answer it. Otherwise, the channel driver
+ will treat it as inband progress, even though all they are
+ getting is silence. (closes issue #12471) Reported by: mthomasslo
+
+2008-12-09 20:44 +0000 [r162273] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_festival.c: Fix double declaration of 'x' on the PPC
+ platform. (closes issue #14038) Reported by: ffloimair
+
+2008-12-09 20:28 +0000 [r162265] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c: If we fail to start a thread for the pbx to run in,
+ we need to be sure to decrease the number of active calls on the
+ system. This fix may relate to ABE-1713, but it is not certain
+ yet.
+
+2008-12-09 20:20 +0000 [r162264] Steve Murphy <murf@digium.com>
+
+ * pbx/ael/ael_lex.c, pbx/ael/ael.flex: In discussion with
+ seanbright on #asterisk-dev, I have added a default rule, and an
+ option to suppress the default rule from being generated in the
+ flex output, for the sake of those OS's where they didn't tweak
+ flex's ECHO macro, and the compiler doesn't like it. The
+ regressions are OK with this.
+
+2008-12-09 19:47 +0000 [r162188-162204] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: Make sure that the timestamp for DTMF is not the same
+ as the previous voice frame and do not send audio when
+ transmitting DTMF as this confuses some equipment. (closes issue
+ #13209) Reported by: ip-rob Patches: 13209.diff uploaded by file
+ (license 11) Tested by: ip-rob, bujones
+
+ * main/rtp.c: Take video into account when early bridging RTP.
+ (closes issue #13535) Reported by: davidw
+
+2008-12-09 18:13 +0000 [r162136] Steve Murphy <murf@digium.com>
+
+ * pbx/ael/ael_lex.c, pbx/ael/ael.flex: Previous fix used ast_malloc
+ and ast_copy_string and messed up the standalone stuff. Fixed.
+
+2008-12-09 17:07 +0000 [r162071] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_phone.c: For some reason, after a distclean, gcc
+ started returning 'value computed is not used'. Fixing (for
+ --enable-dev-mode).
+
+2008-12-09 16:46 +0000 [r162014] Russell Bryant <russell@digium.com>
+
+ * apps/app_disa.c: Allow DISA to handle extensions that start with
+ #. (closes issue #13330) Reported by: jcovert
+
+2008-12-09 16:31 +0000 [r162013] Steve Murphy <murf@digium.com>
+
+ * pbx/ael/ael_lex.c, pbx/pbx_ael.c, include/asterisk/ael_structs.h,
+ pbx/ael/ael.flex: (closes issue #14019) Reported by: ckjohnsonme
+ Patches: 14019.diff uploaded by murf (license 17) Tested by:
+ ckjohnsonme, murf This crash was the result of a few small errors
+ that would combine in 64-bit land to result in a crash. 32-bit
+ land might have seen these combine to mysteriously drop the args
+ to an application call, in certain circumstances. Also, in trying
+ to find this bug, I spotted a situation in the flex input, where,
+ in passing back a 'word' to the parser, it would allocate a
+ buffer larger than necessary. I changed the usage in such
+ situations, so that strdup was not used, but rather, an
+ ast_malloc, followed by ast_copy_string. I removed a field from
+ the pval struct, in u2, that was never getting used, and set in
+ one spot in the code. I believe it was an artifact of a previous
+ fix to make switch cases work invisibly with extens. And, for
+ goto's I removed a '!' from before a strcmp, that has been there
+ since the initial merging of AEL2, that might prevent the proper
+ target of a goto from being found. This was pretty harmless on
+ its own, as it would just louse up a consistency check for users.
+ Many thanks to ckjohnsonme for providing a simplified and
+ complete set of information about the bug, that helped
+ considerably in finding and fixing the problem. Now, to get
+ aelparse up and running again in trunk, and out of its "horribly
+ broken" state, so I can run the regression suite!
+
+2008-12-09 14:52 +0000 [r161948] Russell Bryant <russell@digium.com>
+
+ * main/app.c: Fix a problem with GROUP() settings on a masquerade.
+ The previous code carried over group settings from the old
+ channel to the new one. However, it did nothing with the group
+ settings that were already on the new channel. This patch removes
+ all group settings that already existed on the new channel. I
+ have a more complicated version of this patch which addresses
+ only the most blatant problem with this, which is that a channel
+ can end up with multiple group settings in the same category.
+ However, I could not think of a use case for keeping any of the
+ group settings from the old channel, so I went this route for
+ now. (closes AST-152)
+
+2008-12-08 17:52 +0000 [r161725] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Make the usereqphone option work again.
+ (closes issue #13474) Reported by: mmaguire Patches:
+ 20080912_bug13474.diff uploaded by mmaguire (license 571)
+
+2008-12-05 21:02 +0000 [r161426] Sean Bright <sean.bright@gmail.com>
+
+ * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions
+ 161421 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r161421 | seanbright | 2008-12-05 15:50:23 -0500 (Fri, 05 Dec
+ 2008) | 8 lines Fix build errors on FreeBSD (uint -> unsigned
+ int). (closes issue #14006) Reported by: alphaque Patches:
+ astobj2.h-patch uploaded by alphaque (license 259) (Slightly
+ modified by seanbright) ........
+
+2008-12-05 16:51 +0000 [r161354] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * utils/smsq.c: kill a warning
+
+2008-12-05 14:12 +0000 [r161287] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c: Fix a NULL format string warning found by buildbot.
+
+2008-12-04 18:30 +0000 [r161013] Jeff Peeler <jpeeler@digium.com>
+
+ * main/rtp.c: (closes issue #13835) Reported by: matt_b Tested by:
+ jpeeler This mirrors a check that was present in ast_rtp_read to
+ also be in ast_rtp_raw_write to not schedule sending the receiver
+ report if the remote RTCP endpoint address isn't present in the
+ RTCP structure. Closes AST-142.
+
+2008-12-04 16:44 +0000 [r160943] Mark Michelson <mmichelson@digium.com>
+
+ * main/callerid.c: Fix a callerid parsing issue. If someone
+ formatted callerid like the following: "name <number>" (including
+ the quotation marks), then the parts would be parsed as name:
+ "name number: number This is because the closing quotation mark
+ was not discovered since the number and everything after was
+ parsed out of the string earlier. Now, there is a check to see if
+ the closing quote occurs after the number, so that we can know if
+ we should strip off the opening quote on the name. Closes AST-158
+
+2008-12-03 21:54 +0000 [r160770] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Some compilers warn on null format strings;
+ some don't (caught by buildbot)
+
+2008-12-03 21:38 +0000 [r160764] Jason Parker <jparker@digium.com>
+
+ * channels/chan_agent.c: Only show this warning when we want to
+ show it. (closes issue #13982) Reported by: coolmig Patches:
+ chan_agent.c.patch uploaded by coolmig (license 621)
+
+2008-12-03 20:41 +0000 [r160703] Steve Murphy <murf@digium.com>
+
+ * funcs/func_callerid.c: (closes issue #13597) Reported by:
+ john8675309 Patches: patch.13597 uploaded by murf (license 17)
+ Tested by: murf, john8675309 This patch causes the setcid func to
+ update the CDR clid after setting the channel field.
+
+2008-12-03 17:55 +0000 [r160480-160570] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: During bridge code, the channel bridge may
+ return a retry code, if a transfer was initiated but not yet
+ completed. If the bridge is immediately retried, then we may send
+ a storm of TXREQ packets, even though the first set is sent
+ reliably (retransmitted). Fixes AST-137.
+
+ * pbx/pbx_spool.c: If an entry is added to the directory during a
+ scan when another entry expires, then that new entry will not be
+ processed promptly, but must wait for either a future entry to
+ start or a current entry's retry to occur. If no other entries
+ exist in the directory (other than the new entries) when a bunch
+ expire, then the new entries must wait until another new entry is
+ added to be processed. This was a rather weird race condition,
+ really. Fixes AST-147.
+
+ * pbx/pbx_spool.c: Don't start scanning the directory until all
+ modules are loaded, because some required modules (channels,
+ apps, functions) may not yet be in memory yet. Fixes AST-149.
+
+ * channels/chan_sip.c: Jon Bonilla (Manwe) pointed out on the -dev
+ list: "I guess that having only ip-phones in mind is not a good
+ approach. Since it is possible to have a sip proxy connected to
+ asterisk we could receive a 407 (unauthorized) or 483 (too many
+ hops) as response and dialog ending would not be a good
+ behavior." So modified.
+
+2008-12-02 23:58 +0000 [r160390-160411] Terry Wilson <twilson@digium.com>
+
+ * res/res_features.c: Channel is masqueraded, don't keep alive
+
+ * res/res_features.c: A situation like A calls B, A builtin_atxfers
+ B to C, C parks B would lead to a crash. Thanks to file for
+ telling me how to fix it! (closes issue #13854) Reported by: Adam
+ Lee Tested by: otherwiseguy
+
+2008-12-02 17:42 +0000 [r160297] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: When the text does not match exactly (e.g.
+ RTP/SAVP), then the %n conversion fails, and the resulting
+ integer is garbage. Thus, we must initialize the integer and
+ check it afterwards for success. (closes issue #14000) Reported
+ by: folke Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by
+ folke (license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded
+ by folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff
+ uploaded by folke (license 626)
+
+2008-12-02 01:16 +0000 [r160266] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/astmm.h: make compile with dev mode and malloc
+ debug
+
+2008-12-02 00:25 +0000 [r160207] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/stringfields.h, apps/app_voicemail.c,
+ main/pbx.c, main/frame.c: Ensure that Asterisk builds with
+ --enable-dev-mode, even on the latest gcc and glibc.
+
+2008-12-01 Tilghman Lesher <tilghman@digium.com>
+
+ * Released 1.4.23-rc2
+
+2008-12-01 17:27 +0000 [r160003] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Apply some logic used in iax2_indicate() to
+ iax2_setoption(), as well, since they both have the potential to
+ send control frames in the middle of call setup. We have to wait
+ until we have received a message back from the remote end before
+ we try to send any more frames. Otherwise, the remote end will
+ consider it invalid, and we'll get stuck in an INVAL/VNAK storm.
+
+2008-12-01 16:08 +0000 [r159976] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/manager.c: Get rid of the useless format string and argument
+ in the Bogus/ manager channelname. Noted by kpfleming and name
+ Bogus/manager suggested by eliel
+
+2008-12-01 14:52 +0000 [r159900] Russell Bryant <russell@digium.com>
+
+ * .cleancount: Force a "make clean" to avoid a bizarre build issue
+ ...
+
+2008-12-01 14:05 +0000 [r159897] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/manager.c: make manager compile on OpenBSD. The last (10th)
+ argument to ast_channel_alloc here should be a pointer and NULL
+ is not really a pointer.
+
+2008-11-29 16:58 +0000 [r159808] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/enum.c, utils/frame.c, configure, res/res_agi.c,
+ include/asterisk/module.h, main/logger.c, main/dns.c,
+ include/asterisk/threadstorage.h, include/asterisk/utils.h,
+ include/asterisk/devicestate.h, channels/chan_sip.c,
+ include/asterisk/dundi.h, main/jitterbuf.c,
+ channels/chan_agent.c, configure.ac, utils/astman.c,
+ include/asterisk/cli.h, include/asterisk/channel.h,
+ include/jitterbuf.h, include/asterisk/manager.h,
+ main/ast_expr2.c, Makefile, include/asterisk/logger.h,
+ include/asterisk/res_odbc.h, main/srv.c, channels/chan_misdn.c,
+ include/asterisk/linkedlists.h, include/asterisk/lock.h,
+ include/asterisk/strings.h, makeopts.in,
+ include/asterisk/stringfields.h, utils/check_expr.c,
+ channels/chan_vpb.cc, res/res_features.c, channels/chan_iax2.c:
+ update dev-mode compiler flags to match the ones used by default
+ on Ubuntu Intrepid, so all developers will see the same warnings
+ and errors since this branch already had some printf format
+ attributes, enable checking for them and tag functions that
+ didn't have them format attributes in a consistent way
+
+2008-11-26 20:21 +0000 [r159476-159571] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_oss.c, channels/busy.h (removed),
+ channels/ring_tone.h (added), channels/chan_alsa.c,
+ channels/ringtone.h (removed), channels/busy_tone.h (added),
+ channels/Makefile: rename these files so as to avoid conflicts
+ when users update their working copies and have unversioned files
+ already in place
+
+ * channels, agi/Makefile, utils/Makefile, channels/busy.h (added),
+ Makefile.moddir_rules, Makefile.rules, channels/ringtone.h
+ (added), channels/Makefile: simplify (and slightly bug-fix) the
+ recent developer-oriented COMPILE_DOUBLE mode add channels/busy.h
+ and channels/ringtone.h to the repository instead of generating
+ them repeatedtly; most users do not change the settings to build
+ them, but the Makefile rules are still there if they wish to do
+ so ensure that 'make clean' removes dependency files for .i files
+ that are created in COMPILE_DOUBLE mode
+
+2008-11-25 22:41 +0000 [r159316] Steve Murphy <murf@digium.com>
+
+ * main/cdr.c, channels/chan_iax2.c: (closes issue #12694) Reported
+ by: yraber Patches: 12694.2nd.diff uploaded by murf (license 17)
+ Tested by: murf, laurav Thanks to file (Joshua Colp) for his IAX
+ fix. the change to cdr.c allows no-answer to percolate up into
+ CDR's, and feels like the right place to locate this fix; if BUSY
+ is done here, no-answer should be, too.
+
+2008-11-25 21:56 +0000 [r159246-159269] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Don't try to send a response on a NULL pvt.
+ (closes issue #13919) Reported by: barthpbx Patches:
+ chan_iax2.c.patch uploaded by eliel (license 64) Tested by:
+ barthpbx
+
+ * /, channels/chan_iax2.c: Merged revisions 159245 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25
+ Nov 2008) | 7 lines Regression fix for last security fix. Set the
+ iseqno correctly. (closes issue #13918) Reported by: ffloimair
+ Patches: 20081119__bug13918.diff.txt uploaded by Corydon76
+ (license 14) Tested by: ffloimair ........
+
+2008-11-25 17:34 +0000 [r159158] Russell Bryant <russell@digium.com>
+
+ * main/astobj2.c, include/asterisk/astobj2.h: Add ao2_trylock() to
+ go along with ao2_lock() and ao2_unlock()
+
+2008-11-25 16:23 +0000 [r159096] Terry Wilson <twilson@digium.com>
+
+ * apps/app_festival.c: Add missing variable declaration in the PPC
+ code
+
+2008-11-25 04:50 +0000 [r159025] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_rpt.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: System call ioperm is non-portable, so check for
+ its existence in autoconf. (Closes issue #13863)
+
+2008-11-22 00:04 +0000 [r158629] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/dahdi_compat.h, channels/chan_dahdi.c: (closes
+ issue #13786) Reported by: tzafrir When compiling against Zaptel
+ dahdi_compat will now only define all the DAHDI defines if the
+ Zaptel define is present. Also, there is no such thing as
+ DAHDI_PRI.
+
+2008-11-21 23:14 +0000 [r158603] Steve Murphy <murf@digium.com>
+
+ * res/res_features.c: In reference to the fix made for 13871, I was
+ merging the fix into 1.6.0 and realized I missed the code in the
+ h-exten block, and didn't catch it because my test case had the
+ h-exten commented out. So, this corrects the code I missed, as a
+ preventative against another crash report. Tested with the
+ h-exten defined, all is well.
+
+2008-11-21 23:07 +0000 [r158600] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: The passed extension may not be the same in the list
+ as the current entry, because we strip spaces when copying the
+ extension into the structure. Therefore, use the copied item to
+ place the item into the list. (found by lmadsen on -dev, fixed by
+ me)
+
+2008-11-21 22:05 +0000 [r158539] Russell Bryant <russell@digium.com>
+
+ * main/astobj2.c, include/asterisk/astobj2.h: When compiling with
+ DEBUG_THREADS, report the real file/func/line for
+ ao2_lock/ao2_unlock
+
+2008-11-21 21:19 +0000 [r158483] Steve Murphy <murf@digium.com>
+
+ * res/res_features.c: (closes issue #13871) Reported by: mdu113
+ This one is totally my fault. The code doesn't even create a
+ bridge if the channel CDR has POST_DISABLED. I didn't check for
+ that at the end of the bridge. Fixed with a few small insertions.
+ Tested. Looks good. No cdr generated, no crash, no unnecc. data
+ objects created either.
+
+2008-11-21 15:24 +0000 [r158053-158306] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: This change had somehow gotten reverted due to
+ a completely unrelated commit. Thanks to Theo Belder on the
+ Asterisk-dev list for pointing this out.
+
+ * include/asterisk/file.h, main/frame.c, main/file.c,
+ include/asterisk/frame.h: There was an issue when attempting to
+ reference an embedded frame in a freed ast_filestream. This patch
+ makes use of the ao2 functions to make sure that we do not free
+ an ast_filestream structure until the embedded ast_frame has been
+ "freed" as well. (closes issue #13496) Reported by: fst-onge
+ Patches: filestream_frame_1_4.diff uploaded by putnopvut (license
+ 60) Tested by: putnopvut Closes AST-89
+
+ * channels/chan_sip.c: We don't handle 4XX responses to BYE well.
+ According to section 15 of RFC 3261, we should terminate a dialog
+ if we receive a 481 or 408 in response to our BYE. Since I am
+ aware of at least one phone manufacturer who may sometimes send a
+ 404 as well, I am being liberal and saying that any 4XX response
+ to a BYE should result in a terminated dialog. (closes issue
+ #12994) Reported by: pabelanger Patches: 12994.patch uploaded by
+ putnopvut (license 60) Closes AST-129
+
+ * apps/app_dial.c, channels/chan_sip.c: Make sure to set the hangup
+ cause on the calling channel in the case that ast_call() fails.
+ For incoming SIP channels, this was causing us to send a 603
+ instead of a 486 when the call-limit was reached on the
+ destination channel. (closes issue #13867) Reported by: still_nsk
+ Patches: 13867.diff uploaded by putnopvut (license 60) Tested by:
+ blitzrage
+
+2008-11-20 01:46 +0000 [r158010] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: Merged revision 157977 from
+ https://origsvn.digium.com/svn/asterisk/team/group/issue8824
+ ........ Fixes JIRA ABE-1726 The dial extension could be empty if
+ you are using MISDN_KEYPAD to control ISDN provider features.
+
+2008-11-19 21:34 +0000 [r157859] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/stdtime/Makefile, codecs/gsm/src, main/db1-ast/btree,
+ channels/misdn/Makefile, main/db1-ast/recno, pbx/ael, channels,
+ main/db1-ast/Makefile, main/stdtime, main/db1-ast/hash,
+ codecs/gsm/Makefile, main/db1-ast/db, Makefile.moddir_rules,
+ channels/misdn, main/db1-ast/mpool, pbx/Makefile, Makefile.rules,
+ res/snmp, res/Makefile: the gcc optimizer frequently finds broken
+ code (use of uninitalized variables, unreachable code, etc.),
+ which is good. however, developers usually compile with the
+ optimizer turned off, because if they need to debug the resulting
+ code, optimized code makes that process very difficult. this
+ means that we get code changes committed that weren't adequately
+ checked over for these sorts of problems. with this build system
+ change, if (and only if) --enable-dev-mode was used and
+ DONT_OPTIMIZE is turned on, when a source file is compiled it
+ will actually be preprocessed (into a .i or .ii file), then
+ compiled once with optimization (with the result sent to
+ /dev/null) and again without optimization (but only if the first
+ compile succeeded, of course). while making these changes, i did
+ some cleanup work in Makefile.rules to move commonly-used
+ combinations of flag variables into their own variables, to make
+ the file easier to read and maintain
+
+2008-11-18 22:47 +0000 [r157503] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Add some missing invite state changes
+ necessary in the sip_write function. Not setting the invite state
+ correctly on the call was resulting in the Record application
+ leaving empty files. I also have updated the doxygen comment next
+ to the declaration of the INV_EARLY_MEDIA constant to reflect
+ that we also use this state when we *send* a 18X response to an
+ INVITE. (closes issue #13878) Reported by: nahuelgreco Patches:
+ sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco
+ (license 162) Tested by: putnopvut
+
+2008-11-18 19:13 +0000 [r157365] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_meetme.c: (closes issue #13899) Reported by: akkornel
+ This fix is the result of a bug fix in ast_app_separate_args
+ r124395. If an argument does not exist it should always be set to
+ a null string rather than a null pointer.
+
+2008-11-18 18:25 +0000 [r157305] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, channels/chan_local.c, res/res_features.c,
+ include/asterisk/channel.h, apps/app_followme.c: Fix a crash in
+ the end_bridge_callback of app_dial and app_followme which would
+ occur at the end of an attended transfer. The error occurred
+ because we initially stored a pointer to an ast_channel which
+ then was hung up due to a masquerade. This commit adds a "fixup"
+ callback to the bridge_config structure to allow for
+ end_bridge_callback_data to be changed in the case that a new
+ channel pointer is needed for the end_bridge_callback.
+
+2008-11-15 19:31 +0000 [r157104-157163] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, Makefile.rules: when an individual directory dist-clean
+ is run, run clean in that directory first, and when running
+ top-level dist-clean, do not run subdirectory clean operations
+ twice
+
+ * Makefile.moddir_rules: dist-clean should remove dependency
+ information files as well
+
+ * contrib/asterisk-ng-doxygen: major update to doxygen
+ configuration file: 1) update to doxygen 1.5.x style file, as
+ used in trunk 2) tell doxygen where are header files are, so
+ include-file processing can be done 3) make all macros that are
+ used to define variables/functions be expanded, so that doxygen
+ will properly document the resulting variable/function 4) make
+ all macros that are used to provide the contents of a variable
+ (structure) be expanded, so that doxygen will be able to document
+ the resulting fields 5) suppress compiler attributes
+ (__attribute__(xxx)) from being seen by doxygen, so it will
+ properly match up function definition and usage (for an example
+ of th effect of this, look at the doxygen docs for ast_log() from
+ before and afte this commit)
+
+2008-11-14 15:18 +0000 [r156816] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: If the prompt to reenter a voicemail
+ password timed out, it resulted in the password not being saved,
+ even if the input matched what you gave when first prompted to
+ enter a new password. This is because the return value of
+ ast_readstring was checked, but not checked properly. This bug
+ was discovered by Jared Smith during an Asterisk training course.
+ Thanks for reporting it!
+
+2008-11-14 00:41 +0000 [r156688-156755] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_while.c: ast_waitfordigit() requires that the channel be
+ up, for no good logical reason. This prevents While/EndWhile from
+ working within the "h" extension. Reported by: jgalarneau (for
+ ABE C.2) Fixed by: me
+
+ * main/manager.c: Provide more space for all the data which can
+ appear in an originating channel name. (closes issue #13398)
+ Reported by: bamby Patches: manager.c.diff uploaded by bamby
+ (license 430)
+
+2008-11-13 11:58 +0000 [r156485-156510] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configure, autoconf/ast_gcc_attribute.m4: revert this change...
+ non-functional changes don't belong here
+
+ * configure, autoconf/ast_gcc_attribute.m4: correct minor syntax
+ error... no functional change
+
+2008-11-12 21:18 +0000 [r156386] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_dial.c: When using call limits under 1 second, infinite
+ call lengths are allowed, instead. (closes issue #13851) Reported
+ by: ruddy
+
+2008-11-12 19:36 +0000 [r156297] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c: It turns out that the 0x0XX00 codes being returned
+ for N, X, and Z are off by one, as per conversation with jsmith
+ on #asterisk-dev; he was teaching a class and disconcerted that
+ this published rule was not being followed, with patterns _NXX,
+ _[1-8]22 and _[2-9]22... and NXX was winning, but [1-8] should
+ have been. This change, tested on these 3 patterns now picks the
+ proper one. However, this change may surprise users who set up
+ dialplans based on previous behavior, which has been there for
+ what, 2 and half years or so now.
+
+2008-11-12 19:26 +0000 [r156294] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c: If the SLA thread is not started, then reload
+ causes a memory leak. (closes issue #13889) Reported by: eliel
+ Patches: app_meetme.c.patch uploaded by eliel (license 64)
+
+2008-11-12 19:10 +0000 [r156289] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_meetme.c: For whatever reason, gcc only warned me about
+ the possible use of an uninitialized variable when compiling
+ 1.6.1.
+
+2008-11-12 18:39 +0000 [r156229] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Revert revision 132506, since it
+ occasionally caused IAX2 HANGUP packets not to be sent, and
+ instead, schedule a task to destroy the iax2 pvt structure 10
+ seconds later. This allows the IAX2 HANGUP packet to be queued,
+ transmitted, and ACKed before the pvt is destroyed. (closes issue
+ #13645) Reported by: dzajro Patches:
+ 20081111__bug13645__3.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: vazir Reviewed: http://reviewboard.digium.com/r/51/
+
+2008-11-12 17:53 +0000 [r156178] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_meetme.c: (closes issue #13173) Reported by: pep This
+ change adds an announce_thread responsible for playing
+ announcements to an existing conference. This allows all
+ announcing to be immediately stopped if necessary but more
+ importantly allows other threads that need to play something to
+ not block. There are multiple benefits to this, but the actual
+ bug is for solving the scenario for a channel to be unusable
+ after hang up for the entire duration of the parting
+ announcement. The parting announcement can be extremely long
+ depending on what the user recorded upon joining the conference.
+ Reviewed by Russell on Review Board:
+ http://reviewboard.digium.com/r/25/
+
+2008-11-12 17:38 +0000 [r156167] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c: When doing some tests, I was having a crash at
+ the end of every call if an attended transfer occurred during the
+ call. I traced the cause to the CDR on one of the channels being
+ NULL. murf suggested a check in the end bridge callback to be
+ sure the CDR is non-NULL before proceeding, so that's what I'm
+ adding.
+
+2008-11-12 17:29 +0000 [r156164] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c: Move the sanity check that makes sure "always
+ fork" is not set along with the console option to be after the
+ code that reads options from asterisk.conf. This resolves a
+ situation where Asterisk can start taking up 100% when
+ misconfigured. (Thanks to Bryce Porter (x86 on IRC) for letting
+ me log in to his system to figure out what was causing the 100%
+ CPU problem.)
+
+2008-11-10 21:07 +0000 [r155861] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_agent.c: Channel drivers assume that when their
+ indicate callback is invoked, that the channel on which the
+ callback was called is locked. This patch corrects an instance in
+ chan_agent where a channel's indicate callback is called directly
+ without first locking the channel. This was leading to some
+ observed locking issues in chan_local, but considering that all
+ channel drivers operate under the same expectations, the generic
+ fix in chan_agent is the right way to go. AST-126
+
+2008-11-10 20:49 +0000 [r155803] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/valgrind.txt: I got tired of saying this in every single
+ bugnote referring to this file.
+
+2008-11-09 01:08 +0000 [r155553] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_dial.c, res/res_features.c, include/asterisk/channel.h,
+ apps/app_followme.c: Use static functions here instead of nested
+ ones. This requires a small change to the ast_bridge_config
+ struct as well. To understand the reason for this change, see the
+ following post:
+ http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html
+
+2008-11-07 22:27 +0000 [r155398] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Clarify error message. (closes issue #13809)
+ Reported by: denke Patches: 20081104__bug13809.diff.txt uploaded
+ by Corydon76 (license 14) Tested by: denke
+
+2008-11-06 19:45 +0000 [r155011] Mark Michelson <mmichelson@digium.com>
+
+ * configs/voicemail.conf.sample: The documentation listed the
+ ability to set 'maxmsg' per context. The truth is that you can
+ only set this in the general section or per mailbox. Thus I am
+ updating the sample config file to be more accurate. Thanks to
+ sasargen on IRC for bringing up this issue.
+
+2008-11-05 16:44 +0000 [r154724] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_agent.c: The logic of a strcasecmp call was
+ reversed (closes issue #13841) Reported by: clegall_proformatique
+
+2008-11-05 16:06 +0000 [r154685] Steve Murphy <murf@digium.com>
+
+ * main/channel.c: This fix was prompted by communication from user,
+ who was seeing thousands of error logs... looks like EAGAIN. Made
+ such uninteresting.
+
+2008-11-04 20:49 +0000 [r154365] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: On busy systems, it's possible for the
+ values checked within a single line of code to change, unless the
+ structure is locked to ensure a consistent state. (closes issue
+ #13717) Reported by: kowalma Patches: 20081102__bug13717.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: kowalma
+
+2008-11-04 19:01 +0000 [r154266] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: JIRA ABE-1703 mISDN sets the channel to
+ the wrong state when it receives the indication
+ AST_CONTROL_RINGING.
+
+2008-11-04 18:58 +0000 [r154060-154263] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_h323.c: Make the monitor thread non-detached, so it
+ can be joined (suggested by Russell on -dev list).
+
+ * apps/app_voicemail.c: Attempting to expunge a mailbox when the
+ mailstream is NULL will crash Asterisk. (Closes issue #13829)
+ Reported by: jaroth Patch by: me (modified jaroth's patch)
+
+ * main/rtp.c: Remove the potential for a division by zero error.
+ (Closes issue #13810)
+
+2008-11-03 13:01 +0000 [r153823] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_oss.c, channels/chan_dahdi.c, funcs/func_odbc.c,
+ main/file.c, main/http.c, main/utils.c, pbx/pbx_config.c,
+ res/res_jabber.c: somehow missed a bunch of gcc 4.3.x warnings in
+ this branch on the first pass
+
+2008-11-02 19:51 +0000 [r153651] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/features.h: features.h depends on linkedlists.h,
+ so include it
+
+2008-11-01 18:22 +0000 [r153337] Kevin P. Fleming <kpfleming@digium.com>
+
+ * utils/frame.c, main/cli.c, utils/stereorize.c, main/channel.c,
+ funcs/func_enum.c, channels/chan_dahdi.c, main/manager.c,
+ channels/chan_skinny.c, main/ast_expr2f.c, res/res_agi.c,
+ pbx/ael/ael_lex.c, main/http.c, channels/chan_alsa.c,
+ pbx/ael/ael.flex, formats/format_gsm.c, apps/app_adsiprog.c,
+ formats/format_wav.c, apps/app_festival.c,
+ main/db1-ast/hash/hash_page.c, main/translate.c,
+ res/res_crypto.c, agi/eagi-test.c, formats/format_ogg_vorbis.c,
+ utils/astman.c, channels/chan_oss.c, agi/eagi-sphinx-test.c,
+ pbx/ael/ael.tab.c, main/file.c, pbx/ael/ael.tab.h,
+ apps/app_sms.c, pbx/pbx_dundi.c, res/res_indications.c,
+ utils/streamplayer.c, apps/app_chanspy.c, main/asterisk.c,
+ apps/app_voicemail.c, utils/muted.c, pbx/ael/ael.y,
+ apps/app_authenticate.c, formats/format_wav_gsm.c,
+ res/res_musiconhold.c, channels/chan_iax2.c: fix a bunch of
+ potential problems found by gcc 4.3.x, primarily bare strings
+ being passed to printf()-like functions and ignored results from
+ read()/write() and friends
+
+2008-10-31 22:36 +0000 [r153270] Terry Wilson <twilson@digium.com>
+
+ * res/res_features.c, apps/app_followme.c: Add end_bridge_callback
+ for app_follome and add AUTOLOOP flag to res_features
+
+2008-10-31 Tilghman Lesher <tlesher@digium.com>
+
+ * Asterisk 1.4.23-rc1 released.
+
+2008-10-31 16:30 +0000 [r153114] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Turn off qualify on uncached realtime peers.
+ (Closes issue #13383)
+
+2008-10-31 15:45 +0000 [r153095] Terry Wilson <twilson@digium.com>
+
+ * apps/app_dial.c, res/res_features.c, include/asterisk/channel.h:
+ Recent CDR fixes moved execution of the 'h' exten into the
+ bridging code, so variables that were set after ast_bridge_call
+ was called would not show up in the 'h' exten. Added a callback
+ function to handle setting variables, etc. from w/in the bridging
+ code. Calls back into a nested function within the function
+ calling ast_bridge_call (closes issue #13793) Reported by:
+ greenfieldtech
+
+2008-10-30 20:58 +0000 [r152992] Sean Bright <sean.bright@gmail.com>
+
+ * bootstrap.sh: The -I argument to aclocal needs a space before the
+ include directory name.
+
+2008-10-30 20:33 +0000 [r152922-152958] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_h323.c: Cannot join detached threads. See
+ http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html
+ (Closes issue #13400)
+
+ * channels/chan_local.c: Unlock before returning, when extension
+ doesn't exist. (closes issue #13807) Reported by: eliel Patches:
+ chan_local.c.patch uploaded by eliel (license 64)
+
+2008-10-30 16:53 +0000 [r152811] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/cdr.c: instead of comparing the string pointer to 0, let's
+ compare the value that was actually parsed out of the string
+ (found by sparse)
+
+2008-10-29 05:23 +0000 [r152539] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Fix an incorrect usage of sizeof() (closes
+ issue #13795) Reported by: andrew53 Patches:
+ chan_sip_sizeof.patch uploaded by andrew53 (license 519)
+
+2008-10-29 05:19 +0000 [r152535-152538] Steve Murphy <murf@digium.com>
+
+ * configs/features.conf.sample, apps/app_dial.c, apps/app_queue.c:
+ A little documentation cross-ref between features and dial and
+ queue... I wasted some time (stupidly) trying to get the
+ one-touch parking stuff working, because it didn't occur to me
+ that I had to also have the corresponding options in the dial
+ command! Duh! (In all this time, I never set this up before!) So,
+ to keep some poor fool from suffering the same fate, I made the
+ features.conf.sample file mention the corresponding opts in
+ dial/queue; and the docs for dial/app specifically mention the
+ corresponding decls in the feature.conf file. I hope this doesn't
+ spoil some vast, eternal plan...
+
+ * apps/app_dial.c, res/res_features.c, funcs/func_channel.c,
+ include/asterisk/pbx.h, apps/app_queue.c: The magic trick to
+ avoid this crash is not to try to find the channel by name in the
+ list, which is slow and resource consuming, but rather to pay
+ attention to the result codes from the ast_bridge_call, to which
+ I added the AST_PBX_NO_HANGUP_PEER_PARKED value, which now are
+ returned when a channel is parked. If you get AST_PBX_KEEPALIVE,
+ then don't touch the channel pointer. If you get
+ AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then
+ don't touch the peer pointer. Updated the several places where
+ the results from a bridge were not being properly obeyed, and
+ fixed some code I had introduced so that the results of the
+ bridge were not overridden (in trunk). All the places that
+ previously tested for AST_PBX_NO_HANGUP_PEER now have to check
+ for both AST_PBX_NO_HANGUP_PEER and
+ AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common
+ parking scenarios: 1. A calls B; B answers; A parks B; B hangs up
+ while A is getting the parking slot announcement, immediately
+ after being put on hold. 2. A calls B; B answers; A parks B; B
+ hangs up after A has been hung up, but before the park times out.
+ 3. A calls B; B answers; B parks A; A hangs up while B is getting
+ the parking slot announcement, immediately after being put on
+ hold. 4. A calls B; B answers; B parks A; A hangs up after B has
+ been hung up, but before the park times out. No crash. I also ran
+ the scenarios above against valgrind, and accesses looked good.
+
+2008-10-28 22:32 +0000 [r152368-152463] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Quoting in the wrong direction (Fixes
+ AST-107)
+
+ * apps/app_dial.c: Reset all DIAL variables back to blank, in case
+ Dial is called multiple times per call (which could otherwise
+ lead to inconsistent status reports). (closes issue #13216)
+ Reported by: ruddy Patches: 20081014__bug13216.diff.txt uploaded
+ by Corydon76 (license 14) Tested by: ruddy
+
+2008-10-27 23:28 +0000 [r152286] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Buffer policy setting for half is not
+ needed.
+
+2008-10-27 21:32 +0000 [r152215] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c: Inherit ALL elements of CallerID across a
+ local channel. (closes issue #13368) Reported by: Peter Schlaile
+ Patches: 20080826__bug13368.diff.txt uploaded by Corydon76
+ (license 14)
+
+2008-10-26 20:23 +0000 [r152059] Sean Bright <sean.bright@gmail.com>
+
+ * funcs/func_strings.c: Since passing \0 as the second argument to
+ strchr is valid (and will match the trailing \0 of a string) we
+ need to check that first, otherwise we end up with incorrect
+ results. Fix suggested by reporter. (closes issue #13787)
+ Reported by: meitinger
+
+2008-10-25 10:59 +0000 [r151905] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c: Move AMI initialization to occur after loading
+ modules. This prevents a deadlock when someone tries to initiate
+ a module reload from the AMI just as Asterisk is starting.
+ (closes issue #13778) Reported by: hotsblanc Fix suggested by
+ hotsblanc
+
+2008-10-23 16:04 +0000 [r151763] Terry Wilson <twilson@digium.com>
+
+ * configs/features.conf.sample, res/res_features.c, CHANGES:
+ Backport fix from 1.6.0 that allows you to set
+ parkedcalltransfers=no|caller|callee|both, but default to both
+ which would be the equivalent of the existing behaviour. The
+ problem was that if someone parked a call, the callee and caller
+ would both get assigned the builtin transfer feature, which would
+ not only be potentially giving someone the ability to transfer
+ themselves when they shouldn't have it, but would also dissallow
+ reinviting the media off of the call. (closes issue #12854)
+ Reported by: davidw Patches: parkingfix4.diff.txt uploaded by
+ otherwiseguy Tested by: davidw, otherwiseguy
+
+2008-10-20 04:57 +0000 [r151240-151241] Kevin P. Fleming <kpfleming@digium.com>
+
+ * autoconf/ast_check_pwlib.m4, autoconf/ast_check_openh323.m4,
+ configure.ac: rename this macro to properly reflect what it does
+
+ * autoconf/ast_check_pwlib.m4 (added), autoconf (added),
+ autoconf/acx_pthread.m4 (added), autoconf/ast_func_fork.m4
+ (added), configure, autoconf/ast_gcc_attribute.m4 (added),
+ bootstrap.sh, autoconf/ast_check_gnu_make.m4 (added),
+ autoconf/ast_ext_lib.m4 (added), autoconf/ast_prog_ld.m4 (added),
+ autoconf/ast_c_compile_check.m4 (added),
+ autoconf/ast_c_define_check.m4 (added),
+ autoconf/ast_prog_egrep.m4 (added),
+ autoconf/ast_check_openh323.m4 (added),
+ autoconf/ast_prog_ld_gnu.m4 (added), autoconf/ast_prog_sed.m4
+ (added), acinclude.m4 (removed): break up acinclude.m4 into
+ individual files, which will make it easier to maintain, easier
+ to add new macros (less patching) and will ease maintenance of
+ these macros across Asterisk branches
+
+2008-10-19 19:51 +0000 [r151100-151167] BJ Weschke <bweschke@btwtech.com>
+
+ * main/asterisk.c: As per kpfleming's comments to the prior commit,
+ I'm reverting some of the changes here. A comment was made in bug
+ #13726 "3. The same mistake as in (2) is done in a few other
+ places in the code that check for: #if defined(HAVE_ZAPTEL) ||
+ defined(HAVE_DAHDI) Harmless, but still incorrect." In the case
+ of main/asterisk.c, this is not incorrect because without
+ HAVE_ZAPTEL defined, we're missing the include for ioctl and the
+ namespace that defines DAHDI_TIMERCONFIG which is still required
+ when using Zaptel with the 1.4 branch.
+
+ * main/asterisk.c: Fix the 1.4 branch compile again broken with
+ r150557 when using with Zaptel and not DAHDI (closes issue
+ #13740) reported by: jmls patch by: bweschke
+
+2008-10-18 01:42 +0000 [r150816] BJ Weschke <bweschke@btwtech.com>
+
+ * main/manager.c: Using the GetVar handler in AMI is potentially
+ dangerous (insta-crash [tm]) when you use a dialplan function
+ that requires a channel and then you don't provide one or provide
+ an invalid one in the Channel: parameter. We'll handle this
+ situation exactly the same way it was handled in pbx.c back on
+ r61766. We'll create a bogus channel for the function call and
+ destroy it when we're done. If we have trouble allocating the
+ bogus channel then we're not going to try executing the function
+ call at all and run the risk of crashing. (closes issue #13715)
+ reported by: makoto patch by: bweschke
+
+2008-10-17 17:18 +0000 [r150637] Steve Murphy <murf@digium.com>
+
+ * res/res_features.c: Interesting crash. In this case, you exit the
+ bridge with peer completely GONE. I moved the channel find call
+ up to cover the whole peer CDR reset code segment. This appears
+ to solve the crash without changing the logic at all.
+
+2008-10-17 15:31 +0000 [r150557] Jason Parker <jparker@digium.com>
+
+ * main/asterisk.c, main/channel.c, channels/chan_dahdi.c,
+ configure, configure.ac: Correctly allow chan_dahdi to compile
+ against older versions of Zaptel. Don't always define
+ HAVE_ZAPTEL_CHANALARMS (since we check if it's defined..) Minor
+ cleanup to make things clear. (closes issue #13726) Reported by:
+ tzafrir Patches: dahdi_def.diff uploaded by tzafrir (license 46)
+
+2008-10-16 23:40 +0000 [r150298-150304] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c: Reverting changes from commits 150298 and 150301
+ since I was mistakenly under the assumption that dialplan
+ functions *always* required that a channel be present. I need to
+ go home earlier, I think :)
+
+ * main/manager.c: And don't forget to return on the error condition
+
+ * main/manager.c: Don't try to call a dialplan function's read
+ callback from the manager's GetVar handler if an invalid channel
+ has been specified. Several dialplan functions, including CHANNEL
+ and SIP_HEADER, do not check for NULL-ness of the channel being
+ passed in. (closes issue #13715) Reported by: makoto
+
+2008-10-16 15:56 +0000 [r150124] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: Fix memory leak found by customer
+
+2008-10-16 15:26 +0000 [r150056] Steve Murphy <murf@digium.com>
+
+ * cdr/cdr_odbc.c: This patch is relevant to: ABE-1628 and
+ RYM-150398 and AST-103 in internal Digium bug trackers. These
+ fixes address a really subtle memory corruption problem that
+ would happen in machines heavily loaded in production
+ environments. The corruption would always take the form of the
+ STMT object getting nulled out and one of the unixODBC calls
+ would crash trying to access statement->connection. It isn't
+ fully proven yet, but the server has now been running 2.5 days
+ without appreciable memory growth, or any gain of %cpu, and no
+ crashes. Whether this is the problem or not on that server, these
+ fixes are still warranted. As it turns out, **I** introduced
+ these errors unwittingly, when I corrected another crash earlier.
+ I had formed the build_query routine, and failed to remove
+ mutex_unlock calls in 3 places in the transplanted code. These
+ unlocks would only happen in error situations, but unlocking the
+ mutex early set the code up for a catastrophic failure, it
+ appears. It would happen only once every 100K-200K or more calls,
+ under heavy load... but that is enough. If another crash occurs,
+ with the same MO, I'll come back and remove my confession from
+ the log, and we'll keep searching, but the fact that we have
+ Asterisk dying from an asynchronous wiping of the STMT object,
+ only on some connection error, and that the server has lived for
+ 2.5 days on this code without a crash, sure make it look like
+ this was the problem! Also, in several points, Statement handles
+ are set to NULL after SQLFreeHandle. This was mainly for
+ insurance, to guarantee a crash. As it turns out, the code does
+ not appear to be attempting to use these freed pointers. Asterisk
+ owes a debt of gratitude to Federico Alves and Frediano Ziglio
+ for their untiring efforts in finding this bug, among others.
+
+2008-10-15 21:34 +0000 [r149683-149840] BJ Weschke <bweschke@btwtech.com>
+
+ * CHANGES: Another documentation fix. (closes issue #13708)
+
+ * configs/agents.conf.sample: An update to the
+ documentation/example of agents.conf.sample with the correct
+ parameter for this feature as defined in chan_agent.c (closes
+ issue #13709)
+
+2008-10-15 10:30 +0000 [r149452] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: fix some problems when parsing SIP messages
+ that have the maximum number of headers or body lines that we
+ support
+
+2008-10-14 23:43 +0000 [r149130-149266] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Change this warning to an error message.
+ Suggestion comes from Sean Bright. Thanks Sean!
+
+ * channels/chan_sip.c: Call register_peer_exten even in the case
+ that the peer's IP/port does not change. (closes issue #13309)
+ Reported by: dimas Patches: v2-13309.patch uploaded by dimas
+ (license 88)
+
+ * include/asterisk/audiohook.h, main/audiohook.c: Add a tolerance
+ period for sync-triggered audiohooks so that if packetization of
+ audio is close (but not equal) we don't end up flushing the
+ audiohooks over small inconsistencies in synchronization. Related
+ to issue #13005, and solves the issue for most people who were
+ experiencing the problem. However, a small number of people are
+ still experiencing the problem on long calls, so I am not closing
+ the issue yet
+
+ * apps/app_queue.c: Update the queue with the correct number of
+ calls and whether the call was completed within the service level
+ when a transfer takes place. This way, we do not "break" the
+ leastrecent and fewestcalls strategies by not logging a call
+ until after the transferred call has ended. (closes issue #13395)
+ Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded
+ by Marquis (license 32)
+
+ * channels/chan_sip.c: Don't allow reserved characters to be used
+ in register lines in sip.conf. (closes issue #13570) Reported by:
+ putnopvut
+
+2008-10-14 20:09 +0000 [r149061] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_waitforsilence.c: Check correct values in the return of
+ ast_waitfor(); also, get rid of a possible memory leak. (closes
+ issue #13658) Reported by: explidous Patch by: me
+
+2008-10-14 19:05 +0000 [r148990] Leif Madsen <lmadsen@digium.com>
+
+ * CHANGES: Add in some missing updates to the CHANGES file for
+ sip.conf (closes issue #13100) Reported and patch by:
+ gknispel_proformatique
+
+2008-10-14 19:03 +0000 [r148916-148987] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Some compilers warn, some don't. Fixing.
+
+ * apps/app_voicemail.c: Ensure that mail headers are 7-bit clean,
+ even when UTF-8 characters are used in headers like 'Subject' and
+ 'To'. Closes AST-107.
+
+2008-10-14 17:33 +0000 [r148912] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_local.c: Deadlock prevention in chan_local. (closes
+ issue #13676) Reported by: tacvbo Patches: 13676.patch uploaded
+ by putnopvut (license 60) Tested by: tacvbo
+
+2008-10-14 10:30 +0000 [r148611-148736] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile: on Ubuntu (at least), recent versions of ld in binutils
+ delete all debugging symbols when -x is supplied; since the
+ reasons why -x is being passed are lost in the mists of time,
+ remove it so debugging will work properly
+
+ * main/translate.c: it would be nice if this message printing code
+ had actually been tested before it was committed...
+
+2008-10-10 16:25 +0000 [r147997-148257] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: User not notified of temporary greeting, if
+ ODBC storage is in use. (closes issue #13659) Reported by:
+ moliveras Patches: 20081009__bug13659.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: moliveras
+
+ * apps/app_voicemail.c: When blank, callerid name and number should
+ display "unknown caller" in voicemail emails. (Closes issue
+ #13643)
+
+2008-10-09 18:56 +0000 [r147941] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_features.c: (closes issue #13139) Reported by: krisk84
+ Tested by: krisk84 This change prevents a call that is placed in
+ the parkinglot to be picked up before the PBX is finished. If
+ another extension dials the parking extension before the PBX
+ thread has completed at minimum warnings will occur about the PBX
+ not properly being terminated. At worst, a crash could occur.
+
+2008-10-08 22:22 +0000 [r147681] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: when parsing a text configuration option,
+ ensure that the buffer on the stack is actually large enough to
+ hold the legal values of that option, and also ensure that
+ sscanf() knows to stop parsing if it would overrun the buffer
+ (without these changes, specifying "buffers=...,immediate" would
+ overflow the buffer on the stack, and could not have worked as
+ expected)
+
+2008-10-08 14:51 +0000 [r147517] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_speech_utils.c: If we receive DTMF make sure that the
+ state of the speech structure goes back to being not ready.
+ (issue #LUMENVOX-8)
+
+2008-10-07 23:14 +0000 [r147429-147430] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: revert this change until i can understand
+ why it results in locking order changes
+
+ * channels/chan_dahdi.c: don't start a PBX on incoming PRI call
+ channels until after we're done setting channel variables and
+ other things on the channel, otherwise the channel might go away
+ (if the dialplan hangs up quickly) before we are done, which
+ results in a spectacular crash
+
+2008-10-07 16:48 +0000 [r147193] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_voicemail.c: Make 'imapsecret' an alias to
+ 'imappassword' in voicemail.conf.
+
+2008-10-06 20:52 +0000 [r146711-146799] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_callerid.c, apps/app_speech_utils.c,
+ funcs/func_curl.c, funcs/func_groupcount.c, res/res_smdi.c,
+ channels/chan_sip.c, funcs/func_timeout.c, funcs/func_odbc.c,
+ funcs/func_cdr.c, funcs/func_math.c, channels/chan_iax2.c:
+ Dialplan functions should not actually return 0, unless they have
+ modified the workspace. To signal an error (and no change to the
+ workspace), -1 should be returned instead. (closes issue #13340)
+ Reported by: kryptolus Patches: 20080827__bug13340__2.diff.txt
+ uploaded by Corydon76 (license 14)
+
+ * channels/chan_local.c: Check whether an extension exists in the
+ _call method, rather than the _alloc method, because we need to
+ evaluate the callerid (since that data affects whether an
+ extension exists). (closes issue #13343) Reported by: efutch
+ Patches: 20080915__bug13343.diff.txt uploaded by Corydon76
+ (license 14) Tested by: efutch
+
+2008-10-06 15:57 +0000 [r146643] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: ensure that the private structure for
+ pseudo channels is created without 'leaking' configuration data
+ from other configured channels (closes issue #13555) Reported by:
+ jeffg Patches: issue_13555.patch uploaded by kpfleming (license
+ 421) Tested by: jeffg
+
+2008-10-05 21:17 +0000 [r146448] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c: Fix silly formatting.
+
+2008-10-03 22:51 +0000 [r146244] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_rpt.c: Change some preprocessor macros to struct
+ definitions so that we get app_rpt to build with DAHDI. (closes
+ issue #13576) Reported by: blitzrage
+
+2008-10-03 20:44 +0000 [r146129] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/features.h, res/res_features.c, res/res_agi.c:
+ (closes issue #13425) Reported by: mdu113 Tested by: mdu113
+ Similar to r143204, masquerade the channel in the case of Park
+ being called from AGI.
+
+2008-10-03 17:12 +0000 [r146026] Steve Murphy <murf@digium.com>
+
+ * res/res_features.c: (closes issue #13579) Reported by: dwagner
+ (closes issue #13584) Reported by: dwagner Tested by: murf,
+ putnopvut The thought occurred to me that the res= from the
+ extension spawn was ending up being returned from the bridge.
+ "Thou shalt not poison the return value". Made the change and it
+ appears to allow blind xfers to work as normal. If I'm wrong,
+ reopen the bugs. But it looks good to me! Many thanks to
+ putnopvut for helping me reproduce this!
+
+2008-10-02 16:39 +0000 [r145751-145839] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_odbc.c: Backport support for some of the keyword
+ modifications used in 1.6 (while warning that some options aren't
+ really supported) and add some warning messages. Some credit to
+ oej, who was complaining in #asterisk-dev.
+
+ * res/res_odbc.c: Some sanity checks that may have led to prior
+ crashes, found by codefreeze-lap (murf) on IRC. Also some cleanup
+ of incorrectly-used constants.
+
+2008-10-01 17:18 +0000 [r145479] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/realtime_pgsql.sql: Update the realtime_pgsql.sql
+ script to create the setinterfacevar column. (closes issue
+ #13549) Reported by: fiddur
+
+2008-10-01 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.22 released.
+
+2008-09-09 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.22-rc5 released.
+
+2008-09-09 15:40 +0000 [r142063] Russell Bryant <russell@digium.com>
+
+ * res/res_features.c: Ensure that the stored CDR reference is still
+ valid after the bridge before poking at it. Also, keep the
+ channel locked while messing with this CDR. (fixes crashes
+ reported in issue #13409)
+
+2008-09-08 21:10 +0000 [r141809] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix pedantic mode of chan_sip to only check
+ the remote tag of an endpoint once a dialog has been confirmed.
+ Up until that point, it is possible and legal for the far-end to
+ send provisional responses with a different To: tag each time.
+ With this patch applied, these provisional messages will not
+ cause a matching problem. (closes issue #11536) Reported by: ibc
+ Patches: 11536v2.patch uploaded by putnopvut (license 60)
+
+2008-09-08 21:02 +0000 [r141806] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c: When doing an async goto, detect if the channel is
+ already in the middle of a masquerade. This can happen when
+ chan_local is trying to optimize itself out. If this happens,
+ fail the async goto instead of bursting into flames. (closes
+ issue #13435) Reported by: geoff2010
+
+2008-09-08 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.22-rc4 released.
+
+2008-09-08 20:15 +0000 [r141741] Jason Parker <jparker@digium.com>
+
+ * Makefile, redhat (removed): Remove RPM package targets from
+ Makefile (and all associated parts). This has never worked in
+ 1.4, and we decided that it makes no sense to be done here. There
+ are many distros out there that already have "proper" spec files
+ that can be (re)used. Closes issue #13113 Closes issue #10950
+ Closes issue #10952
+
+2008-09-08 16:26 +0000 [r141678] Russell Bryant <russell@digium.com>
+
+ * configure, configure.ac: Actually use Zaptel CFLAGS if using
+ Zaptel instead of DAHDI This fixes building against Zaptel when
+ using a custom path
+
+2008-09-06 20:13 +0000 [r141565] Steve Murphy <murf@digium.com>
+
+ * channels/chan_sip.c: This fix comes from Joshua Colp The
+ Brilliant, who, given the trace, came up with a solution. This
+ will most likely will close 13235 and 13409. I'll wait till
+ Monday to verify, and then close these bugs.
+
+2008-09-06 15:23 +0000 [r141503] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_agi.c: Reverting behavior change (AGI should not exit
+ non-zero on SUCCESS) (closes issue #13434) Reported by:
+ francesco_r
+
+2008-09-05 21:10 +0000 [r141217-141366] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_agent.c: Agent's should not try to call a channel's
+ indicate callback if the channel has been hung up. It will likely
+ crash otherwise ABE-1159
+
+ * apps/app_voicemail.c: Since greetings are not stored in IMAP, we
+ should not be DISPOSE'ing of them the same way we do with other
+ messages. (closes issue #13414) Reported by: mthomasslo Patches:
+ 13414v2.patch uploaded by putnopvut (license 60) Tested by:
+ mthomasslo
+
+ * channels/chan_sip.c: Commit 140417 had a logic flaw in it which
+ caused port 5060 to always be used when dialing a peer if no
+ explicit port was specified. This broke the behavior of
+ implicitly using the port from which the peer registered if no
+ port is specified. This commit fixes the logic flaw. (closes
+ issue #13424) Reported by: mdu113 Patches: 13424.patch uploaded
+ by putnopvut (license 60) Tested by: mdu113
+
+2008-09-05 14:15 +0000 [r141094-141156] Steve Murphy <murf@digium.com>
+
+ * main/channel.c: A small change to prevent double-posting of
+ CDR's; thanks to Daniel Ferrer for bringing it to our attention
+
+ * pbx/ael/ael-test/ref.ael-vtest25 (added),
+ pbx/ael/ael-test/ael-vtest25/extensions.ael (added),
+ pbx/ael/ael-test/ael-vtest25 (added), pbx/ael/ael_lex.c,
+ pbx/ael/ael-test/ref.ael-test6, pbx/ael/ael.flex: (closes issue
+ #13357) Reported by: pj Tested by: murf (closes issue #13416)
+ Reported by: yarns Tested by: murf If you find this message
+ overly verbose, relax, it's probably not meant for you. This
+ message is meant for probably only two people in the whole world:
+ me, or the poor schnook that has to maintain this code because
+ I'm either dead or unavailable at the moment. This fix solves two
+ reports, both having to do with embedding a function call in a
+ ${} construct. It was tricky because the funccall syntax has
+ parenthesis () in it. And up till now, the 'word' token in the
+ flex stuff didn't allow that, because it would tend to steal the
+ LP and RP tokens. To be truthful, the "word" token was the
+ trickiest, most unstable thing in the whole lexer. I was lucky it
+ made this long without complaints. I had to choose every
+ character in the pattern with extreme care, and I knew that
+ someday I'd have to revisit it. Well, the day has come. So, my
+ brilliant idea (and I'm being modest), was to use the surrounding
+ ${} construct to make a state machine and capture everything in
+ it, no matter what it contains. But, I have to now treat the word
+ token like I did with comments, in that I turn the whole thing
+ into a state-machine sort of spec, with new contexts
+ "curlystate", "wordstate", and "brackstate". Wait a minute,
+ "brackstate"? Yes, well, it didn't take very many regression
+ tests to point out if I do this for ${} constructs, I also have
+ to do it with the $[] constructs, too. I had to create a separate
+ pcbstack2 and pcbstack3 because these constructs can occur inside
+ macro argument lists, and when we have two state machines
+ operating on the same structures we'd get problems otherwise. I
+ guess I could have stopped at pcbstack2 and had the brackstate
+ stuff share it, but it doesn't hurt to be safe. So, the pcbpush
+ and pcbpop routines also now have versions for "2" and "3". I had
+ to add the {KEYWORD} construct to the initial pattern for "word",
+ because previously word would match stuff like "default7",
+ because it was a longer match than the keyword "default". But,
+ not any more, because the word pattern only matches only one or
+ two characters now, and it will always lose. So, I made it the
+ winner again by making an optional match on any of the keywords
+ before it's normal pattern. I added another regression test to
+ make sure we don't lose this in future edits, and had to fix just
+ one regression, where it no longer reports a 'cascaded' error,
+ which I guess is a plus. I've given some thought as to whether to
+ apply these fixes to 1.4 and the 1.6.x releases, vs trunk; I
+ decided to put it in 1.4 because one of the bug reports was
+ against 1.4; and it is unexpected that AEL cannot handle this
+ situation. It actually reduced the amount of useless "cascade"
+ error messages that appeared in the regressions (by one line,
+ ehhem). There is a possible side-effect in that it does now do
+ more careful checking of what's in those ${} constructs, as far
+ as matching parens, and brackets are concerned. Some users may
+ find a an insidious problem and correct it this way. This should
+ be exceedingly rare, I hope.
+
+2008-09-04 17:00 +0000 [r141028] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_features.c, res/res_agi.c: (closes issue #11979) Fixes
+ multiple parking problems: Crash when executing a park on an
+ extension dialed by AGI due to not returning the proper return
+ code. Crash when using a builtin feature that was a subset of a
+ enabled dynamic feature. Crash due to always hanging up the peer
+ despite the fact that the peer was supposed to be parked.
+
+2008-09-03 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.22-rc3 released.
+
+2008-09-03 14:29 +0000 [r140850] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Fix voicemail forwarding when using ODBC
+ storage. (closes issue #13387) Reported by: moliveras Patches:
+ 13387.patch uploaded by putnopvut (license 60) Tested by:
+ putnopvut, moliveras
+
+2008-09-03 13:24 +0000 [r140816] Russell Bryant <russell@digium.com>
+
+ * main/poll.c: Don't freak out if the poll emulation receives NULL
+ for the pollfds array (closes issue #13307) Reported by: jcovert
+
+2008-09-02 23:47 +0000 [r140751] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: After adding the context checking to
+ app_voicemail for IMAP storage, I left out a crucial place to
+ copy the context to the vm_state structure. This is the
+ correction.
+
+2008-09-02 23:36 +0000 [r140670-140747] Steve Murphy <murf@digium.com>
+
+ * main/cdr.c: I am turning the warnings generated in ast_cdr_free
+ and post_cdr into verbose level 2 messages. Really, they matter
+ little to end users. You either get the CDR's you wanted, or you
+ don't, and it is a bug.
+
+ * main/channel.c: After reconsidering, with respect to 13409,
+ ast_cdr_detach should be OK, better in fact, than ast_cdr_free,
+ which generates lots of useless warnings that will undoubtably
+ generate complaints.
+
+ * main/channel.c, main/pbx.c: (closes issue #13409) Reported by:
+ tomaso Patches: asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by
+ tomaso (license 564) I basically spent the day, verifying that
+ this patch solves the problem, and doesn't hurt in non-problem
+ cases. Why valgrind did not plainly reveal this leak absolutely
+ mystifies and stuns me. Many, many thanks to tomaso for finding
+ and providing the fix.
+
+2008-09-02 18:14 +0000 [r140605] Sean Bright <sean.bright@gmail.com>
+
+ * channels/chan_iax2.c: Make sure to use the correct length of the
+ mohinterpret and mohsuggest buffers when copying configuration
+ values. (closes issue #13336) Reported by:
+ decryptus_proformatique Patches:
+ chan_iax2_mohinterpret_mohsuggest_general_settings.patch uploaded
+ by decryptus (license 555)
+
+2008-08-29 17:34 +0000 [r140417-140488] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c, apps/app_queue.c, channels/chan_iax2.c: After
+ working on the ao2_containers branch, I noticed something a bit
+ strange. In all cases where we provide a callback function to
+ ao2_container_alloc, the callback function would only return 0 or
+ CMP_MATCH. After inspecting the ao2_callback() code carefully, I
+ found that if you're only looking for one specific item, then you
+ should return CMP_MATCH | CMP_STOP. Otherwise, astobj2 will
+ continue traversing the current bucket until the end searching
+ for more matches. In cases like chan_iax2 where in 1.4, all the
+ peers are shoved into a single bucket, this makes for potentially
+ terrible performance since the entire bucket will be traversed
+ even if the peer is one of the first ones come across in the
+ bucket. All the changes I have made were for cases where the
+ callback function defined was passed to ao2_container_alloc so
+ that calls to ao2_find could find a unique instance of whatever
+ object was being stored in the container.
+
+ * apps/app_voicemail.c: Add context checking when retrieving a
+ vm_state. This was causing a problem for people who had
+ identically named mailboxes in separate voicemail contexts. This
+ commit affects IMAP storage only. (closes issue #13194) Reported
+ by: moliveras Patches: 13194.patch uploaded by putnopvut (license
+ 60) Tested by: putnopvut, moliveras
+
+ * channels/chan_sip.c: Fix SIP's parsing so that if a port is
+ specified in a string to Dial(), it is not ignored. (closes issue
+ #13355) Reported by: acunningham Patches: 13355v2.patch uploaded
+ by putnopvut (license 60) Tested by: acunningham
+
+2008-08-27 19:49 +0000 [r140299] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix tag checking in get_sip_pvt_byid_locked
+ when in pedantic mode. The problem was that the wrong tags would
+ be compared depending on the direction of the call. (closes issue
+ #13353) Reported by: flefoll Patches:
+ chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll
+ (license 244)
+
+2008-08-26 16:49 +0000 [r140115] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: add HAVE_PRI if define around
+ dahdi_close_pri_fd
+
+2008-08-26 16:07 +0000 [r140060] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Fix some bogus scheduler usage in chan_sip.
+ This code used the return value of a completely unrelated
+ function to determine whether the scheduler should be run or not.
+ This would have caused the scheduler to not run in cases where it
+ should have. Also, leave a note about another scheduler issue
+ that needs to be addressed at some point.
+
+2008-08-26 15:57 +0000 [r140056] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: (closes issue #12071) Reported by: tzafrir
+ Patches: dahdi_close.diff uploaded by tzafrir (license 46) Tested
+ by: tzafrir, jpeeler This patch fixes closing open file
+ descriptors in the case of an error.
+
+2008-08-26 15:27 +0000 [r140051] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Fix a race condition with the IAX scheduler
+ thread. A lock and condition are used here to allow newly
+ scheduled tasks to wake up the scheduler just in case the new
+ task needs to run sooner than the current wakeup time when the
+ thread is sleeping. However, there was a race condition such that
+ a newly scheduled task would not properly wake up the scheduler
+ or affect the wake up period. The order of execution would have
+ been: 1) Scheduler thread determines wake up time of N ms. 2)
+ Another thread schedules a task and signals the condition, with
+ an execution time of < N ms. 3) Scheduler thread locks and goes
+ to sleep for N ms. By moving the sleep time determination to
+ inside the critical section, this possibility is avoided.
+
+2008-08-26 15:22 +0000 [r140050] Terry Wilson <twilson@digium.com>
+
+ * Makefile: sounds/Makefile installs sounds using the "new"
+ language directory structure, but languageprefix needs to be set
+ = yes for sounds in subdirectories (digits/1, etc.) to play as
+ the correct language. Fix the generation of asterisk.conf to
+ include languageprefix=yes
+
+2008-08-26 14:09 +0000 [r140029] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: correct a file location in an error
+ message
+
+2008-08-25 21:47 +0000 [r139927] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c: Fix a typo I made. Lesson learned, apply the
+ patch if one exists.
+
+2008-08-25 21:31 +0000 [r139909] Sean Bright <sean.bright@gmail.com>
+
+ * build_tools/get_moduleinfo, build_tools/get_makeopts: Some
+ versions of awk (nawk, for example) don't like empty regular
+ expressions so be slightly more verbose. (closes issue #13374)
+ Reported by: dougm Patches: 13374.diff uploaded by seanbright
+ (license 71) Tested by: dougm
+
+2008-08-25 20:46 +0000 [r139869] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Make SIPADDHEADER() propagate indefinitely
+
+2008-08-25 15:52 +0000 [r139769] Mark Michelson <mmichelson@digium.com>
+
+ * main/config.c: Fix the logic in config_text_file_save so that if
+ an UpdateConfig manager action is issued and the file specified
+ in DstFileName does not yet exist, an error is not returned.
+ (closes issue #13341) Reported by: vadim Patches: 13341.patch
+ uploaded by putnopvut (license 60) (with small modification from
+ seanbright)
+
+2008-08-25 15:33 +0000 [r139764] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c, res/res_features.c: This patch reverts the changes
+ made via 139347, and 139635, as users are seeing adverse
+ difference. I will un-close 13251. Back to the drawing board/
+ concept/ beginning/ whatever!
+
+2008-08-22 22:24 +0000 [r139635] Steve Murphy <murf@digium.com>
+
+ * res/res_features.c: I found some problems with the code I
+ committed earlier, when I merged them into trunk, so I'm coming
+ back to clean up. And, in the process, I found an error in the
+ code I added to trunk and 1.6.x, that I'll fix using this patch
+ also.
+
+2008-08-22 21:36 +0000 [r139621] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c: (closes issue #13359) Reported by: Laureano
+ Patches: originate_channel_check.patch uploaded by Laureano
+ (license 265)
+
+2008-08-22 19:45 +0000 [r139456-139553] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/threadstorage.h: Fix compilation when
+ DEBUG_THREAD_LOCALS is selected (closes issue #13298) Reported
+ by: snuffy Patches: bug13298_20080822.diff uploaded by snuffy
+ (license 35)
+
+ * main/frame.c: Remove show_frame_stats_deprecated since it is not
+ used anywhere and causes build errors if building under dev-mode
+ with TRACE_FRAMES selected in menuselect. (closes issue #13362)
+ Reported by: snuffy
+
+ * channels/chan_iax2.c: Fix the build. Thanks, mvanbaak!
+
+ * channels/chan_iax2.c: Prevent a deadlock in chan_iax2 resulting
+ from incorrect locking order between iax2_pvt and ast_channel
+ structures. AST-13
+
+2008-08-21 23:39 +0000 [r139387] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Fixes loop that could possibly never exit
+ in the event of a channel never being able to be opened or
+ specify after a restart. (closes issue #11017)
+
+2008-08-21 23:03 +0000 [r139347] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c, res/res_features.c: (closes issue #13251) Reported
+ by: sergee Tested by: murf THis is a bold move for a static
+ release fix, but I wouldn't have made it if I didn't feel
+ confident (at least a *bit* confident) that it wouldn't mess
+ everyone up. The reasoning goes something like this: 1. We simply
+ cannot do anything with CDR's at the current point (in pbx.c,
+ after the __ast_pbx_run loop). It's way too late to have any
+ affect on the CDRs. The CDR is already posted and gone, and the
+ remnants have been cleared. 2. I was very much afraid that moving
+ the running of the 'h' extension down into the bridge code (where
+ it would be now practical to do it), would result in a lot more
+ calls to the 'h' exten, so I implemented it as another exten
+ under another name, but found, to my pleasant surprise, that
+ there was a 1:1 correspondence to the running of the 'h' exten in
+ the pbx_run loop, and the new spot at the end of the bridge. So,
+ I ifdef'd out the current 'h' loop, and moved it into the bridge
+ code. The only difference I can see is the stuff about the
+ AST_PBX_KEEPALIVE, and hopefully, if this is still an important
+ decision point, I can replicate it if there are complaints. To be
+ perfectly honest, the KEEPALIVE situation is not totally clear to
+ me, and how it relates to a post-bridge situation is less clear.
+ I suspect the users will point out everything in total clarity if
+ this steps on anyone's toes! 3. I temporarily swap the bridge_cdr
+ into the channel before running the 'h' exten, which makes it
+ possible for users to edit the cdr before it goes out the door.
+ And, of course, with the endbeforehexten config var set, the
+ users can also get at the billsec/duration vals. After the h
+ exten finishes, the cdr is swapped back and processing continues
+ as normal. Please, all who deal with CDR's, please test this
+ version of Asterisk, and file bug reports as appropriate!
+
+2008-08-21 10:11 +0000 [r139283] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * channels/chan_gtalk.c: Apply fix for issue #13310 to branch 1.4,
+ too.
+
+2008-08-20 22:14 +0000 [r139213] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c: Fix a crash in the ChanSpy application. The
+ issue here is that if you call ChanSpy and specify a spy group,
+ and sit in the application long enough looping through the
+ channel list, you will eventually run out of stack space and the
+ application with exit with a seg fault. The backtrace was always
+ inside of a harmless snprintf() call, so it was tricky to track
+ down. However, it turned out that the call to snprintf() was just
+ the biggest stack consumer in this code path, so it would always
+ be the first one to hit the boundary. (closes issue #13338)
+ Reported by: ruddy
+
+2008-08-20 19:52 +0000 [r139151] Shaun Ruffell <sruffell@digium.com>
+
+ * codecs/codec_dahdi.c: Fix bug where the samples were not accurate
+ when in G723 mode, which would cause the timestamp field of the
+ RTP header to be invalid.
+
+2008-08-20 19:35 +0000 [r139145] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Backport support
+ for Zaptel/DAHDI channel-level alarms from trunk/1.6, because not
+ doing so just makes it difficult for people with channels that
+ are in alarm when Asterisk starts up to get them going once the
+ alarm is cleared (closes issue #12160) Reported by: tzafrir
+ Patches: asterisk-chanalarms_14.patch uploaded by tzafrir
+ (license 46) Tested by: tzafrir
+
+2008-08-20 17:14 +0000 [r139074] Steve Murphy <murf@digium.com>
+
+ * main/cdr.c: (closes issue #13263) Reported by: brainy Tested by:
+ murf The specialized reset routine is tromping on the flags field
+ of the CDR. I made a change to not reset the DISABLED bit. This
+ should get rid of this problem.
+
+2008-08-20 15:37 +0000 [r139015] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: sip_read should properly handle a NULL
+ return from sip_rtp_read. (closes issue #13257) Reported by:
+ travishein
+
+2008-08-19 23:22 +0000 [r138949] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/dahdi_compat.h: add DAHDI_POLICY_WHEN_FULL
+ compatability define for Zaptel
+
+2008-08-19 23:17 +0000 [r138942] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_agent.c: Reset agent_pvt variables back to the
+ values in agents.conf (from what the corresponding channel
+ variables were set to) when the agent logs out. (closes issue
+ #13098) Reported by: davidw Patches:
+ 20080731__issue13098_agent_ackcall_not_reset.diff uploaded by
+ bbryant (license 36) Tested by: davidw
+
+2008-08-19 22:56 +0000 [r138938] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Add configuration option to
+ chan_dahdi.conf to allow buffering policy and number of buffers
+ to be configured per channel. Syntax: buffers=<num of
+ buffers>,<policy> Where the number of buffers is some
+ non-negative integer and the policy is either "full", "half", or
+ "immediate".
+
+2008-08-19 18:50 +0000 [r138685-138886] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c: Add a lock and unlock prior to the
+ destruction of the chanspy_ds lock to ensure that no other
+ threads still have it locked. While this should not happen under
+ normal circumstances, it appears that if the spyer and spyee hang
+ up at nearly the same time, the following may occur. 1.
+ ast_channel_free is called on the spyee's channel. 2. The chanspy
+ datastore is removed from the spyee's channel in
+ ast_channel_free. 3. In the spyer's thread, the spyer attempts to
+ remove and destroy the datastore from the spyee channel, but the
+ datastore has already been removed in step 2, so the spyer
+ continues in the code. 4. The spyee's thread continues and calls
+ the datastore's destroy callback, chanspy_ds_destroy. This
+ involves locking the chanspy_ds. 5. Now the spyer attempts to
+ destroy the chanspy_ds lock. The problem is that in step 4, the
+ spyee has locked this lock, meaning that the spyer is attempting
+ to destroy a lock which is currently locked by another thread.
+ The backtrace provided in issue #12969 supports the idea that
+ this is possible (and has even occurred). This commit does not
+ close the issue, but should help in preventing one type of crash
+ associated with the use of app_chanspy.
+
+ * apps/app_queue.c: Change the inequalities used in app_queue with
+ regards to timeouts from being strict to non-strict for more
+ accuracy. (closes issue #13239) Reported by: atis Patches:
+ app_queue_timeouts_v2.patch uploaded by atis (license 242)
+
+2008-08-18 16:57 +0000 [r138663] Kevin P. Fleming <kpfleming@digium.com>
+
+ * codecs/codec_dahdi.c: look for transcoder in proper place based
+ on build against Zaptel or DAHDI
+
+2008-08-18 11:57 +0000 [r138569] Sean Bright <sean.bright@gmail.com>
+
+ * channels/chan_dahdi.c: You know what's awesome? Code that
+ compiles... ;)
+
+2008-08-18 02:05 +0000 [r138516] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: fix compilation warnings
+
+2008-08-16 01:12 +0000 [r138309-138360] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: fixes use count to properly decrement if
+ an active dahdi channel is destroyed allowing module to be
+ unloaded
+
+ * channels/chan_dahdi.c: add forgotten locks around ss_thread_count
+ in ss_thread for dahdi restart
+
+2008-08-15 22:33 +0000 [r138258] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: More fixes for
+ realtime peers. (closes issue #12921) Reported by: Nuitari
+ Patches: 20080804__bug12921.diff.txt uploaded by Corydon76
+ (license 14) 20080815__bug12921.diff.txt uploaded by Corydon76
+ (license 14) Tested by: Corydon76
+
+2008-08-15 21:28 +0000 [r138119-138238] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: initialize condition variable
+ ss_thread_complete using ast_cond_init
+
+ * channels/chan_dahdi.c: declared static mutexes using
+ AST_MUTEX_DEFINE_STATIC macro
+
+ * channels/chan_dahdi.c: Fixes the dahdi restart functionality.
+ Dahdi restart allows one to restart all DAHDI channels, even if
+ they are currently in use. This is different from unloading and
+ then loading the module since unloading requires the use count to
+ be zero. Reloading the module is different in that the signalling
+ is not changed from what it was originally configured. Also, this
+ fixes not closing all the file descriptors for D-channels upon
+ module unload (which would prevent loading the module
+ afterwards). (closes issue #11017)
+
+2008-08-15 15:07 +0000 [r138027] Russell Bryant <russell@digium.com>
+
+ * main/autoservice.c: Ensure that when a hangup occurs in
+ autoservice, that a hangup frame gets properly deferred to be
+ read from the channel owner when it gets taken out of
+ autoservice. (closes issue #12874) Reported by: dimas Patches:
+ v1-12874.patch uploaded by dimas (license 88)
+
+2008-08-15 14:51 +0000 [r137847-138023] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_strings.c: Additional check for more string specifiers
+ than arguments. (closes issue #13299) Reported by: adomjan
+ Patches: 20080813__bug13299.diff.txt uploaded by Corydon76
+ (license 14) func_strings.c-sprintf.patch uploaded by adomjan
+ (license 487) Tested by: adomjan
+
+ * channels/chan_dahdi.c: Oops, wrong direction
+
+ * channels/chan_dahdi.c: When creating the secondary subchannel
+ name, it is necessary to compare to the existing channel name
+ without the "Zap/" or "DAHDI/" prefix, since our test string is
+ also without that prefix. (closes issue #13027) Reported by:
+ dferrer Patches: chan_zap-1.4.21.1_fix2.patch uploaded by dferrer
+ (license 525) (Slightly modified by me, to compensate for both
+ names)
+
+2008-08-14 14:05 +0000 [r137731] Russell Bryant <russell@digium.com>
+
+ * configs/sip.conf.sample: Comments in this config file were
+ aligned only if your tab size was set to 8. So, convert tabs to
+ spaces so that things should be aligned regardless of what tab
+ size you use in your editor.
+
+2008-08-14 02:03 +0000 [r137677-137679] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Zaptel-to-DAHDI.txt: forgot one module name that changed
+
+ * include/asterisk/dahdi_compat.h, channels/chan_dahdi.c,
+ build_tools/menuselect-deps.in, configure, configure.ac,
+ codecs/codec_dahdi.c: add support for Zaptel versions that
+ contain the new transcoder interface
+
+2008-08-13 21:35 +0000 [r137580] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Register DAHDISendKeypadFacility
+ application if dahdi_chan_mode is set to DAHDI + Zap. Mark
+ ZapSendKeypadFacility application as deprecated on usage.
+
+2008-08-13 20:46 +0000 [r137527-137530] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Zaptel-to-DAHDI.txt (added): add document describing what users
+ will need to be aware of when upgrading to this version and using
+ DAHDI
+
+ * apps/app_meetme.c: remove some more chan_zap references
+
+ * doc/asterisk-conf.txt, channels/chan_dahdi.c: document
+ dahdichanname option in doc/asterisk-conf.txt make chan_dahdi
+ read its configuration from zapata.conf if dahdichanname has been
+ set to 'no'
+
+2008-08-13 14:33 +0000 [r137348-137405] Sean Bright <sean.bright@gmail.com>
+
+ * doc/cdrdriver.txt: Update docs to reflect the change to cdr_tds
+
+ * cdr/cdr_tds.c: Bring cdr_tds in line with the other CDR backends
+ and have it try to store CDR(userfield) if it is set. The new
+ behavior is to check for the userfield column on module load, and
+ if it exists, we will store CDR(userfield) when CDRs are written.
+ A similar patch already went into trunk and 1.6.0. (closes issue
+ #13290) Reported by: falves11
+
+2008-08-11 13:33 +0000 [r137188] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_meetme.c: convert this module to be able to handle DAHDI
+ or Zaptel (reported on asterisk-users, don't know how this got
+ missed before)
+
+2008-08-11 00:20 +0000 [r137138] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_odbc.c: Deallocate database connection handle on
+ disconnect, as we allocate another one on connect. (closes issue
+ #13271) Reported by: dveiga
+
+2008-08-09 17:11 +0000 [r136999] Russell Bryant <russell@digium.com>
+
+ * configure, configure.ac: Ensure PBX_DAHDI_TRANSCODE will evaluate
+ to 0 if not found instead of empty. pointed out by tzafrir on
+ #asterisk-dev
+
+2008-08-09 15:25 +0000 [r136946] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/compat.h, include/asterisk/astobj2.h: Merged
+ revisions 136945 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r136945 | tilghman | 2008-08-09 10:24:36 -0500 (Sat, 09 Aug 2008)
+ | 2 lines Regression fixes for Solaris ........
+
+2008-08-08 00:15 +0000 [r136726] Steve Murphy <murf@digium.com>
+
+ * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
+ pbx/ael/ael-test/ref.ael-vtest13,
+ pbx/ael/ael-test/ref.ael-ntest10, pbx/pbx_ael.c,
+ include/asterisk/ael_structs.h: (closes issue #13236) Reported
+ by: korihor Wow, this one was a challenge! I regrouped and ran a
+ new strategy for setting the ~~MACRO~~ value; I set it once per
+ extension, up near the top. It is only set if there is a switch
+ in the extension. So, I had to put in a chunk of code to detect a
+ switch in the pval tree. I moved the code to insert the set of
+ ~~exten~~ up to the beginning of the gen_prios routine, instead
+ of down in the switch code. I learned that I have to push the
+ detection of the switches down into the code, so everywhere I
+ create a new exten in gen_prios, I make sure to pass onto it the
+ values of the mother_exten first, and the exten next. I had to
+ add a couple fields to the exten struct to accomplish this, in
+ the ael_structs.h file. The checked field makes it so we don't
+ repeat the switch search if it's been done. I also updated the
+ regressions.
+
+2008-08-07 18:25 +0000 [r136560] Kevin P. Fleming <kpfleming@digium.com>
+
+ * build_tools/menuselect-deps.in, configure, configure.ac: change
+ the required dependency for codec_dahdi to only be satisfied by
+ DAHDI and not Zaptel, as the new transcoder interface is only in
+ DAHDI
+
+2008-08-07 18:14 +0000 [r136544] Shaun Ruffell <sruffell@digium.com>
+
+ * codecs/codec_dahdi.c: Updated codec_dahdi to use the new
+ transcoder interface in the first DAHDI release. Codec dahdi no
+ longer functions with the transcoder interface in zaptel at this
+ time (which the last zaptel release was 1.4.11). NOTE: Still
+ needs an update to the configure script to make sure that
+ codec_dahdi is only built if the new transcoder interface is
+ present in the drivers. (Issue: DAHDI-42)
+
+2008-08-07 16:50 +0000 [r136488] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c: Update persistent state on all exit conditions.
+ (closes issue #12916) Reported by: sgenyuk Patches:
+ app_queue.patch.txt uploaded by neutrino88 (license 297) Tested
+ by: sgenyuk, aragon
+
+2008-08-07 16:30 +0000 [r136404-136484] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/doxyref.h: add a raw list of all libraries that
+ any part of Asterisk links directly to
+
+ * apps/app_voicemail.c: work around a bug in gcc-4.2.3 that
+ incorrectly ignores the casting away of 'const' for pointers when
+ the developer knows it is safe to do so
+
+ * Makefile: remove config.cache during distclean, in case the user
+ is using autoconf caching
+
+2008-08-07 01:31 +0000 [r136304-136348] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c: Also, parse
+ useincomingcalleridonzaptransfer (and add appropriate deprecation
+ warnings).
+
+ * channels/chan_dahdi.c: For backwards compatibility with previous
+ 1.4 versions which used "zapchan" in users.conf, ensure that we
+ still support it.
+
+2008-08-06 21:18 +0000 [r136241] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn_config.c, channels/chan_misdn.c,
+ configs/misdn.conf.sample: * The allowed_bearers setting in
+ misdn.conf misspelled one of its options: digital_restricted. *
+ Fixed some other spelling errors and typos.
+
+2008-08-06 20:42 +0000 [r136238] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: We only need to unregister the QueueStatus
+ manager command once on an unload
+
+2008-08-06 20:14 +0000 [r136190] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/rc.redhat.asterisk: -C option takes a filename,
+ not a directory path. (closes issue #13007) Reported by:
+ klaus3000
+
+2008-08-06 18:58 +0000 [r136168] Russell Bryant <russell@digium.com>
+
+ * Makefile: Remove the use of --no-print-directory when compiling
+ subdirectories. This allows vim :make functionality to work
+ properly when errors have occurred in the build. Without printing
+ the directories, vim did not know how to find the file that the
+ error occurred in. If the extra bit of build noise annoys anyone,
+ just let me know, and I'll make this optional.
+
+2008-08-06 15:58 +0000 [r136062] Mark Michelson <mmichelson@digium.com>
+
+ * main/rtp.c, channels/chan_skinny.c: Since adding the
+ AST_CONTROL_SRCUPDATE frame type, there are places where
+ ast_rtp_new_source may be called where the tech_pvt of a channel
+ may not yet have an rtp structure allocated. This caused a crash
+ in chan_skinny, which was fixed earlier, but now the same crash
+ has been reported against chan_h323 as well. It seems that the
+ best solution is to modify ast_rtp_new_source to not attempt to
+ set the marker bit if the rtp structure passed in is NULL. This
+ change to ast_rtp_new_source also allows the removal of what is
+ now a redundant pointer check from chan_skinny. (closes issue
+ #13247) Reported by: pj
+
+2008-08-06 03:53 +0000 [r135899-135949] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c: Fix a longstanding bug in channel walking logic,
+ and fix the explanation to make sense. (Closes issue #13124)
+
+ * main/translate.c: Since powerof() can return an error condition,
+ it's foolhardy not to detect and deal with that condition.
+ (Related to issue #13240)
+
+ * include/asterisk/threadstorage.h, include/asterisk/utils.h: 1)
+ Bugfix for debugging code 2) Reduce compiler warnings for another
+ section of debugging code (Closes issue #13237)
+
+2008-08-06 00:29 +0000 [r135841-135850] Mark Michelson <mmichelson@digium.com>
+
+ * /: Remove properties that should not be here
+
+ * apps/app_skel.c: Revert inadvertent changes to app_skel that
+ occurred when I was testing for a memory leak
+
+ * include/asterisk/abstract_jb.h, main/channel.c, /,
+ apps/app_skel.c, main/abstract_jb.c, main/fixedjitterbuf.h:
+ Merging the issue11259 branch. The purpose of this branch was to
+ take into account "burps" which could cause jitterbuffers to
+ misbehave. One such example is if the L option to Dial() were
+ used to inject audio into a bridged conversation at regular
+ intervals. Since the audio here was not passed through the
+ jitterbuffer, it would cause a gap in the jitterbuffer's
+ timestamps which would cause a frames to be dropped for a brief
+ period. Now ast_generic_bridge will empty and reset the
+ jitterbuffer each time it is called. This causes injected audio
+ to be handled properly. ast_generic_bridge also will empty and
+ reset the jitterbuffer if it receives an AST_CONTROL_SRCUPDATE
+ frame since the change in audio source could negatively affect
+ the jitterbuffer. All of this was made possible by adding a new
+ public API call to the abstract_jb called ast_jb_empty_and_reset.
+ (closes issue #11259) Reported by: plack Tested by: putnopvut
+
+2008-08-05 23:13 +0000 [r135799] Steve Murphy <murf@digium.com>
+
+ * apps/app_dial.c, main/cdr.c, main/channel.c, res/res_features.c,
+ include/asterisk/cdr.h: (closes issue #12982) Reported by: bcnit
+ Tested by: murf I discovered that also, in the previous bug fixes
+ and changes, the cdr.conf 'unanswered' option is not being
+ obeyed, so I fixed this. And, yes, there are two 'answer' times
+ involved in this scenario, and I would agree with you, that the
+ first answer time is the time that should appear in the CDR. (the
+ second 'answer' time is the time that the bridge was begun). I
+ made the necessary adjustments, recording the first answer time
+ into the peer cdr, and then using that to override the bridge
+ cdr's value. To get the 'unanswered' CDRs to appear, I purposely
+ output them, using the dial cmd to mark them as DIALED (with a
+ new flag), and outputting them if they bear that flag, and you
+ are in the right mode. I also corrected one small mention of the
+ Zap device to equally consider the dahdi device. I heavily tested
+ 10-sec-wait macros in dial, and without the macro call; I tested
+ hangups while the macro was running vs. letting the macro
+ complete and the bridge form. Looks OK. Removed all the
+ instrumentation and debug.
+
+2008-08-05 21:34 +0000 [r135747] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: In a conversion to use ast_strlen_zero, the
+ meaning of the flag IAX_HASCALLERID was perverted. This change
+ reverts IAX2 to the original meaning, which was, that the
+ callerid set on the client should be overridden on the server,
+ even if that means the resulting callerid is blank. In other
+ words, if you set "callerid=" in the IAX config, then the
+ callerid should be overridden to blank, even if set on the
+ client. Note that there's a distinction, even on realtime,
+ between the field not existing (NULL in databases) and the field
+ existing, but set to blank (override callerid to blank).
+
+2008-08-05 13:25 +0000 [r135597] Sean Bright <sean.bright@gmail.com>
+
+ * main/cli.c: Use PATH_MAX for filenames
+
+2008-08-04 20:15 +0000 [r135536] Russell Bryant <russell@digium.com>
+
+ * configs/chan_dahdi.conf.sample: fix a config sample typo
+
+2008-08-04 17:07 +0000 [r135479-135482] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/rc.mandrake.asterisk: Define ASTSBINDIR for script
+
+ * apps/app_voicemail.c: Memory leak on unload (closes issue #13231)
+ Reported by: eliel Patches: app_voicemail.leak.patch uploaded by
+ eliel (license 64)
+
+2008-08-04 16:26 +0000 [r135473] Russell Bryant <russell@digium.com>
+
+ * configs/chan_dahdi.conf.sample: Add a minor clarification to the
+ documentation of mohinterpret and mohsuggest
+
+2008-08-01 11:43 +0000 [r135055-135058] Michiel van Baak <michiel@vanbaak.info>
+
+ * apps/app_ices.c: make app_ices compile on OpenBSD.
+
+ * channels/chan_skinny.c: fix some potential deadlocks in
+ chan_skinny (closes issue #13215) Reported by: qwell Patches:
+ 2008080100_bug13215.diff.txt uploaded by mvanbaak (license 7)
+ Tested by: mvanbaak
+
+2008-07-31 22:18 +0000 [r134983] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/http.c: accomodate users who seem to lack a sense of humor
+ :-)
+
+2008-07-31 21:53 +0000 [r134976] Tilghman Lesher <tlesher@digium.com>
+
+ * sample.call, main/manager.c, pbx/pbx_spool.c: Specify codecs in
+ callfiles and manager, to allow video calls to be set up from
+ callfiles and AMI. (closes issue #9531) Reported by: Geisj
+ Patches: 20080715__bug9531__1.4.diff.txt uploaded by Corydon76
+ (license 14) 20080715__bug9531__1.6.0.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: Corydon76
+
+2008-07-31 19:37 +0000 [r134915] Russell Bryant <russell@digium.com>
+
+ * apps/app_ices.c: Get app_ices working again (closes issue #12981)
+ Reported by: dlogan Patches:
+ 20080709__app_ices_v2_update_trunk.diff uploaded by bbryant
+ (license 36) 20080709__app_ices_v2_update_14.diff uploaded by
+ bbryant (license 36) Tested by: bbryant
+
+2008-07-31 19:23 +0000 [r134883] Steve Murphy <murf@digium.com>
+
+ * res/res_features.c: (closes issue #11849) Reported by: greyvoip
+ Tested by: murf OK, a few days of debugging, a bunch of
+ instrumentation in chan_sip, main/channel.c, main/pbx.c, etc. and
+ 5 solid notebook pages of notes later, I have made the small
+ tweek necc. to get the start time right on the second CDR when: A
+ Calls B B answ. A hits Xfer button on sip phone, A dials C and
+ hits the OK button, A hangs up C answers ringing phone B and C
+ converse B and/or C hangs up But does not harm the scenario
+ where: A Calls B B answ. B hits xfer button on sip phone, B dials
+ C and hits the OK button, B hangs up C answers ringing phone A
+ and C converse A and/or C hangs up The difference in start times
+ on the second CDR is because of a Masquerade on the B channel
+ when the xfer number is sent. It ends up replacing the CDR on the
+ B channel with a duplicate, which ends up getting tossed out. We
+ keep a pointer to the first CDR, and update *that* after the
+ bridge closes. But, only if the CDR has changed. I hope this
+ change is specific enough not to muck up any current CDR-based
+ apps. In my defence, I assert that the previous information was
+ wrong, and this change fixes it, and possibly other similar
+ scenarios. I wonder if I should be doing the same thing for the
+ channel, as I did for the peer, but I can't think of a scenario
+ this might affect. I leave it, then, as an exersize for the
+ users, to find the scenario where the chan's CDR changes and
+ loses the proper start time.
+
+2008-07-31 16:45 +0000 [r134814] Russell Bryant <russell@digium.com>
+
+ * channels/iax2-parser.c: In case we have some processing threads
+ that free more frames than they allocate, do not let the frame
+ cache grow forever. (closes issue #13160) Reported by: tavius
+ Tested by: tavius, russell
+
+2008-07-31 15:56 +0000 [r134758] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Add more timeout checks into app_queue,
+ specifically targeting areas where an unknown and potentially
+ long time has just elapsed. Also added a check to try_calling()
+ to return early if the timeout has elapsed instead of potentially
+ setting a negative timeout for the call (thus making it have *no*
+ timeout at all). (closes issue #13186) Reported by:
+ miquel_cabrespina Patches: 13186.diff uploaded by putnopvut
+ (license 60) Tested by: miquel_cabrespina
+
+2008-07-30 22:39 +0000 [r134704] Tilghman Lesher <tlesher@digium.com>
+
+ * main/sched.c, include/asterisk/sched.h: Oops, wrong define
+
+2008-07-30 22:02 +0000 [r134652] Steve Murphy <murf@digium.com>
+
+ * pbx/pbx_ael.c: (closes issue #13197) Reported by: pj (closes
+ issue #13051) Reported by: pj This patch substitutes commas in
+ the expr supplied to the if () statement, as in if ( expr ) ...
+ This solves both the bugs above, and makes the source symmetric
+ with switch statements, which were earlier reported to need this
+ sort of treatment. I tested this using the examples, both for the
+ compiler and at run time. Looks good.
+
+2008-07-30 21:38 +0000 [r134649] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, configure.ac: Qwell pointed out, via IRC, that the
+ previous fix only worked when explicitly set. When nothing is
+ set, and the option is implied, it breaks, because configure sets
+ the prefix to 'NONE'. Fixing.
+
+2008-07-30 20:37 +0000 [r134540-134595] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_dundi.c: Reduce stack consumption by 12.5% of the max
+ stack size to fix a crash when compiled with LOW_MEMORY. (closes
+ issue #13154) Reported by: edantie
+
+ * funcs/func_curl.c: Fix a memory leak in func_curl. Every thread
+ that used this function leaked an allocation the size of a
+ pointer. (reported by jmls in #asterisk-dev)
+
+2008-07-30 19:47 +0000 [r134480-134536] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, configure.ac: Only override sysconfdir and mandir when
+ prefix=/usr (closes issue #13093) Reported by: pabelanger
+
+ * res/res_agi.c: launch_netscript sometimes returns -1, which fails
+ to set AGISTATUS. Map failure to -1, so that AGISTATUS is always
+ set. (closes issue #13199) Reported by: smw1218
+
+2008-07-30 18:31 +0000 [r134475] Mark Michelson <mmichelson@digium.com>
+
+ * main/app.c: Fix a spot where a function could return without
+ bringing a channel out of autoservice.
+
+2008-07-30 15:29 +0000 [r134254-134352] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile: use the proper method for building version.h
+
+ * include/asterisk/dahdi_compat.h, apps/app_dahdibarge.c,
+ channels/chan_dahdi.c, apps/app_meetme.c, apps/app_flash.c,
+ apps/app_dahdiscan.c, apps/app_dahdiras.c, codecs/codec_dahdi.c:
+ build against the now-typedef-free dahdi/user.h
+
+2008-07-29 15:54 +0000 [r134223] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Merging the imap_consistency branch. The
+ main aim of this branch was to make the IMAP code function in the
+ same manner as the ODBC code does, eliminating the need for so
+ many IMAP-specific code chunks. The focal point of all of this
+ work was to make the various macros (e.g. RETRIEVE, DISPOSE)
+ functionally equivalent. While doing the above work, I also fixed
+ a few bugs that I came across in my testing. Among these were 1.
+ Fixed message forwarding. This was completely broken when using
+ IMAP. 2. Fixed the inability to save new messages as old and vice
+ versa. 3. Fixed the "delete" options in voicemail.conf when using
+ IMAP storage. Even though a few bugs were fixed and the code is a
+ lot more consistent, the one thing that was *not* improved in
+ this branch was performance. The merge of this to trunk may not
+ come immediately due to the amount of work it will probably
+ involve. (closes issue #12764) Reported by: balsamcn
+
+2008-07-28 21:50 +0000 [r134161] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Detect when sox fails to raise the volume,
+ because sox can't read the file. (closes issue #12939) Reported
+ by: rickbradley Patches: 20080728__bug12939.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: rickbradley
+
+2008-07-26 15:31 +0000 [r133980] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c, include/asterisk/doxyref.h: Add the licensing
+ section to the docs in 1.4, as well, so that we can work on
+ having an accurate list for each version of Asterisk that is
+ supported
+
+2008-07-25 18:00 +0000 [r133649-133709] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Remove unnecessary mmap flag (Closes issue
+ #13161)
+
+ * main/channel.c, channels/chan_agent.c, main/devicestate.c: Fix
+ some errant device states by making the devicestate API more
+ strict in terms of the device argument (only without the unique
+ identifier appended). (closes issue #12771) Reported by: davidw
+ Patches: 20080717__bug12771.diff.txt uploaded by Corydon76
+ (license 14) Tested by: davidw, jvandal, murf
+
+2008-07-25 15:00 +0000 [r133578] Russell Bryant <russell@digium.com>
+
+ * /, LICENSE: Merged revisions 133577 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r133577 | russell | 2008-07-25 10:00:13 -0500 (Fri, 25 Jul 2008)
+ | 2 lines Fix the IAX2 URI for calling Digium ........
+
+2008-07-25 14:40 +0000 [r133572] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: We need to make sure to null-terminate the
+ "name" portion of SIP URI parameters so that there are no bogus
+ comparisons. Thanks to bbryant for pointing this out.
+
+2008-07-24 21:17 +0000 [r133361-133488] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Fix rtautoclear and rtcachefriends (Closes
+ issue #12707)
+
+ * /: Blocked revisions 133360 via svnmerge ........ r133360 |
+ tilghman | 2008-07-23 22:46:01 -0500 (Wed, 23 Jul 2008) | 2 lines
+ This part was not correctly patched for AST-2008-010. ........
+
+2008-07-23 21:49 +0000 [r133295] Jason Parker <jparker@digium.com>
+
+ * channels/chan_dahdi.c: inbandrelease is gone - it's now
+ inbanddisconnect
+
+2008-07-23 21:05 +0000 [r133226-133237] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/stringfields.h, main/utils.c: revert an
+ optimization that broke ABI... thanks russell!
+
+ * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c,
+ apps/app_dahdibarge.c, channels/chan_dahdi.c,
+ apps/app_dahdiras.c: make some more changes to the dahdi/zap
+ channel name support stuff to ensure allthe globals are 'const',
+ and clean up mmichelson's changes to app_chanspy to simplify the
+ code
+
+2008-07-23 19:39 +0000 [r132974-133169] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c,
+ channels/chan_dahdi.c: As suggested by seanbright, the
+ PSEUDO_CHAN_LEN in app_chanspy should be set at load time, not at
+ compile time, since dahdi_chan_name is determined at load time.
+ Also changed the next_unique_id_to_use to have the static
+ qualifier. Also added the dahdi_chan_name_len variable so that
+ strlen(dahdi_chan_name) isn't necessary. Thanks to seanbright for
+ the suggestion.
+
+ * apps/app_chanspy.c: Zap/pseudo is ten characters, but
+ DAHDI/pseudo is twelve. The strncmp call in next_channel should
+ account for this.
+
+ * apps/app_chanspy.c: Update the "last" channel in next_channel in
+ app_chanspy so that the same pseudo channel isn't constantly
+ returned. related to issue #13124
+
+ * channels/chan_dahdi.c: Small cleanup. Move the declaration of the
+ DAHDI_SPANINFO variable to the block where it is used. This
+ allows one less #ifdef HAVE_PRI to clutter things up. Thanks to
+ Tzafrir for pointing this out on #asterisk-dev
+
+ * channels/chan_dahdi.c: Fix building of chan_dahdi when HAVE_PRI
+ is not defined.
+
+2008-07-23 15:52 +0000 [r132872-132942] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: ensure that after a channel is created, if
+ it happened to be in 'channel alarm' state, when that alarm
+ clears we won't generate a spurious 'alarm cleared' message
+ (closes issue #12160) Reported by: tzafrir
+
+ * include/asterisk/stringfields.h, main/utils.c: minor optimization
+ for stringfields: when a field is being set to a larger value
+ than it currently contains and it happens to be the most recent
+ field allocated from the currentl pool, it is possible to 'grow'
+ it without having to waste the space it is currently using (or
+ potentially even allocate a new pool)
+
+2008-07-23 11:37 +0000 [r132826] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.c: another Fix because of r119585, this
+ commit has broken high frequented BRI Ports, there was a
+ possibility that a channel, that was marked as in_use would be
+ reused later, the corresponding port could got stuck then. So it
+ is recommended to upgrade for chan_misdn users.
+
+2008-07-22 22:14 +0000 [r132790] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Allow Spiraled INVITEs to work correctly
+ within Asterisk. Prior to this change, a spiraled INVITE would
+ cause a 482 Loop Detected to be sent to the caller. With this
+ change, if a potential loop is detected, the Request-URI is
+ inspected to see if it has changed from what was originally
+ received. If pedantic mode is on, then this inspection is fully
+ RFC 3261 compliant. If pedantic mode is not on, then a string
+ comparison is used to test the equality of the two R-URIs. This
+ has been tested by using OpenSER to rewrite the R-URI and send
+ the INVITE back to Asterisk. (closes issue #7403) Reported by:
+ stephen_dredge
+
+2008-07-22 22:11 +0000 [r132784-132787] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/options.h, main/asterisk.c,
+ apps/app_dahdibarge.c, channels/chan_dahdi.c, apps/app_flash.c,
+ apps/app_dahdiras.c: fix up namespace pollution for
+ dahdi_chan_mode enum correct registration of AMI actions in
+ chan_dahdi; in zap-only mode, only register the Zap flavors of
+ the actions (and use Zap prefixes for headers and acks), but in
+ dahdi+zap mode, register both Zap and DAHDI flavors of actions
+
+ * Makefile.rules: add rules to create preprocessor output... useful
+ for debugging macros
+
+2008-07-22 21:19 +0000 [r132713] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/iax.conf.sample, /, channels/chan_iax2.c: Merged
+ revisions 132711 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r132711 | tilghman | 2008-07-22 16:14:10 -0500 (Tue, 22 Jul 2008)
+ | 2 lines Fixes for AST-2008-010 and AST-2008-011 ........
+
+2008-07-22 21:17 +0000 [r132704-132712] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: ensure that if any alarms exist at channel
+ creation time, they are handled identically to if they occurred
+ later, so that later alarm clearing will work properly and 'make
+ sense' (closes issue #12160) Reported by: tzafrir
+
+ * configure, configure.ac, acinclude.m4: make AST_C_COMPILE_CHECK
+ able to print a 'pretty' description of what it is doing
+
+2008-07-22 20:10 +0000 [r132645] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c, doc/sip-retransmit.txt (added): The most
+ common question on the #asterisk iRC channel and on mailing lists
+ seems to be in regards to an error message when retransmit fails.
+ This is frequently misunderstood as a failure of Asterisk, not a
+ failure of the network to reach the other party. This document
+ tries to assist the Asterisk user in sorting out these issues by
+ explaining the logic and pointing at some possible causes.
+ Hopefully, we will get other questions now :-)
+
+2008-07-22 19:57 +0000 [r132571-132642] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: correct wording in comment
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: use renamed
+ libpri API call for controlling this feature (was improperly
+ named before)
+
+ * channels/chan_dahdi.c: teach chan_dahdi how to find the D-channel
+ on BRI spans, and don't attempt to use channel 24 as a D-channel
+ on spans of unexpected sizes
+
+2008-07-21 20:51 +0000 [r132506-132507] Brett Bryant <bbryant@digium.com>
+
+ * apps/app_sendtext.c: Fix a bug where SENDTEXTSTATUS isn't set
+ properly when it isn't supported on a channel (yet _another_
+ useful patch by eliel). (issue #13081) Reported by: eliel
+ Patches: app_sendtext1.4.c uploaded by eliel (license 64) Tested
+ by: eliel
+
+ * channels/chan_iax2.c: Fix a bug in 1.4 branch with iax2 channels
+ not being removed when a call was rejected (from the calling box,
+ not the box that denied the registration). Related to revisions
+ 132466 in trunk, and 132467 in 1.6.0. Earlier I had accidently
+ tested 1.4 with a backport from those revisions, so I didn't see
+ this problem (oops).
+
+2008-07-19 16:45 +0000 [r132311] Kevin P. Fleming <kpfleming@digium.com>
+
+ * LICENSE: grant a license exception to allow distribution of
+ Asterisk binaries that use the UW IMAP Toolkit (which is licensed
+ under a non-GPL-compatible license)
+
+2008-07-18 19:06 +0000 [r131970-132112] Tilghman Lesher <tlesher@digium.com>
+
+ * main/say.c: Fix for Taiwanese number syntax (closes issue #12319)
+ Reported by: CharlesWang Patches: saynumber-tw-1.4.18.1.patch
+ uploaded by CharlesWang (license 444)
+
+ * main/config.c: Textual clarification (closes issue #13106)
+ Reported by: flefoll Patches:
+ config.c.br14.120173.patch-unknown-directive uploaded by flefoll
+ (license 244)
+
+ * include/asterisk/sched.h, channels/chan_iax2.c: Spinlock within
+ the destroy, to allow a scheduled job to continue, if it's
+ waiting on the mutex which the destroy thread has.
+
+ * main/sched.c: Oops
+
+ * main/sched.c, include/asterisk/sched.h: Preserve ABI
+ compatibility with last change
+
+ * main/sched.c, include/asterisk/sched.h, channels/chan_iax2.c:
+ Make the ast_assert call within ast_sched_del report something
+ useful.
+
+2008-07-18 16:15 +0000 [r131921] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/dlfcn.c (removed), main/loader.c, main/Makefile,
+ include/asterisk/dlfcn-compat.h (removed): remove the dlfcn
+ compatibility stuff, because no platforms that Asterisk currently
+ runs on it use it, and it doesn't build anyway
+
+2008-07-18 15:34 +0000 [r131915] Brett Bryant <bbryant@digium.com>
+
+ * res/res_features.c: Fix a bug in blind transfers where the
+ BLINDTRANSFER variable isn't always set to the other end of the
+ blind transfer. (closes issue #12586)
+
+2008-07-17 20:35 +0000 [r131790] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c: Revert part of issue #5620 (revision 6965)
+ as it appears that it was in error. This should fix talk call
+ progress on analog lines. (closes issue #12178) Reported by:
+ michael-fig Patches: 20080717__bug12178.diff.txt uploaded by
+ Corydon76 (license 14)
+
+2008-07-16 22:17 +0000 [r131491] Brett Bryant <bbryant@digium.com>
+
+ * channels/chan_iax2.c: Fix a bug in iax2 registration that allowed
+ peers to register with case-insensitive names (user_cmp_cb and
+ peer_cmp_cb are now both case-sensitive). (closes issue #13091)
+
+2008-07-16 21:46 +0000 [r131480] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Apparently, in certain cases, a callno is
+ already destroyed when iax2_destroy is called.
+
+2008-07-16 20:47 +0000 [r131421] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Always ensure that the channel's tech_pvt
+ reference is NULL after calling the destroy callback. (closes
+ issue #13060) Reported by: jpgrayson Patches:
+ chan_iax2_tech_pvt_crash.patch uploaded by jpgrayson (license
+ 492)
+
+2008-07-16 20:23 +0000 [r131299-131369] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Move the init_queue call back to where it used
+ to be (changed Sept 12 last year). It was moved then to prevent a
+ memory leak. Since then, the same memory leak recurred and was
+ fixed in a better way. Now it has been found that the placement
+ of this init_queue call can cause problems if a realtime queue
+ has values changed to an empty string. The problem is that the
+ default value for that queue parameter would not be set. (closes
+ issue #13084) Reported by: elbriga
+
+ * apps/app_queue.c: Apparently, "thread safety" is important,
+ whatever that means. :P (Thanks Russell!)
+
+ * apps/app_queue.c: Make absolutely certain that the transfer
+ datastore is removed from the calling channel once the caller is
+ finished in the queue. This could have weird con- sequences when
+ dialing local queue members when multiple transfers occur on a
+ single call. Also fixed a memory leak that would occur when an
+ attended transfer occurred from a queue member. (closes issue
+ #13047) Reported by: festr
+
+2008-07-16 17:53 +0000 [r131242] Steve Murphy <murf@digium.com>
+
+ * pbx/pbx_ael.c: (closes issue #13090) Reported by: murf The
+ problem was that, esoteric as it is, because the hangerupper
+ context immediately preceded the std-priv-extent macro, that the
+ checking code accidentally would fall from traversing hangerupper
+ into the std-priv-exten macro, where it would hit the hangerupper
+ in the 'includes', and proceed into an infinite recursion. A
+ small fix to traverse into the statements of the context instead
+ of the context solves this issue. I also added some commented out
+ printfs for debug, which were pretty handy in the face of a dorky
+ gdb. This was a problem around since the package was first
+ written; but evidently pretty rare in turning up in the field.
+
+2008-07-15 17:47 +0000 [r131012] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/cdr.c: remove 4 lines of redundant code. (closes issue
+ #13080) Reported by: gknispel_proformatique Patches:
+ trunk_ast_cdr_setapp.patch uploaded by gknispel (license 261)
+
+2008-07-15 17:19 +0000 [r130889-130959] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, channels/chan_sip.c: astman_send_error does not
+ need a newline appended -- the API takes care of that for us.
+ (closes issue #13068) Reported by: gknispel_proformatique
+ Patches: asterisk_1_4_astman_send.patch uploaded by gknispel
+ (license 261) asterisk_trunk_astman_send.patch uploaded by
+ gknispel (license 261)
+
+ * channels/chan_iax2.c: Override the callerid in all cases when the
+ callerid is set in the user, not just when a remote callerid is
+ set. Also, if not set in the user, allow the remote CallerID to
+ pass through. (closes issue #12875) Reported by: dimas Patches:
+ 20080714__bug12875.diff.txt uploaded by Corydon76 (license 14)
+
+2008-07-14 17:50 +0000 [r130792] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c: Add a check to the CAN_EARLY_BRIDGE macro in
+ app_dial to be sure there are no audiohooks present on the
+ channels involved. This fixed a one-way audio situation I had in
+ my test setup. I couldn't find any open issues that suggested
+ one-way audio with regards to mixmonitor (or other audiohook)
+ usage, though.
+
+2008-07-14 17:10 +0000 [r130735] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/dnsmgr.c: notify the user that dnsmgr refresh wont work when
+ dnsmgr is not enabled. Previously this command would
+ automagically appear and disappear. This was confusing. (closes
+ issue #12796) Reported by: chappell Patches:
+ dnsmgr_refresh_3.diff uploaded by chappell (license 8) Tested by:
+ russell, chappell, mvanbaak
+
+2008-07-14 10:38 +0000 [r130634] Russell Bryant <russell@digium.com>
+
+ * main/audiohook.c: Bump up the debug level for a message.
+
+2008-07-13 22:48 +0000 [r130573] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/manager.c: fix memory leak when originate from manager
+ cannot create a thread (closes issue #13069) Reported by:
+ gknispel_proformatique Patches:
+ asterisk_trunk_action_originate.patch uploaded by gknispel
+ (license 261) Tested by: gknispel_proformatique, mvanbaak
+
+2008-07-13 17:56 +0000 [r130514] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Reverting 2 changesets, as it breaks
+ incoming IAX2 calls (Related to issue #12963) Reported by:
+ mvanbaak
+
+2008-07-12 10:25 +0000 [r130373] Michiel van Baak <michiel@vanbaak.info>
+
+ * pbx/pbx_ael.c: in 1.4 the functions still have | as argument
+ seperator. This commit fixes the use of RAND in the ael random
+ function. (closes issue #13061) Reported by: danpwi
+
+2008-07-11 22:23 +0000 [r130298-130317] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile: forcibly remove the modules that are changing names
+
+ * include/asterisk/options.h, main/asterisk.c, cdr/cdr_csv.c,
+ Makefile, main/channel.c, apps/app_dahdibarge.c,
+ channels/chan_dahdi.c, doc/hardware.txt, apps/app_flash.c,
+ apps/app_dahdiras.c, main/file.c,
+ contrib/utils/zones2indications.c, include/asterisk/channel.h,
+ channels/chan_iax2.c: a whole pile of Zaptel/DAHDI compatibility
+ work, with lots more to come... this tree is not yet ready for
+ users to be easily upgrading or switching, but it needs to be :-)
+
+2008-07-11 20:03 +0000 [r130173-130236] Mark Michelson <mmichelson@digium.com>
+
+ * main/audiohook.c: Remove redundant logic
+
+ * main/audiohook.c: Fix a typo in audiohook_read_frame_both. While
+ this change has not been proven to fix any specific issue, it is
+ incorrect and could cause unforeseen problems.
+
+2008-07-11 18:51 +0000 [r130102-130169] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Ensure that a destination callno of 0 will
+ not match for frames that do not start a dialog (new, lagrq, and
+ ping). (closes issue #12963) Reported by: russellb Patches:
+ chan_iax2_dup_new_fix4.patch uploaded by jpgrayson (license 492)
+
+ * channels/chan_agent.c: Pass the devicestate from an underlying
+ channel up through the Agent channel. This should make the Agent
+ always report the correct device state, even when the underlying
+ channel is used for other purposes. (closes issue #12773)
+ Reported by: davidw Patches: 20080710__bug12773.diff.txt uploaded
+ by Corydon76 (license 14) Tested by: davidw
+
+2008-07-11 16:08 +0000 [r130039-130042] Kevin P. Fleming <kpfleming@digium.com>
+
+ * doc/configuration.txt, configs/extensions.conf.sample,
+ configs/sla.conf.sample, configs/zapata.conf.sample (removed),
+ contrib/scripts/autosupport, README,
+ configs/chan_dahdi.conf.sample (added), channels/chan_dahdi.c,
+ include/asterisk/doxyref.h, doc/sla.tex, doc/ael.txt,
+ configs/extensions.ael.sample, configs/smdi.conf.sample: new
+ installations should be using DAHDI instead of Zaptel, so the
+ sample config file is now chan_dahdi.conf instead of zapata.conf
+ also, convert remaining references to zapata.conf in various
+ places
+
+ * configs/zapata.conf.sample, channels/chan_dahdi.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: add support for a
+ configuration parameter for 'inband audio during RELEASE', which
+ is currently mandatory in libpri-1.4.4 but will become
+ configurable in libpri-1.4.5 later today (related to issue
+ #13042)
+
+2008-07-11 14:18 +0000 [r129970] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/astobj.h: add a simple ASTOBJ_TRYWRLOCK macro
+ ...
+
+2008-07-11 14:14 +0000 [r129907-129967] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/astmm.c: simplify calculation
+
+ * main/astmm.c: fix a flaw found while experimenting with structure
+ alignment and padding; low-fence checking would not work properly
+ on 64-bit platforms, because the compiler was putting 4 bytes of
+ padding between the fence field and the allocation memory block
+ added a very obvious runtime warning if this condition reoccurs,
+ so the developer who broke it can be chastised into fixing it :-)
+
+ * sounds/Makefile: don't attempt to set user/group ownership of
+ extracted sound files (reported on asterisk-users) (closes issue
+ #13059)
+
+2008-07-10 21:57 +0000 [r129741-129803] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Correctly deal with duplicate NEW frames
+ (due to retransmission). Also, fixup the destination call number
+ matching to be more strict and reliable. (closes issue #12963)
+ Reported by: jpgrayson Patches: chan_iax2_dup_new_fix3.patch
+ uploaded by jpgrayson (license 492) Tested by: jpgrayson,
+ Corydon76
+
+ * res/res_config_odbc.c: Oops
+
+2008-07-10 16:03 +0000 [r129567] Russell Bryant <russell@digium.com>
+
+ * sample.call: Note that pbx_spool.so is the module used for call
+ files (inspired by a question in #asterisk)
+
+2008-07-10 13:57 +0000 [r129505] Sean Bright <sean.bright@gmail.com>
+
+ * main/editline: Update svn:ignore
+
+2008-07-09 19:32 +0000 [r129436] Mark Michelson <mmichelson@digium.com>
+
+ * main/rtp.c: Fix a problem where inbound rfc2833 audio would be
+ sent to the core instead of being P2P bridged. When the core
+ regenerated the rfc2833 packet for the outbound leg, the SSRC
+ would be different than the RTP audio on the call leg causing
+ DTMF detection issues on the far end. (closes issue #12955)
+ Reported by: tonyredstone Patches: dynamic_rtp.patch uploaded by
+ tsearle (license 373) Tested by: tonyredstone
+
+2008-07-09 13:41 +0000 [r129343] Sean Bright <sean.bright@gmail.com>
+
+ * main/editline/makelist (removed), main/editline/makelist.in
+ (added), main/editline/configure, main/editline/Makefile.in,
+ main/editline/configure.in: Look for the system installed awk
+ instead of assuming it's at /usr/bin/awk. Pointed out by jmls via
+ #asterisk-dev.
+
+2008-07-08 21:31 +0000 [r129158-129208] Mark Michelson <mmichelson@digium.com>
+
+ * doc/imapstorage.txt: Update documentation to have the correct
+ option name
+
+ * apps/app_voicemail.c, doc/imapstorage.txt: Backport TCP-related
+ timeouts to IMAP voicemail in 1.4 since it should solve bugs
+ people are experiencing. Specifically, there are times where
+ communication with the IMAP server causes system calls to block
+ forever. If this should happen when querying the mailbox so that
+ chan_sip's do_monitor thread can send MWI to a phone, it means
+ that SIP calls cannot be processed any more. The timeout options
+ are outlined in doc/imapstorage.txt. Defaults for the timeouts
+ are sixty seconds. (closes issue #12987) Reported by: mthomasslo
+
+2008-07-08 20:27 +0000 [r129047-129149] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_dial.c, channels/chan_sip.c, include/asterisk/causes.h:
+ Cause SIP to return a 480 instead of a 404 when a sip peer
+ exists, but is not registered. (closes issue #12885) Reported by:
+ ibc Patches: 20080701__bug12885__2.diff.txt uploaded by Corydon76
+ (license 14) Tested by: ibc
+
+ * channels/chan_iax2.c: Timestamp decoding for video mini-frames is
+ bogus, because the timestamp only includes 15 bits, unlike voice
+ frames, which contain a 16-bit timestamp. (closes issue #13013)
+ Reported by: jpgrayson Patches: chan_iax2_unwrap_ts.patch
+ uploaded by jpgrayson (license 492)
+
+2008-07-08 09:52 +0000 [r128912-128950] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Don't hangup the call if we can't resolve
+ the Contact if there's a proxy route set for the call. ---- This
+ comment was added a while ago and today it hit me badly. /* OEJ:
+ Possible issue that may need a check: If we have a proxy route
+ between us and the device, should we care about resolving the
+ contact or should we just send it? */
+
+ * channels/chan_sip.c: Fix issues where repeated messages where
+ ignored, but retransmitted reliably instead of unreliably.
+ Reported by: johan Patches: 12746.txt uploaded by oej (license
+ 306) Tested by: johan (issue #12746)
+
+2008-07-08 00:01 +0000 [r128812-128856] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Check for non-NULL before stripping
+ characters. (closes issue #12954) Reported by: bfsworks Patches:
+ 20080701__bug12954.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: deti
+
+ * apps/app_voicemail.c: Stop using deprecated method, as requested
+ by Kevin.
+
+2008-07-07 22:41 +0000 [r128795] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Fix handling of when a pvt disappears.
+ Properly return the pvt locked and don't hold the pvt lock while
+ destroying the ast_channel. (closes issue #13014) Reported by:
+ jpgrayson Patches: chan_iax2_ast_iax2_new2.patch uploaded by
+ jpgrayson (license 492)
+
+2008-07-07 20:47 +0000 [r128737] Sean Bright <sean.bright@gmail.com>
+
+ * channels/chan_iax2.c: Remove spurious trailing whitespace from
+ log messages and fix a spelling error in a log message. (closes
+ issue #13017) Reported by: jpgrayson Patches:
+ chan_iax2_space_after_newline.patch uploaded by jpgrayson
+ (license 492) chan_iax2_spelling.patch uploaded by jpgrayson
+ (license 492)
+
+2008-07-07 17:02 +0000 [r128639] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_iax2.c: By using the iaxdynamicthreadcount to
+ identify a thread, it was possible for thread identifiers to be
+ duplicated. By using a globally-unique monotonically- increasing
+ integer, this is now avoided. (closes issue #13009) Reported by:
+ jpgrayson Patches: chan_iax2_dyn_threadnum.patch uploaded by
+ jpgrayson (license 492)
+
+2008-07-07 16:51 +0000 [r128637] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configure, configure.ac: use tzafrir's patch to fix this problem
+ properly... i made the previous set of changes without thoroughly
+ testing them, doh! (closes issue #12911) Reported by: tzafrir
+ Patches: custum_dahdi_configure_2.diff uploaded by tzafrir
+ (license 46) Tested by: tzafrir
+
+2008-07-04 16:11 +0000 [r127973-128029] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_config.c: Move the free down one
+
+ * main/pbx.c, include/asterisk/pbx.h, pbx/pbx_config.c: Fix the
+ 'dialplan remove extension' logic, so that it a) works with
+ cidmatch, and b) completes contexts correctly when the extension
+ is ambiguous. (closes issue #12980) Reported by: licedey Patches:
+ 20080703__bug12980.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Corydon76
+
+2008-07-03 22:20 +0000 [r127754-127895] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/Makefile: remove this, it has been moved to the main
+ Makefile
+
+ * Makefile, main/editline/np/vis.c: a couple of small
+ Solaris-related fixes (closes issue #11885) Reported by: snuffy,
+ asgaroth
+
+ * configure, main/Makefile, configure.ac, acinclude.m4: ensure that
+ DAHDI_INCLUDE and ZAPTEL_INCLUDE are added in all the places
+ needed improve AST_EXT_LIB_CHECK to accept (and remember)
+ additional CFLAGS data like it does in trunk already (closes
+ issue #12911) Reported by: tzafrir
+
+2008-07-03 00:16 +0000 [r127663] Steve Murphy <murf@digium.com>
+
+ * main/cdr.c, main/channel.c, channels/chan_dahdi.c, main/pbx.c,
+ channels/chan_sip.c, res/res_features.c, include/asterisk/cdr.h:
+ The CDRfix4/5/6 omnibus cdr fixes. (closes issue #10927) Reported
+ by: murf Tested by: murf, deeperror (closes issue #12907)
+ Reported by: falves11 Tested by: murf, falves11 (closes issue
+ #11849) Reported by: greyvoip As to 11849, I think these changes
+ fix the core problems brought up in that bug, but perhaps not the
+ more global problems created by the limitations of CDR's
+ themselves not being oriented around transfers. Reopen if necc,
+ but bug reports are not the best medium for enhancement
+ discussions. We need to start a second-generation CDR
+ standardization effort to cover transfers. (closes issue #11093)
+ Reported by: rossbeer Tested by: greyvoip, murf
+
+2008-07-02 20:47 +0000 [r127560] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_agent.c: Fix thread-safety of some of the
+ pbx_builtin_getvar_helper calls
+
+2008-07-02 19:47 +0000 [r127501] Tilghman Lesher <tlesher@digium.com>
+
+ * main/acl.c: Merged revisions 127466 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r127466 |
+ tilghman | 2008-07-02 13:31:11 -0500 (Wed, 02 Jul 2008) | 6 lines
+ Solaris fix (closes issue #12949) Reported by: snuffy Patches:
+ bug_12949.diff uploaded by snuffy (license 35) ........
+
+2008-07-01 23:36 +0000 [r127244] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Add error message to failed open(2) calls
+ inside the copy() function of app_voicemail. This idea came as
+ part of my work in helping to resolve issue #12764.
+
+2008-07-01 20:25 +0000 [r126999-127133] Tilghman Lesher <tlesher@digium.com>
+
+ * build_tools/cflags.xml, channels/chan_iax2.c: Disable the old,
+ slow search for matching callno in chan_iax2 (but allow it to be
+ reenabled for debugging)
+
+ * channels/chan_iax2.c: Oops
+
+ * channels/chan_iax2.c: Change around how we schedule pings and
+ lagrqs, and fix a reason why the jobs were not getting properly
+ cancelled. (closes issue #12903) Reported by: stevedavies
+ Patches: 20080620__bug12903__2.diff.txt uploaded by Corydon76
+ (license 14) Tested by: stevedavies
+
+ * channels/chan_iax2.c: Suppress annoying warning by finding the
+ remaining cases where the callno is not in the hash.
+
+2008-07-01 14:59 +0000 [r126735-126902] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Use domain part of SIP uri in register=
+ configuration as fromdomain. Reported by: one47 Patches:
+ sip-reg-fromdom2.dpatch uploaded by one47 (license 23) (closes
+ issue #12474)
+
+ * channels/chan_sip.c: Handle escaped URI's in call pickups. Patch
+ by oej and IgorG. Reported by: IgorG Patches:
+ bug12299-11062-v2.patch uploaded by IgorG (license 20) Tested by:
+ IgorG, oej (closes issue #12299)
+
+ * configs/sip.conf.sample: Clear up documentation on "domain="
+ setting in sip.conf Reported by: davidw (closes issue #12413)
+
+ * channels/chan_sip.c: Report 200 OK to all in-dialog OPTIONs
+ requests (to confirm that the dialog exist). Don't bother
+ checking the request URI. (closes issue #11264) Reported by: ibc
+
+ * channels/chan_sip.c: Fix bad XML for hold notification. Reported
+ by: gowen72 Patches: hold.patch uploaded by gowen72 (license 432)
+ (closes issue #12942)
+
+2008-06-30 23:11 +0000 [r126680] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Load the proper channel configuration file
+ based on which driver was detected.
+
+2008-06-30 22:30 +0000 [r126674] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/zapata.conf.sample: Add note about other names for
+ EuroISDN
+
+2008-06-30 16:05 +0000 [r126573] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/lock.h: Fix a typo in the non-DEBUG_THREADS
+ version of the recently added DEADLOCK_AVOIDANCE() macro. This
+ caused the lock to not actually be released, and as a result, not
+ avoid deadlocks at all. This resolves the issues reported in the
+ last while about Asterisk locking up all over the place (and most
+ commonly, in chan_iax2). (closes issue #12927) (closes issue
+ #12940) (closes issue #12925) (potentially closes others ...)
+
+2008-06-30 12:50 +0000 [r126516] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Send all responses to an INVITE reliably, so
+ that we retransmit if we don't get an ACK and also fail if we
+ don't get the very same precious ACK. Based on patch by tsearle,
+ with my own additions. (closes issue #12951) Reported by: tsearle
+ Patches: busy_retransmit.patch uploaded by tsearle (license 373)
+
+2008-06-29 18:05 +0000 [r126395] Kevin P. Fleming <kpfleming@digium.com>
+
+ * pbx/Makefile: ignore warnings for prototypes in GTK headers
+
+2008-06-27 22:01 +0000 [r125740-126056] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: When we get a 408 Timeout, don't stop trying
+ to re-register. (closes issue #12863) Reported by: ricvil
+
+ * include/asterisk/tonezone_compat.h: Since HAVE_DAHDI is defined
+ to HAVE_ZAPTEL in dahdi_compat.h, we must first check for
+ HAVE_ZAPTEL. (closes issue #12938) Reported by: opticron Patches:
+ tonezone_compat.diff uploaded by opticron (license 267)
+
+ * main/utils.c, include/asterisk/lock.h: In this debugging
+ function, copy to a buffer instead of using potentially unsafe
+ pointers.
+
+ * channels/chan_local.c: Add proper deadlock avoidance. (closes
+ issue #12914) Reported by: ozan Patches:
+ 20080625__bug12914.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: ozan
+
+2008-06-26 23:03 +0000 [r125587] Jason Parker <jparker@digium.com>
+
+ * main/utils.c: Make sure to unlock the lock_info lock (huh?).
+ Possible deadlock?
+
+2008-06-26 22:52 +0000 [r125476-125585] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Add the interface of a queue member to the
+ output of the "queue show" command so that it can easily be
+ associated with a queue member's name. This helps so that the
+ appropriate queue member can be removed or paused since the
+ interface is required, not the member's name. (closes issue
+ #12783) Reported by: davevg Patches: app_queue.diff uploaded by
+ davevg (license 209) with small mod from me
+
+ * apps/app_queue.c: Backport of attended transfer queue_log patch
+ from trunk. This patch allows for attended transfers to be logged
+ in the queue_log the same way that blind transfers have always
+ been. It was decided by popular opinion on the asterisk-dev
+ mailing list that this should be backported to 1.4. Thanks to
+ everyone who gave an opinion.
+
+ * apps/app_queue.c: Prior to this patch, the "queue show" command
+ used cached information for realtime queues instead of giving
+ up-to-date info. Now realtime is queried for the latest and
+ greatest in queue info. (closes issue #12858) Reported by: bcnit
+ Patches: queue_show.patch uploaded by putnopvut (license 60)
+
+2008-06-26 16:32 +0000 [r125384] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Add support for peer realm based auth (a few
+ missing lines, the rest is well documented but never worked)
+
+2008-06-26 15:30 +0000 [r125327] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: ensure that (whenever possible) if we
+ generate a log message because an ioctl() call to DAHDI/Zaptel
+ failed, that we include the reason it failed by including the
+ stringified error number (issue AST-80)
+
+2008-06-26 11:01 +0000 [r125218-125276] Tilghman Lesher <tlesher@digium.com>
+
+ * main/rtp.c: Check for rtcp structure before trying to delete
+ schedule. (closes issue #12872) Reported by: destiny6628 Patches:
+ 20080621__bug12872.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: destiny6628
+
+ * configs/agents.conf.sample: Document ackcall=always. (closes
+ issue #12852) Reported by: davidw
+
+2008-06-25 22:21 +0000 [r125132] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_rpt.c, include/asterisk/dahdi_compat.h,
+ channels/chan_dahdi.c, configure,
+ include/asterisk/tonezone_compat.h (added), configure.ac: allow
+ tonezone to live in a different place than DAHDI/Zaptel, since
+ dahdi-tools and dahdi-linux are now separate packages and can be
+ installed in different places don't include tonezone.h in
+ dahdi_compat.h, because only a couple of modules need it get
+ app_rpt building again after the DAHDI changes (closes issue
+ #12911) Reported by: tzafrir
+
+2008-06-25 00:46 +0000 [r124908-124965] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c: Pvt deadlock causes some channels to get
+ stuck in Reserved status. (closes issue #12621) Reported by:
+ fabianoheringer Patches: 20080612__bug12621.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: fabianoheringer
+
+ * apps/app_voicemail.c: Occasionally control characters find their
+ way into CallerID. These need to be stripped prior to placing
+ CallerID in the headers of an email. (closes issue #12759)
+ Reported by: RobH Patches: 20080602__bug12759__2.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: RobH
+
+ * channels/chan_sip.c: Don't access the pvt structure if unable to
+ acquire the lock. (closes issue #12162) Reported by: norman
+ Patches: 12162-lockfail.diff uploaded by qwell (license 4)
+
+2008-06-23 21:22 +0000 [r124743] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c: emit a warning if the old IAX2 call
+ searching code finds a call when the new code did not... so that
+ we can get rid of the old code in 2-3 months
+
+2008-06-22 02:54 +0000 [r124540] Steve Murphy <murf@digium.com>
+
+ * apps/app_forkcdr.c: (closes issue #12910) Reported by: chris-mac
+ Sorry, my testing did not contain the simple case of forkCDR(v),
+ I am much embarrassed to admit. If I had, I would have more
+ solidly initialized the opts element for varset.
+
+2008-06-20 23:12 +0000 [r124395-124450] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_rpt.c: usleep with a value over 1,000,000 is
+ nonportable. Changing to use sleep() instead. (closes issue
+ #12814) Reported by: pputman Patches: app_rtp_sleep.patch
+ uploaded by pputman (license 81)
+
+ * main/app.c: If the last character in a string to be parsed is the
+ delimiter, then we should count that final empty string as an
+ additional argument.
+
+2008-06-20 21:14 +0000 [r124372] Jeff Gehlbach <jeffg@opennms.org>
+
+ * doc/asterisk-mib.txt, doc/digium-mib.txt: Fix issues in
+ digium-mib.txt and asterisk-mib.txt to placate smilint - bug
+ 12905
+
+2008-06-20 20:16 +0000 [r124182-124315] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c: When using a Local channel, started by a
+ call file, with a destination of an AGI script, the AGI script
+ does not always get notified of a hangup if the underlying
+ channel hangs up early. (closes issue #11833) Reported by: IgorG
+ Patches: local_hangup-v1.diff uploaded by IgorG (license 20)
+
+ * channels/chan_dahdi.c: It's possible for a hangup to be received,
+ even just after the initial cid spill. (closes issue #12453)
+ Reported by: Alex728 Patches: 20080604__bug12453.diff.txt
+ uploaded by Corydon76 (license 14)
+
+2008-06-19 20:28 +0000 [r124112] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Fix IMAP forwarding so that messages are
+ sent to the proper mailbox. (closes issue #12897) Reported by:
+ jaroth Patches: destination_forward.patch uploaded by jaroth
+ (license 50)
+
+2008-06-19 19:55 +0000 [r124066] Brett Bryant <bbryant@digium.com>
+
+ * main/utils.c: Merge revision 124064 from trunk. Add errors that
+ report any locks held by threads when they are being closed.
+
+2008-06-19 16:58 +0000 [r123710-123930] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c: Change informative messages to use the _multiple
+ variant when multiple formats are possible. (Closes issue #12848)
+ Reported by klaus3000
+
+ * build_tools/strip_nonapi: Only process 40 arguments (20 files) at
+ once with xargs, because some older shells may force xargs to
+ separate on an odd boundary. (Closes issue #12883) Reported by
+ Nik Soggia
+
+ * configs/sip.conf.sample: Correct description of notifyringing
+ option. (Closes issue #12890) Reported by gminet
+
+ * main/asterisk.c: The RDTSC instruction was introduced on the
+ Pentium line of microprocessors, and is not compatible with
+ certain 586 clones, like Cyrix. Hence, asking for i386
+ compatibility was always incorrect. See
+ http://en.wikipedia.org/wiki/RDTSC (Closes issue #12886) Reported
+ by tecnoxarxa
+
+ * main/say.c, doc/lang (added), doc/lang/hebrew.ods (added): Add
+ support for saying numbers in Hebrew. (closes issue #11662)
+ Reported by: greenfieldtech Patches: say.c.patch-12042008
+ uploaded by greenfieldtech (license 369) Hebrew-Sounds.ods
+ uploaded by greenfieldtech (with signficant changes to the
+ spreadsheet by me)
+
+ * pbx/pbx_spool.c: Set the variables top-down, so that if a script
+ sets a variable more than once, the last one will take
+ precedence. (closes issue #12673) Reported by: phber Patches:
+ 20080519__bug12673.diff.txt uploaded by Corydon76 (license 14)
+
+2008-06-17 20:26 +0000 [r123485] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Make chan_sip build under dev mode with
+ compilers >= GCC 4.2 Thanks to jpeeler for alerting me of this
+
+2008-06-17 18:56 +0000 [r123391] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Fix 3 more places where not locking the
+ structure could cause the wrong lock to be unlocked. (Closes
+ issue #12795)
+
+2008-06-17 18:09 +0000 [r123274-123333] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Cisco BTS sends SIP responses with a tab
+ between the Cseq number and SIP request method in the Cseq:
+ header. Asterisk did not handle this properly, but with this
+ patch, all is well. (closes issue #12834) Reported by: tobias_e
+ Patches: 12834.patch uploaded by putnopvut (license 60) Tested
+ by: tobias_e
+
+ * apps/app_queue.c: davidw pointed out that the holdtime
+ calculation used by app_queue does not use "boxcar" filtering as
+ the comments say. The term "boxcar" means that the number of
+ samples used to calculate stays constant, with new samples
+ replacing the oldest ones. The queue holdtime calculation uses
+ all holdtime samples collected since the queue was loaded, so the
+ comment has been changed to be accurate. (closes issue #12781)
+ Reported by: davidw
+
+2008-06-17 15:48 +0000 [r123271] Russell Bryant <russell@digium.com>
+
+ * main/astobj2.c: Fix a memory leak in astobj2 that was pointed out
+ by seanbright. When a container got destroyed, the underlying
+ bucket list entry for each object that was in the container at
+ that time did not get free'd.
+
+2008-06-16 19:50 +0000 [r123110-123113] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_mgcp.c, channels/chan_dahdi.c,
+ channels/chan_skinny.c, channels/chan_h323.c,
+ channels/chan_iax2.c: Port "hasvoicemail" change from SIP to
+ other channel drivers
+
+ * channels/chan_sip.c: People expect that if "hasvoicemail" is set
+ in users.conf, even if "mailbox" isn't set, that SIP will detect
+ a mailbox. (closes issue #12855) Reported by: PLL Patches:
+ 20080614__bug12855__2.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: PLL
+
+2008-06-16 12:31 +0000 [r122869-122919] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Only compare the first 15 characters so that
+ even if the charset is specified we still accept it as SDP.
+ (closes issue #12803) Reported by: lanzaandrea Patches:
+ chan_sip.c.diff uploaded by lanzaandrea (license 496)
+
+ * channels/chan_sip.c: Don't send a BYE on a dialog that is already
+ gone during a REFER. (closes issue #12865) Reported by: flefoll
+ Patches: chan_sip.c.br14.121495.patch-ALREADYGONE uploaded by
+ flefoll (license 244)
+
+2008-06-13 21:44 +0000 [r122713] Mark Michelson <mmichelson@digium.com>
+
+ * main/autoservice.c: Short circuit the loop in autoservice_run if
+ there are no channels to poll. If we continued, then the result
+ would be calling poll() with a NULL pollfd array. While this is
+ fine with POSIX's poll(2) system call, those who use Asterisk's
+ internal poll mechanism (Darwin systems) would have a failed
+ assertion occur when poll is called. (related to issue #10342)
+
+2008-06-13 18:57 +0000 [r122663] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/dahdi_compat.h, res/res_musiconhold.c: fixed
+ dahdi compatability header from assuming either dahdi or zaptel
+ is installed (may not have either)
+
+2008-06-13 17:45 +0000 [r122617] Terry Wilson <twilson@digium.com>
+
+ * apps/app_dial.c: Remove extra option from previous solution
+ attempt
+
+2008-06-13 17:36 +0000 [r122613] Jeff Peeler <jpeeler@digium.com>
+
+ * configure, configure.ac: (closes issue #12846) Reported by:
+ Netview Tested by: jpeeler Use correct location to search for
+ tonezone.
+
+2008-06-13 16:29 +0000 [r122589] Terry Wilson <twilson@digium.com>
+
+ * apps/app_dial.c, res/res_features.c: This should fix the behavior
+ of the 'T' dial feature being passed incorrectly to the
+ transferee when builtin_atxfers are used. Also, doing a
+ builtin_atxfer to parking was broken and is fixed here as well.
+ (closes issue #11898) Reported by: sergee Tested by: otherwiseguy
+
+2008-06-12 19:08 +0000 [r122314] Jeff Peeler <jpeeler@digium.com>
+
+ * main/indications.c, include/asterisk/dahdi_compat.h (added),
+ main/loader.c, main/channel.c, channels/chan_dahdi.c (added),
+ configure, apps/app_zapscan.c (removed), apps/app_zapras.c
+ (removed), main/app.c, include/asterisk/options.h,
+ apps/app_rpt.c, channels/chan_mgcp.c, apps/app_read.c,
+ channels/chan_zap.c (removed), apps/app_page.c,
+ include/asterisk/indications.h, apps/app_dahdiras.c (added),
+ configure.ac, apps/app_disa.c, include/asterisk/channel.h,
+ apps/app_getcpeid.c, apps/app_queue.c, apps/app_zapbarge.c
+ (removed), channels/chan_misdn.c, apps/app_flash.c,
+ build_tools/menuselect-deps.in, funcs/func_channel.c,
+ main/file.c, res/snmp/agent.c, contrib/utils/zones2indications.c,
+ codecs/codec_dahdi.c (added), res/res_indications.c,
+ pbx/pbx_config.c, makeopts.in, apps/app_chanspy.c,
+ main/asterisk.c, apps/app_dahdibarge.c (added),
+ apps/app_meetme.c, include/asterisk/autoconfig.h.in,
+ apps/app_dahdiscan.c (added), acinclude.m4,
+ res/res_musiconhold.c, codecs/codec_zap.c (removed),
+ channels/chan_iax2.c: Adds DAHDI support alongside Zaptel. DAHDI
+ usage favored, but all Zap stuff should continue working. Release
+ announcement to follow.
+
+2008-06-12 18:50 +0000 [r122311] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Properly play a holdtime message if the
+ announce-holdtime option is set to "once." (closes issue #12842)
+ Reported by: ramonpeek Patches: patch001.diff uploaded by
+ ramonpeek (license 266)
+
+2008-06-12 18:22 +0000 [r122259] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Fix some race conditions that cause
+ ast_assert() to report that chan_iax2 tried to remove an entry
+ that wasn't in the scheduler
+
+2008-06-12 15:46 +0000 [r122208] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_parkandannounce.c, res/res_features.c: (closes issue
+ #12193) Reported by: davidw Patch by: Corydon76, modified by me
+ to work properly with ParkAndAnnounce app
+
+2008-06-12 15:18 +0000 [r122130-122137] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c: Flipflop the sections for two options, since
+ the section for 'X' (exit context) may otherwise absorb
+ keypresses meant for 's' (admin/user menu). (closes issue #12836)
+ Reported by: blitzrage Patches: 20080611__bug12836.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: blitzrage
+
+ * main/channel.c: Occasionally, the alertpipe loses its nonblocking
+ status, so detect and correct that situation before it causes a
+ deadlock. (Reported and tested by ctooley via #asterisk-dev)
+
+2008-06-12 14:51 +0000 [r122127] Steve Murphy <murf@digium.com>
+
+ * main/cdr.c, apps/app_forkcdr.c: Arkadia tried to warn me, but the
+ code added to ast_cdr_busy, _failed, and _noanswer was redundant.
+ Didn't spot it until I was resolving conflicts in trunk. Ugh.
+ Redundant code removed. It wasn't harmful. Just dumb.
+
+2008-06-12 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.21 released.
+
+2008-06-06 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.21-rc2 released.
+
+2008-06-05 18:03 +0000 [r120731-120735] Russell Bryant <russell@digium.com>
+
+ * UPGRADE-1.2.txt: fix filename
+
+ * UPGRADE-1.2.txt (added), UPGRADE.txt: Add the UPGRADE.txt file
+ from Asterisk 1.2, for handy reference.
+
+2008-06-05 16:56 +0000 [r120675] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * res/res_jabber.c: Ignore appended resource when comparing JIDs.
+
+2008-06-05 16:38 +0000 [r120671] Russell Bryant <russell@digium.com>
+
+ * doc/smdi.txt, res/res_smdi.c: It turns out that searching on the
+ forwarding station isn't very useful for most people, so pull in
+ the changes that allow searching for SMDI messages based on other
+ components of the SMDI message. Also, update the SMDI
+ documentation.
+
+2008-06-04 22:05 +0000 [r120513] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Make sure that the string we set will survive
+ the unref of the queue member. Thanks to Russell, who pointed
+ this out.
+
+2008-06-04 18:35 +0000 [r120425] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_zap.c: If we fail to setup the PRI request channel,
+ don't continue, exit with an error. (closes issue #11989)
+ Reported by: Corydon76 Patches: 20080213__zap_memleak.diff.txt
+ uploaded by Corydon76 (license 14)
+
+2008-06-04 16:26 +0000 [r120371] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_config.c: Make the "dialplan remove include" CLI command
+ actually work. Also, tweak some formatting, and make the success
+ message a little bit more clear. (closes AST-52)
+
+2008-06-04 14:11 +0000 [r120285] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Tab completion when removing a member should
+ give the member's interface, not the name, since the interface is
+ what is expected for the command. (closes issue #12783) Reported
+ by: davevg
+
+2008-06-04 13:31 +0000 [r120282] Joshua Colp <jcolp@digium.com>
+
+ * main/pbx.c, pbx/pbx_config.c: Fix a log message and add a message
+ for when the dialplan is done reloading. (closes issue #12716)
+ Reported by: chappell Patches: dialplan_reload_2.diff uploaded by
+ chappell (license 8)
+
+2008-06-03 22:41 +0000 [r120226] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_loopback.c: Due to incorrect use of the
+ AST_LIST_INSERT_HEAD() macro the loopback switch cannot perform
+ any translation on the extension number before searching for it
+ in the target context. (closes issue #12473) Reported by:
+ chappell Patches: pbx_loopback.c.diff uploaded by chappell
+ (license 8)
+
+2008-06-03 22:15 +0000 [r120173] Jeff Peeler <jpeeler@digium.com>
+
+ * main/config.c: (closes issue #11594) Reported by: yem Tested by:
+ yem This change decreases the buffer size allocated on the stack
+ substantially in config_text_file_load when LOW_MEMORY is turned
+ on. This change combined with the fix from revision 117462
+ (making mkintf not copy the zt_chan_conf structure) was enough to
+ prevent the crash.
+
+2008-06-03 21:34 +0000 [r120168] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Fix another place where peer->callno could
+ change at a very bad time, and also fix a place where a peer was
+ used after the reference was released. (inspired by rev 120001)
+
+2008-06-03 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.21-rc1 released.
+
+2008-06-03 18:23 +0000 [r120001-120061] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c: When listing the manager users, managers in
+ users.conf are not shown, even though they are allowed to
+ connect. (closes issue #12594) Reported by: bkruse Patches:
+ 12594-managerusers-2.diff uploaded by qwell (license 4) Tested
+ by: bkruse
+
+ * channels/chan_iax2.c: Save the callno when we're poking, because
+ our peer structure could change during deadlock avoidance (and
+ thus we unlock the wrong callno, causing a cascade failure).
+ (closes issue #12717) Reported by: gewfie Patches:
+ 20080525__bug12717.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: gewfie
+
+2008-06-03 15:26 +0000 [r119929-119966] Steve Murphy <murf@digium.com>
+
+ * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
+ pbx/ael/ael-test/ref.ael-vtest13,
+ pbx/ael/ael-test/ref.ael-vtest17,
+ pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
+ pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test5,
+ pbx/ael/ael-test/ref.ael-test15: Updated the regressions on AEL.
+ Hadn't updated this for the changes I made to preserve ${EXTEN}
+ in switches, which affected several tests because it adds extra
+ priorities, and at least one needed to be updated because of the
+ removal of the empty extension warning message.
+
+ * pbx/pbx_ael.c: as per
+ http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html,
+ which is a message from Philipp Kempgen, requesting that the
+ WARNING that an extension is empty be reduced to a NOTICE or
+ less, as empty extensions are syntactically possible, and no big
+ deal. With which I agree, and have removed that WARNING message
+ entirely. I think it is not necessary to see this message. It
+ didn't state that a NoOp() was inserted automatically on your
+ behalf, and really, as users, who cares? Why freak out dialplan
+ writers with unnecessary warnings? The details of the
+ machinations a compiler goes thru to produce working assembly
+ code is of little interest to most programmers-- we will follow
+ the unix principal of doing our work silently.
+
+2008-06-03 14:46 +0000 [r119926] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Treat ECONNREFUSED as an error that will
+ stop further retransmissions. (issue #AST-58, patch from
+ Switchvox)
+
+2008-06-02 20:08 +0000 [r119742-119838] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Revert a change made for issue #12479. This
+ change caused a regression such that a dial string such as
+ (IAX2/foo) did not automatically fall back to dialing the 's'
+ extension anymore. (closes issue #12770) Reported by: dagmoller
+
+ * main/manager.c: Improve CLI command blacklist checking for the
+ command manager action. Previously, it did not handle case or
+ whitespace properly. This made it possible for blacklisted
+ commands to get executed anyway. (closes issue #12765)
+
+2008-06-02 14:32 +0000 [r119740] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * channels/chan_gtalk.c, res/res_jabber.c: Do not link the guest
+ account with any configured XMPP client (in jabber.conf). The
+ actual connection is made when a call comes in Asterisk. Fix the
+ ast_aji_get_client function that was not able to retrieve an XMPP
+ client from its JID. (closes issue #12085) Reported by: junky
+ Tested by: phsultan
+
+2008-06-02 12:30 +0000 [r119687] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Even of the first PING or LAGRQ doesn't get
+ sent because it comes up too soon, make sure to reschedule so it
+ gets sent later.
+
+2008-06-02 09:29 +0000 [r119585-119636] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.c: fixed compile issue when dev-mode is
+ enabled
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h: Added
+ counter for unhandled_bmsg Print, this prevents the logs to be
+ flooded to fast and save CPU in this error scenario. Added
+ 'last_used' element to bc structure, when a bchannel changes from
+ used to free this exact time will be marked in last_used. When a
+ new channel is requested the find_free_chan function will check
+ if the new empty channel was used within the last second, if yes
+ it will search for the next channel, if no it will return this
+ channel. This simple mechanism has prooven to prevent race
+ conditions where the NT and TE tried to allocate the exact same
+ channel at the same time (RELEASE cause: 44).
+
+2008-06-02 01:06 +0000 [r119530-119533] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Change a debug message to an actual debug
+ message
+
+ * apps/app_dial.c: Fix another typo in documentation
+
+2008-06-01 20:47 +0000 [r119478] Michiel van Baak <michiel@vanbaak.info>
+
+ * apps/app_dial.c: small typo fix 'retires' => 'retries'
+
+2008-05-30 21:17 +0000 [r119404] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c: When joinempty=strict, it only failed on join
+ if there were busy members. If all members were logged out OR
+ paused, then it (incorrectly) let callers join the queue. (closes
+ issue #12451) Reported by: davidw
+
+2008-05-30 19:46 +0000 [r119354] Joshua Colp <jcolp@digium.com>
+
+ * main/autoservice.c: Fix a bug I found while testing for another
+ issue.
+
+2008-05-30 16:44 +0000 [r119301] Michiel van Baak <michiel@vanbaak.info>
+
+ * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk,
+ contrib/init.d/rc.debian.asterisk,
+ contrib/init.d/rc.mandrake.asterisk,
+ contrib/init.d/rc.redhat.asterisk,
+ contrib/init.d/rc.gentoo.asterisk,
+ contrib/init.d/rc.slackware.asterisk: dont use a bashism way to
+ check the $VERSION variable. The rc/init.d scripts, and
+ safe_asterisk work on normal sh now again. Tested on: OpenBSD 4.2
+ (me) Debian etch (me) Ubuntu Hardy (me and loloski) FC9 (loloski)
+ (closes issue #12687) Reported by: loloski Patches:
+ 20080529-12687-safe_asterisk-fixversion.diff.txt uploaded by
+ mvanbaak (license 7) Tested by: loloski, mvanbaak
+
+2008-05-30 12:55 +0000 [r119076-119238] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 119237 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30
+ May 2008) | 7 lines - Instead of only enforcing destination call
+ number checking on an ACK, check all full frames except for PING
+ and LAGRQ, which may be sent by older versions too quickly to
+ contain the destination call number. (As suggested by Tim Panton
+ on the asterisk-dev list) - Merge changes from
+ team/russell/iax2-frame-race, which prevents PING and LAGRQ from
+ being sent before the destination call number is known. ........
+
+ * main/autoservice.c: Fix a race condition in channel autoservice.
+ There was still a small window of opportunity for a DTMF frame,
+ or some other deferred frame type, to come in and get dropped.
+ (closes issue #12656) (closes issue #12656) Reported by: dimas
+ Patches: v3-12656.patch uploaded by dimas (license 88) -- with
+ some modifications by me
+
+ * include/asterisk/audiohook.h: Oddly enough, all of the contents
+ of audiohook.h were in there twice. I have removed the second
+ copy.
+
+2008-05-29 20:24 +0000 [r119071] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_zap.c: Call waiting tone occurs too often, because
+ it's getting serviced by both subchannels. (closes issue #11354)
+ Reported by: cahen Patches: 20080512__bug11354.diff.txt uploaded
+ by Corydon76 (license 14)
+
+2008-05-29 19:04 +0000 [r118956-119012] Russell Bryant <russell@digium.com>
+
+ * apps/app_milliwatt.c: - Fix a typo in the argument to Playtones -
+ use ast_safe_sleep() instead of calling the wait application
+ (thanks to tilghman for pointing these out!)
+
+ * /, channels/chan_iax2.c: Merged revisions 119008 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r119008 | russell | 2008-05-29 13:45:21 -0500 (Thu, 29
+ May 2008) | 7 lines Merge changes from
+ team/russell/iax2-another-fix-to-the-fix As described in the
+ following post to the asterisk-dev mailing list, only enforce
+ destination call numbers when processing an ACK.
+ http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html
+ (closes issue #12631) ........
+
+ * apps/app_milliwatt.c: - Mark app_milliwatt dependent on
+ res_indications (thanks to jsmith) - fix a typo in a log message
+ (thanks to qwell)
+
+ * apps/app_milliwatt.c: Change milliwatt to use the proper tone by
+ default (1004 Hz) instead of 1000 Hz. An option is there to use
+ 1000 Hz for anyone that might want it.
+
+2008-05-29 17:33 +0000 [r118953-118954] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/lock.h: Define also when not DEBUG_THREADS
+
+ * channels/chan_mgcp.c, channels/chan_zap.c, channels/chan_sip.c,
+ channels/chan_agent.c, channels/chan_alsa.c, main/utils.c,
+ include/asterisk/lock.h, channels/chan_iax2.c: Add some debugging
+ code that ensures that when we do deadlock avoidance, we don't
+ lose the information about how a lock was originally acquired.
+
+2008-05-29 00:25 +0000 [r118858] Steve Murphy <murf@digium.com>
+
+ * main/cdr.c, apps/app_forkcdr.c: (closes issue #10668) (closes
+ issue #11721) (closes issue #12726) Reported by: arkadia Tested
+ by: murf These changes: 1. revert the changes made via bug 10668;
+ I should have known that such changes, even tho they made sense
+ at the time, seemed like an omission, etc, were actually integral
+ to the CDR system via forkCDR. It makes sense to me now that
+ forkCDR didn't natively end any CDR's, but rather depended on
+ natively closing them all at hangup time via traversing and
+ closing them all, whether locked or not. I still don't completely
+ understand the benefits of setvar and answer operating on locked
+ cdrs, but I've seen enough to revert those changes also, and stop
+ messing up users who depended on that behavior. bug 12726 found
+ reverting the changes fixed his changes, and after a long review
+ and working on forkCDR, I can see why. 2. Apply the suggested
+ enhancements proposed in 10668, but in a completely compatible
+ way. ForkCDR will behave exactly as before, but now has new
+ options that will allow some actions to be taken that will
+ slightly modify the outcome and side-effects of forkCDR. Based on
+ conversations I've had with various people, these small tweaks
+ will allow some users to get the behavior they need. For
+ instance, users executing forkCDR in an AGI script will find the
+ answer time set, and DISPOSITION set, a situation not covered
+ when the routines were first written. 3. A small problem in the
+ cdr serializer would output answer and end times even when they
+ were not set. This is now fixed.
+
+2008-05-28 16:10 +0000 [r118716] Brett Bryant <bbryant@digium.com>
+
+ * channels/chan_iax2.c: merge revision 118702 from trunk to 1.4 --
+ Fixes a bug in chan_iax that uses send_command to poke a peer
+ while a channel is unlocked in some cases, and because it can
+ cause seemingly random failures could be related to some bugs in
+ the tracker...
+
+2008-05-28 14:23 +0000 [r118558-118646] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add an
+ option to use the source IP address of RTP as the destination IP
+ address of UDPTL when a specific option is enabled. If the remote
+ side is properly configured (ports forwarded) then UDPTL will
+ flow. (closes issue #10417) Reported by: cstadlmann
+
+ * channels/chan_sip.c: Fix an issue where codec preferences were
+ not set on dialogs that were not authenticated via a user or peer
+ and allow framing to work without rtpmap in the SDP. (closes
+ issue #12501) Reported by: slimey
+
+2008-05-27 19:15 +0000 [r118551] Tilghman Lesher <tlesher@digium.com>
+
+ * main/cli.c: When showing an error message for a command, don't
+ shorten the command output, as it tends to confuse the user (it's
+ fine for suggesting other commands, however). Reported by:
+ seanbright (on #asterisk-dev) Fixed by: me
+
+2008-05-27 19:07 +0000 [r118509] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c: Russell noted to me that in the case that
+ separate threads use their own addressing system, the fix I made
+ for issue 12376 does not guarantee uniqueness to the datastores'
+ uids. Though I know of no system that works this way, I am going
+ to change this right now to prevent trying to track down some
+ future bug that may occur and cause untold hours of debugging
+ time to track down. The change involves using a global counter
+ which increases with each new chanspy_ds which is created. This
+ guarantees uniqueness.
+
+2008-05-27 18:58 +0000 [r118465] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c: NULL character should terminate only commands
+ back to the core, not log messages to the console. (closes issue
+ #12731) Reported by: seanbright Patches:
+ 20080527__bug12731.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: seanbright
+
+2008-05-27 17:17 +0000 [r118416] Michiel van Baak <michiel@vanbaak.info>
+
+ * apps/app_voicemail.c: small update to the g() option of
+ app_voicemail to note that gain changes only work on zap channels
+ right now. issue #12578 shows it's not clear right now.
+
+2008-05-27 16:38 +0000 [r118365] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c: Add a unique id to the datastore allocated in
+ app_chanspy since it is possible that multiple spies may be
+ listening to the same channel. (closes issue #12376) Reported by:
+ DougUDI Patches: 12376_chanspy_uid.diff uploaded by putnopvut
+ (license 60) Tested by: destiny6628 (closes issue #12243)
+ Reported by: atis
+
+2008-05-27 15:45 +0000 [r118358] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/queues.conf.sample: Add a note that pbx_config.so is
+ needed for Local channels. (Closes issue #12671)
+
+2008-05-25 16:02 +0000 [r118251] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Realtime flag affects construction in
+ multiple ways, so consulting whether rtcachefriends was set was
+ done too soon (needed to be done inside build_peer, not just as a
+ flag to build_peer). Also, fullcontact needed to be
+ reconstructed, because realtime separates the embedded ';' into
+ multiple fields. (closes issue #12722) Reported by: barthpbx
+ Patches: 20080525__bug12722.diff.txt uploaded by Corydon76
+ (license 14) Tested by: barthpbx (Much of the discussion happened
+ on #asterisk-dev for diagnosing this issue)
+
+2008-05-23 21:21 +0000 [r118163] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_zap.c: Fix a few things I missed to ensure
+ zt_chan_conf structure is not modified in mkintf
+
+2008-05-23 13:18 +0000 [r118052-118055] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/utils.h: Add format type checking for recently
+ de-inlined function
+
+ * doc/cli.txt (added), doc/00README.1st: Add information on using
+ the Asterisk console, including tab command line completion.
+ (Closes issue #12681)
+
+2008-05-23 12:30 +0000 [r118048] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/utils.h, main/utils.c: Don't declare a function
+ that takes variable arguments as inline, because it's not valid,
+ and on some compilers, will emit a warning.
+ http://gcc.gnu.org/onlinedocs/gcc/Inline.html#Inline (closes
+ issue #12289) Reported by: francesco_r Patches by Tilghman, final
+ patch by me
+
+2008-05-22 18:53 +0000 [r117809-117899] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c: Also remove preamble from asynchronous events
+ (reported by jsmith on #asterisk-dev)
+
+ * funcs/func_realtime.c: Take into account the length of delimiters
+ when calculating result string length. (closes issue #12696)
+ Reported by: adomjan Patches: func_realtime.c-longdelimiter.patch
+ uploaded by adomjan (license 487)
+
+2008-05-21 20:11 +0000 [r117582] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_zap.c: Ensure that passed in zt_chan_conf structure
+ is not modified in mkintf.
+
+2008-05-21 19:38 +0000 [r117574] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Apply the autoframing setting to dialogs
+ that do not get matched against a user or peer.
+
+2008-05-21 18:44 +0000 [r117519-117523] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_spool.c: Revert accidental commit of the last change
+
+ * main/asterisk.c, pbx/pbx_spool.c: Strip the preamble from the
+ output also when -rx is not being used (Related to issue #12702)
+
+2008-05-21 18:28 +0000 [r117479-117514] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c: Don't filter the magic character in the network
+ verboser. It gets filtered once it reaches the client. (related
+ to issue #12702, pointed out by tilghman)
+
+ * main/asterisk.c, pbx/pbx_gtkconsole.c: 1) Don't print the verbose
+ marker in front of every message from ast_verbose() being sent to
+ remote consoles. 2) Fix pbx_gtkconsole to filter out the verbose
+ marker. (related to issue #12702)
+
+ * main/asterisk.c: Don't display the verbose marker for calls to
+ ast_verbose() that do not include a VERBOSE_PREFIX in front of
+ the message. (closes issue #12702) Reported by: johnlange Patched
+ by me
+
+2008-05-21 16:58 +0000 [r117462] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_zap.c: Pass a pointer for the conf parameter to the
+ function mkintf rather than the whole zt_chan_conf structure.
+
+2008-05-20 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.20 released.
+
+2008-05-14 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.20-rc3 released.
+
+2008-05-14 12:51 +0000 [r116230] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Accept text messages even with Content-Type:
+ text/plain;charset=Södermanländska
+
+2008-05-13 23:47 +0000 [r116088] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, include/asterisk/lock.h: A change to the way
+ channel locks are handled when DEBUG_CHANNEL_LOCKS is defined.
+ After debugging a deadlock, it was noticed that when
+ DEBUG_CHANNEL_LOCKS is enabled in menuselect, the actual origin
+ of channel locks is obscured by the fact that all channel locks
+ appear to happen in the function ast_channel_lock(). This code
+ change redefines ast_channel_lock to be a macro which maps to
+ __ast_channel_lock(), which then relays the proper file name,
+ line number, and function name information to the core lock
+ functions so that this information will be displayed in the case
+ that there is some sort of locking error or core show locks is
+ issued.
+
+2008-05-13 21:17 +0000 [r115990-116038] Russell Bryant <russell@digium.com>
+
+ * channels/chan_local.c: Fix a deadlock involving channel
+ autoservice and chan_local that was debugged and fixed by
+ mmichelson and me. We observed a system that had a bunch of
+ threads stuck in ast_autoservice_stop(). The reason these threads
+ were waiting around is because this function waits to ensure that
+ the channel list in the autoservice thread gets rebuilt before
+ the stop() function returns. However, the autoservice thread was
+ also locked, so the autoservice channel list was never getting
+ rebuilt. The autoservice thread was stuck waiting for the channel
+ lock on a local channel. However, the local channel was locked by
+ a thread that was stuck in the autoservice stop function. It
+ turned out that the issue came down to the local_queue_frame()
+ function in chan_local. This function assumed that one of the
+ channels passed in as an argument was locked when called.
+ However, that was not always the case. There were multiple cases
+ in which this channel was not locked when the function was
+ called. We fixed up chan_local to indicate to this function
+ whether this channel was locked or not. The previous assumption
+ had caused local_queue_frame() to improperly return with the
+ channel locked, where it would then never get unlocked. (closes
+ issue #12584) (related to issue #12603)
+
+ * main/autoservice.c: Fix an issue that I noticed in autoservice
+ while mmichelson and I were debugging a different problem. I
+ noticed that it was theoretically possible for two threads to
+ attempt to start the autoservice thread at the same time. This
+ change makes the process of starting the autoservice thread,
+ thread-safe.
+
+2008-05-13 20:28 +0000 [r115944] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_alsa.c: Use the right flag to open the audio in
+ non-blocking. (closes issue #12616) Reported by:
+ nicklewisdigiumuser
+
+2008-05-13 18:36 +0000 [r115884] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c: If the socket dies (read returns 0=EOF), return
+ immediately. (Closes issue #12637)
+
+2008-05-12 17:51 +0000 [r115735] Mark Michelson <mmichelson@digium.com>
+
+ * main/utils.c: If a thread holds no locks, do not print any
+ information on the thread when issuing a core show locks command.
+ This will help to de-clutter output somewhat. Russell said it
+ would be fine to place this improvement in the 1.4 branch, so
+ that's why it's going here too.
+
+2008-05-09 16:34 +0000 [r115579] Joshua Colp <jcolp@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Improve res_ninit and res_ndestroy autoconf logic on the Darwin
+ platform.
+
+2008-05-08 19:19 +0000 [r115545-115568] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Remove debug output.
+
+ * /, channels/chan_iax2.c: Merged revisions 115564 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08
+ May 2008) | 25 lines Fix a race condition that bbryant just found
+ while doing some IAX2 testing. He was running Asterisk trunk
+ running IAX2 calls through a few Asterisk boxes, however, the
+ audio was extremely choppy. We looked at a packet trace and saw a
+ storm of INVAL and VNAK frames being sent from one box to
+ another. It turned out that what had happened was that one box
+ tried to send a CONTROL frame before the 3 way handshake had
+ completed. So, that frame did not include the destination call
+ number, because it didn't have it yet. Part of our recent work
+ for security issues included an additional check to ensure that
+ frames that are supposed to include the destination call number
+ have the correct one. This caused the frame to be rejected with
+ an INVAL. The frame would get retransmitted for forever, rejected
+ every time ... This race condition exists in all versions that
+ got the security changes, in theory. However, it is really only
+ likely that this would cause a problem in Asterisk trunk. There
+ was a control frame being sent (SRCUPDATE) at the _very_
+ beginning of the call, which does not exist in 1.2 or 1.4.
+ However, I am fixing all versions that could potentially be
+ affected by the introduced race condition. These changes are what
+ bbryant and I came up with to fix the issue. Instead of simply
+ dropping control frames that get sent before the handshake is
+ complete, the code attempts to wait a little while, since in most
+ cases, the handshake will complete very quickly. If it doesn't
+ complete after yielding for a little while, then the frame gets
+ dropped. ........
+
+ * channels/chan_sip.c: Don't give up on attempting an outbound
+ registration if we receive a 408 Timeout. (closes issue #12323)
+
+ * contrib/scripts/postgres_cdr.sql (removed): remove
+ postgres_cdr.sql, as the CDR schema is in realtime_pgsql.sql, as
+ well (closes issue #9676)
+
+ * contrib/init.d/rc.debian.asterisk: Don't exit the script if
+ Asterisk is not running. (closes issue #12611)
+
+ * main/pbx.c: Don't use a channel before checking for channel
+ allocation failure. (closes issue #12609) Reported by: edantie
+
+ * contrib/init.d/rc.debian.asterisk: Use the same method for
+ executing Asterisk as the rest of the script. (closes issue
+ #12611) Reported by: b_plessis
+
+2008-05-07 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.20-rc2 released.
+
+2008-05-07 18:17 +0000 [r115512-115517] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Track peer references when stored in the
+ sip_pvt struct as the peer related to a qualify ping or a
+ subscription. This fixes some realtime related crashes. (closes
+ issue #12588) (closes issue #12555)
+
+2008-05-06 19:55 +0000 [r115418-115422] Jason Parker <jparker@digium.com>
+
+ * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 115421
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r115421 | qwell | 2008-05-06 14:54:57 -0500 (Tue, 06 May 2008) |
+ 7 lines read requires an argument on some non-bash shells (closes
+ issue #12593) Reported by: bkruse Patches:
+ getilbc.sh_12593_v1.diff uploaded by bkruse (license 132)
+ ........
+
+ * res/res_musiconhold.c: Switch to using ast_random() rather than
+ just rand(). This does not fix the bug reported, but I believe it
+ is correct. (from issue #12446) Patches: bug_12446.diff uploaded
+ by snuffy (license 35)
+
+2008-05-06 19:31 +0000 [r115415] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c: Don't print the terminating NUL. (Closes issue
+ #12589)
+
+2008-05-06 13:54 +0000 [r115341] Joshua Colp <jcolp@digium.com>
+
+ * configure, configure.ac: Add in missing argument.
+
+2008-05-05 22:50 +0000 [r115333] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, main/logger.c: Separate verbose output from CLI
+ output, by using a preamble. (closes issue #12402) Reported by:
+ Corydon76 Patches: 20080410__no_verbose_in_rx_output.diff.txt
+ uploaded by Corydon76 (license 14)
+ 20080501__no_verbose_in_rx_output__1.4.diff.txt uploaded by
+ Corydon76 (license 14)
+
+2008-05-05 22:10 +0000 [r115327] Joshua Colp <jcolp@digium.com>
+
+ * build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
+ configure.ac: Make sure that either the main speex library
+ contains preprocess functions or that speexdsp does. If both fail
+ then speex stuff can not be built.
+
+2008-05-05 21:41 +0000 [r115320] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Don't consider a caller "handled" until the
+ caller is bridged with a queue member. There was too much of an
+ opportunity for the member to hang up (either during a delay,
+ announcement, or overly long agi) between the time that he
+ answered the phone and the time when he actually was bridged with
+ the caller. The consequence of this was that if the member hung
+ up in that interval, then proper abandonment details would not be
+ noted in the queue log if the caller were to hang up at any point
+ after the member hangup. (closes issue #12561) Reported by:
+ ablackthorn
+
+2008-05-05 20:17 +0000 [r115308-115312] Tilghman Lesher <tlesher@digium.com>
+
+ * Makefile: Reverse order, such that user configs override default
+ selections
+
+ * include/asterisk/res_odbc.h: Err, the documentation on the return
+ value of ast_odbc_backslash_is_escape is exactly backwards.
+
+2008-05-05 19:49 +0000 [r115297-115304] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Avoid putting opaque="" in Digest
+ authentication. This patch came from switchvox. It fixes
+ authentication with Primus in Canada, and has been in use for a
+ very long time without causing problems with any other providers.
+ (closes issue AST-36)
+
+2008-05-05 03:22 +0000 [r115285] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk,
+ contrib/init.d/rc.debian.asterisk,
+ contrib/init.d/rc.mandrake.asterisk,
+ contrib/init.d/rc.redhat.asterisk,
+ contrib/init.d/rc.gentoo.asterisk,
+ contrib/init.d/rc.slackware.asterisk: When starting Asterisk, bug
+ out if Asterisk is already running. (closes issue #12525)
+ Reported by: explidous Patches: 20080428__bug12525.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: mvanbaak
+
+2008-05-04 02:09 +0000 [r115276-115282] Joshua Colp <jcolp@digium.com>
+
+ * configure, acinclude.m4: Expand the test function for GCC
+ attributes so that more complex attributes are properly
+ recognized.
+
+ * include/asterisk/compiler.h: For my next trick I will make these
+ work with what our autoconf header file gives us.
+
+ * configure, acinclude.m4: Treat warnings as errors when checking
+ if a GCC attribute exists. We have to do this as GCC will just
+ ignore the attribute and pop up a warning, it won't actually fail
+ to compile.
+
+2008-05-02 20:25 +0000 [r115257] Brett Bryant <bbryant@digium.com>
+
+ * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac, CHANGES: Add new "pri show version" command to show
+ the libpri version for support reasons.
+
+2008-05-02 14:28 +0000 [r115196] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/sched.h: Clarify a comment that was, well, just
+ wrong. It turns out that ignoring the way that macros expand.
+ Instead, I have clarified in the comment why the macro will work
+ even if the scheduler id for the task to be deleted changes
+ during the execution of the macro.
+
+2008-05-01 23:20 +0000 [r115017-115102] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/res_odbc.h: Change the comment of deprecated to
+ an actual compiler deprecation
+
+ * main/utils.c: '#' is another reserved character for URIs that
+ also needs to be escaped. (closes issue #10543) Reported by:
+ blitzrage Patches: 20080418__bug10543.diff.txt uploaded by
+ Corydon76 (license 14)
+
+2008-05-01 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.20-rc1 released.
+
+2008-04-30 16:30 +0000 [r114891] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/dlinkedlists.h (added), channels/chan_iax2.c:
+ Merge changes from team/russell/iax2_find_callno and
+ iax2_find_callno_1.4 These changes address a critical performance
+ issue introduced in the latest release. The fix for the latest
+ security issue included a change that made Asterisk randomly
+ choose call numbers to make them more difficult to guess by
+ attackers. However, due to some inefficient (this is by far, an
+ understatement) code, when Asterisk chose high call numbers,
+ chan_iax2 became unusable after just a small number of calls. On
+ a small embedded platform, it would not be able to handle a
+ single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run
+ more than about 16 IAX2 channels. Ouch. These changes address
+ some performance issues of the find_callno() function that have
+ bothered me for a very long time. On every incoming media frame,
+ it iterated through every possible call number trying to find a
+ matching active call. This involved a mutex lock and unlock for
+ each call number checked. So, if the random call number chosen
+ was 20000, then every media frame would cause 20000 locks and
+ unlocks. Previously, this problem was not as obvious since
+ Asterisk always chose the lowest call number it could. A second
+ container for IAX2 pvt structs has been added. It is an astobj2
+ hash table. When we know the remote side's call number, the pvt
+ goes into the hash table with a hash value of the remote side's
+ call number. Then, lookups for incoming media frames are a very
+ fast hash lookup instead of an absolutely insane array traversal.
+ In a quick test, I was able to get more than 3600% more IAX2
+ channels on my machine with these changes.
+
+2008-04-30 16:23 +0000 [r114890] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Don't crash on bad SIP replys. Fix created
+ in Huntsville together with Mark M (putnopvut) (closes issue
+ #12363) Reported by: jvandal Tested by: putnopvut, oej
+
+2008-04-30 14:46 +0000 [r114875-114880] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/iax2.h, channels/chan_iax2.c: use the ARRAY_LEN macro
+ for indexing through the iaxs/iaxsl arrays so that the size of
+ the arrays can be adjusted in one place, and change the size of
+ the arrays from 32768 calls to 2048 calls when LOW_MEMORY is
+ defined
+
+ * Makefile.rules: pay attention to *all* header files for
+ dependency tracking, not just the local ones (inspired by r578 of
+ asterisk-addons by tilghman)
+
+2008-04-29 19:40 +0000 [r114848] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Use the MACRO_CONTEXT and MACRO_EXTEN channel
+ variables instead of the channel's macrocontext and macroexten
+ fields. This is needed because if macros are daisy-chained, the
+ incorrect context and extension are placed on the new channel. I
+ also added locking to the channel prior to accessing these
+ variables as noted in trunk's janitor project file. (closes issue
+ #12549) Reported by: darren1713 Patches:
+ app_queue.c.macroextenpatch uploaded by darren1713 (license 116)
+ (with modifications from me) Tested by: putnopvut
+
+2008-04-29 17:08 +0000 [r114829] Jason Parker <jparker@digium.com>
+
+ * res/res_config_pgsql.c: Change warning message to debug, since
+ there are cases where 0 results is perfectly fine.
+
+2008-04-29 12:53 +0000 [r114823] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 114822
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr
+ 2008) | 2 lines stop script from appending source code if run
+ multiple times ........
+
+2008-04-28 04:47 +0000 [r114708] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, channels/chan_gtalk.c: When modules are
+ embedded, they take on a different name, without the ".so"
+ extension. Specifically check for this name, when we're checking
+ if a module is loaded. (Closes issue #12534)
+
+2008-04-27 01:26 +0000 [r114695] Sean Bright <sean.bright@gmail.com>
+
+ * configure, configure.ac: When we don't explicitly pass a path to
+ the --with-tds configure option, we may end up finding tds.h in
+ /usr/local/include instead of /usr/include. If this happens, the
+ grep that looks for the version (from tdsver.h) will fail and
+ we'll have some problems during the build.
+
+2008-04-26 13:15 +0000 [r114689] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/vmail.cgi: Clicking forward without selecting a
+ message leaves an errant .lock file. (closes issue #12528)
+ Reported by: pukepail Patches: patch.diff uploaded by pukepail
+ (license 431)
+
+2008-04-25 21:54 +0000 [r114673] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Use consistent logic for checking to see if
+ a call number has been chosen yet. Also, remove some redundant
+ logic I recently added in a fix.
+
+2008-04-25 19:32 +0000 [r114662] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c: Move the unlock of the spyee channel to
+ outside the start_spying() function so that the channel is not
+ unlocked twice when using whisper mode.
+
+2008-04-25 15:53 +0000 [r114649] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/zapata.conf.sample, configs/iax.conf.sample,
+ configs/iaxprov.conf.sample, configs/sip.conf.sample: Reference
+ documentation files that actually exist. (closes issue #12516)
+ Reported by: linuxmaniac Patches: diff_rev114611.patch uploaded
+ by linuxmaniac (license 472)
+
+2008-04-24 21:35 +0000 [r114624-114632] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Re-invite RTP during a masquerade so that,
+ for instance, an AMI redirect of two channels which are natively
+ bridged will preserve audio on both channels. This prevents a
+ problem with Asterisk not re-inviting due to one of the channels
+ having being a zombie. (closes issue #12513) Reported by:
+ mneuhauser Patches:
+ asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by
+ mneuhauser (license 425)
+
+ * apps/app_queue.c: Output of channel variables when
+ eventwhencalled=vars was set was being truncated two characters.
+ This patch corrects the problem. (closes issue #12493) Reported
+ by: davidw
+
+ * channels/chan_local.c: Resolve a deadlock in chan_local by
+ releasing the channel lock temporarily. (closes issue #11712)
+ Reported by: callguy Patches: 11712.patch uploaded by putnopvut
+ (license 60) Tested by: acunningham
+
+2008-04-24 19:53 +0000 [r114621] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c: Ensure that when we set the accountcode,
+ it actually shows up in the CDR. (Fix for AMI Originate) (Closes
+ issue #12007)
+
+2008-04-24 15:55 +0000 [r114608] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Fix a silly mistake in a change I made
+ yesterday that caused chan_iax2 to blow up very quickly. (issue
+ #12515)
+
+2008-04-24 14:55 +0000 [r114603] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Only have one max-forwards header in
+ outbound REFERs. Discovered in the Asterisk SIP Masterclass in
+ Orlando. Thanks Joe!
+
+2008-04-23 22:18 +0000 [r114597-114600] Russell Bryant <russell@digium.com>
+
+ * main/http.c: Improve some broken cookie parsing code. Previously,
+ manager login over HTTP would only work if the mansession_id
+ cookie was first. Now, the code builds a list of all of the
+ cookies in the Cookie header. This fixes a problem observed by
+ users of the Asterisk GUI. (closes AST-20)
+
+ * apps/app_chanspy.c, main/http.c: Fix an issue that caused getting
+ the correct next channel to not always work. Also, remove setting
+ the amount of time to wait for a digit from 5 seconds back down
+ to 1/10 of a second. I believe this was so the beep didn't get
+ played over and over really fast, but a while back I put in
+ another fix for that issue. (closes issue #12498) Reported by:
+ jsmith Patches: app_chanspy_channel_walk.trunk.patch uploaded by
+ jsmith (license 15)
+
+2008-04-23 18:28 +0000 [r114594] Jason Parker <jparker@digium.com>
+
+ * res/res_musiconhold.c: Fix reload/unload for res_musiconhold
+ module. (closes issue #11575) Reported by: sunder Patches:
+ M11575_14_rev3.diff uploaded by junky (license 177)
+ bug11575_trunk.diff.txt uploaded by jamesgolovich (license 176)
+
+2008-04-23 17:55 +0000 [r114587-114591] Russell Bryant <russell@digium.com>
+
+ * main/manager.c, include/asterisk/manager.h: Store the manager
+ session ID explicitly as 4 byte ID instead of a ulong. The
+ mansession_id cookie is coded to be limited to 8 characters of
+ hex, and this could break logins from 64-bit machines in some
+ cases. (inspired by AST-20)
+
+ * channels/chan_iax2.c: Fix find_callno_locked() to actually return
+ the callno locked in some more cases.
+
+2008-04-23 16:51 +0000 [r114584] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Add 502 support for both directions, not
+ only one... (see r114571)
+
+2008-04-23 14:54 +0000 [r114579] Joshua Colp <jcolp@digium.com>
+
+ * main/pbx.c: Instead of stopping dialplan execution when SayNumber
+ attempts to say a large number that it can not print out a
+ message informing the user and continue on. (closes issue #12502)
+ Reported by: bcnit
+
+2008-04-22 23:51 +0000 [r114571] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Treat a 502 just like a 503, when it comes
+ to processing a response code
+
+2008-04-22 22:15 +0000 [r114522-114558] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: When we receive a full frame that is
+ supposed to contain our call number, ensure that it has the
+ correct one. (closes issue #10078) (AST-2008-006)
+
+ * main/rtp.c, main/channel.c, formats/format_pcm.c, main/file.c: I
+ thought I was going to be able to leave 1.4 alone, but that was
+ not the case. I ran into some problems with G.722 in 1.4, so I
+ have merged in all of the fixes in this area that I have made in
+ trunk/1.6.0, and things are happy again.
+
+ * res/res_musiconhold.c: Trivial change to read the number of
+ samples from a frame before calling ast_write()
+
+ * res/res_features.c: After a parked call times out, allow the call
+ back to the parker to time out. (closes issue #10890)
+
+ * channels/chan_iax2.c: If the dial string passed to the call
+ channel callback does not indicate an extension, then consider
+ the extension on the channel before falling back to the default.
+ (closes issue #12479) Reported by: darren1713 Patches:
+ exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license
+ 116)
+
+ * channels/chan_sip.c, include/asterisk/sched.h: Merge changes from
+ team/russell/issue_9520 These changes make sure that the
+ reference count for sip_peer objects properly reflects the fact
+ that the peer is sitting in the scheduler for a scheduled
+ callback for qualifying peers or for expiring registrations.
+ Without this, it was possible for these callbacks to happen at
+ the same time that the peer was being destroyed. This was
+ especially likely to happen with realtime peers, and for people
+ making use of the realtime prune CLI command. (closes issue
+ #9520) Reported by: kryptolus Committed patch by me
+
+2008-04-21 14:39 +0000 [r114322] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Only drop audio if we receive it without a
+ progress indication. We allow other frames through such as DTMF
+ because they may be needed to complete the call. (closes issue
+ #12440) Reported by: aragon
+
+2008-04-19 13:57 +0000 [r114297-114299] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_playback.c: Ensure that help text terminates with a
+ newline
+
+ * res/res_musiconhold.c: MOH usage information needs a terminating
+ newline, or else "asterisk -rx 'help moh reload'" will hang.
+ Reported via -dev list, fixed by me.
+
+2008-04-18 21:48 +0000 [r114275-114284] Russell Bryant <russell@digium.com>
+
+ * main/manager.c: Don't destroy a manager session if poll() returns
+ an error of EAGAIN.
+
+ * Makefile: ensure directories are created before we try to install
+ stuff into them
+
+ * Makefile: SUBDIRS_INSTALL is already listed as a subtarget for
+ bininstall
+
+2008-04-18 17:44 +0000 [r114257] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_zap.c, main/callerid.c: Clearing up error messages
+ so they make a bit more sense. Also removing a redundant error
+ message. Issue AST-15
+
+2008-04-18 15:24 +0000 [r114248] Russell Bryant <russell@digium.com>
+
+ * channels/chan_agent.c: Ensure that we don't ast_strdupa(NULL)
+ (closes issue #12476) Reported by: davidw Patch by me
+
+2008-04-18 13:33 +0000 [r114245] Sean Bright <sean.bright@gmail.com>
+
+ * channels/chan_sip.c: Only complete the SIP channel name once for
+ 'sip show channel <channel>'
+
+2008-04-18 06:49 +0000 [r114242] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_setcallerid.c: For consistency sake, ensure that the
+ values that ${CALLINGPRES} returns are valid as an input to
+ SetCallingPres. (Closes issue #12472)
+
+2008-04-17 22:15 +0000 [r114230] Russell Bryant <russell@digium.com>
+
+ * main/autoservice.c: Remove redundant safety net. The check for
+ the autoservice channel list state accomplishes the same goal in
+ a better way. (issue #12470) Reported By: atis
+
+2008-04-17 21:03 +0000 [r114207-114226] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c: Declaration of the peer channel in this scope
+ was making it so the peer variable defined in the outer scope was
+ never set properly, therefore making iterating through the
+ channel list always restart from the beginning. This bug would
+ have affected anyone who called chanspy without specifying a
+ first argument. (closes issue #12461) Reported by: stever28
+
+ * main/frame.c, include/asterisk/dsp.h: Add prototype for
+ ast_dsp_frame_freed. I'm not sure how this was compiling
+ before...
+
+ * main/dsp.c, main/frame.c, include/asterisk/frame.h: It was
+ possible for a reference to a frame which was part of a freed DSP
+ to still be referenced, leading to memory corruption and eventual
+ crashes. This code change ensures that the dsp is freed when we
+ are finished with the frame. This change is very similar to a
+ change Russell made with translators back a month or so ago.
+ (closes issue #11999) Reported by: destiny6628 Patches:
+ 11999.patch uploaded by putnopvut (license 60) Tested by:
+ destiny6628, victoryure
+
+2008-04-17 16:23 +0000 [r114204] Russell Bryant <russell@digium.com>
+
+ * Makefile: Fix the bininstall target to install from subdirs, as
+ well. (closes issue AST-8, patch from bmd at switchvox)
+
+2008-04-17 13:42 +0000 [r114198] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * res/res_jabber.c: Use keepalives effectively in order diagnose
+ bug #12432.
+
+2008-04-17 12:56 +0000 [r114195] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_agi.c: Add special case for when the agi cannot be
+ executed, to comply with the documentation that we return failure
+ in that case. (closes issue #12462) Reported by: fmueller
+ Patches: 20080416__bug12462.diff.txt uploaded by Corydon76
+ (license 14) Tested by: fmueller
+
+2008-04-17 10:51 +0000 [r114191] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_chanspy.c: Make sure we have enough room for the
+ recording's filename.
+
+2008-04-16 20:46 +0000 [r114184] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c: use the ZT_SET_DIALPARAMS ioctl properly by
+ initializing the structure to all zeroes in case it contains
+ fields that we don't write values into (which it does as of
+ Zaptel 1.4.10) (closes issue #12456) Reported by: fnordian
+
+2008-04-16 19:59 +0000 [r114180] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_vpb.cc: Backport revisions for latest vpb drivers
+ to 1.4 (Closes issue #12457)
+
+2008-04-16 17:30 +0000 [r114173] Jason Parker <jparker@digium.com>
+
+ * channels/chan_zap.c: Fix "fallthrough" behavior here, so config
+ options in a previously configured user don't override settings
+ in general. (closes issue #12458) Reported by: tzafrir Patches:
+ chanzap_users_sections.diff uploaded by tzafrir (license 46)
+
+2008-04-16 14:10 +0000 [r114167] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_meetme.c: Include the proper headers for using mkdir on
+ FreeBSD. (closes issue #12430) Reported by: ys Patches:
+ app_meetme.c.diff uploaded by ys (license 281)
+
+2008-04-15 20:26 +0000 [r114148] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Handle subscribe queues in all situations...
+ Thanks to festr_ on irc for telling me about this bug.
+
+2008-04-15 17:17 +0000 [r114120-114138] Jason Parker <jparker@digium.com>
+
+ * contrib/scripts/autosupport: Update Digium autosupport script,
+ for more useful information. (closes issue #12452) Reported by:
+ angler Patches: autosupport.diff uploaded by angler (license 106)
+
+ * apps/app_queue.c: Allow autofill to work in the general section
+ of queues.conf. Additionally, don't try to (re)set options when
+ they have empty values in realtime (all unset columns would have
+ an empty value). (closes issue #12445) Reported by: atis Patches:
+ 12445-autofill.diff uploaded by qwell (license 4)
+
+ * channels/chan_h323.c: The call_token on the pvt can occasionally
+ be NULL, causing a crash. If it is NULL, we can skip this
+ channel, since it can't the one we're looking for. (closes issue
+ #9299) Reported by: vazir
+
+2008-04-14 17:41 +0000 [r114106-114117] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c: Increase the retry count when attempting to show
+ channels. This apparently cleared an issue someone was seeing
+ when attempting to show channels when the load was high. (closes
+ issue #11667) Reported by: falves11 Patches: 11677.txt uploaded
+ by russell (license 2) Tested by: falves11
+
+ * apps/app_dial.c, apps/app_queue.c: If the datastore has been
+ moved to another channel due to a masquerade, then freeing the
+ datastore here causes an eventual double free when the new
+ channel hangs up. We should only free the datastore if we were
+ able to successfully remove it from the channel we are
+ referencing (i.e. the datastore was not moved). (closes issue
+ #12359) Reported by: pguido
+
+ * main/channel.c: Save a local copy of the generate callback prior
+ to unlocking the channel in case the generate callback goes NULL
+ on us after the channel is unlocked. Thanks to Russell for
+ pointing this need out to me.
+
+2008-04-14 14:52 +0000 [r114100-114103] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: It is possible for the remote side to say
+ they want T38 but not give any capabilities. (closes issue
+ #12414) Reported by: MVF
+
+ * main/rtp.c: Don't change the SSRC when a new source comes into
+ play, this might happen quite often and depending on the remote
+ side... they might not like this. (closes issue #12353) Reported
+ by: dimas
+
+2008-04-11 22:32 +0000 [r114083] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_iax2.c: Several places in the code called
+ find_callno() (which releases the lock on the pvt structure) and
+ then immediately locked the call and did things with it.
+ Unfortunately, the call can disappear between the find_callno and
+ the lock, causing Bad Stuff(tm) to happen. Added
+ find_callno_locked() function to return the callno withtout
+ unlocking for instances that it is needed. (issue #12400)
+ Reported by: ztel
+
+2008-04-11 21:35 +0000 [r114072] Jason Parker <jparker@digium.com>
+
+ * main/pbx.c: It's possible that a channel can have an async goto
+ on the successful execution of an application as well. Closes
+ issue #12172.
+
+2008-04-11 15:44 +0000 [r114045-114063] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_features.c: Fix a race condition that may happen between
+ a sip hangup and a "core show channel" command. This patch adds
+ locking to prevent the resulting crash. (closes issue #12155)
+ Reported by: tsearle Patches: show_channels_crash2.patch uploaded
+ by tsearle (license 373) Tested by: tsearle
+
+ * main/utils.c, include/asterisk/lock.h: Fix 1.4 build when
+ LOW_MEMORY is enabled.
+
+ * channels/chan_sip.c: Be sure that we're not about to set
+ bridgepvt NULL prior to dereferencing it. (closes issue #11775)
+ Reported by: fujin
+
+2008-04-10 17:26 +0000 [r114035] Jason Parker <jparker@digium.com>
+
+ * main/file.c: Only try to prefix language if we are not using an
+ absolute path (suffix it otherwise).
+ en/var/lib/asterisk/sounds/blah.gsm is a very silly path. (closes
+ issue #12379) Reported by: kuj Patches: 12379-absolutepath.diff
+ uploaded by qwell (license 4) Tested by: kuj, qwell
+
+2008-04-10 15:58 +0000 [r114021-114032] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_voicemail.c: Forgot the 1.4 branch for russian language
+ fix. (closes issue #12404) Reported by: IgorG Patches:
+ voicemail_ru_hardcoded-v1.patch uploaded by IgorG (license 20)
+
+ * apps/app_meetme.c: Create the directory where name recordings
+ will go if it does not exist. (closes issue #12311) Reported by:
+ rkeene Patches: 12311-mkdir.diff uploaded by qwell (license 4)
+
+ * channels/chan_sip.c: Don't add custom URI options if they don't
+ exist OR they are empty. (closes issue #12407) Reported by:
+ homesick Patches: uri_options-1.4.diff uploaded by homesick
+ (license 91)
+
+2008-04-09 20:54 +0000 [r113927] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: We need to set the persistant_route [sic]
+ parameter for the sip_pvt during the initial INVITE, no matter if
+ we're building the route set from an INVITE request or response.
+ (closes issue #12391) Reported by: benjaminbohlmann Tested by:
+ benjaminbohlmann
+
+2008-04-09 18:57 +0000 [r113874] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_csv.c, configs/cdr.conf.sample: If the [csv] section does
+ not exist in cdr.conf, then an unload/load sequence is needed to
+ correct the problem. Track whether the load succeeded with a
+ variable, so we can fix this with a simple reload event, instead.
+
+2008-04-09 16:50 +0000 [r113784] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: If we receive an AUTHREQ from the remote
+ server and we are unable to reply (for example they have a secret
+ configured, but we do not) then queue a hangup frame on the
+ Asterisk channel. This will cause the channel to hangup and a
+ HANGUP to be sent via IAX2 to the remote side which is the proper
+ thing to do in this scenario. (closes issue #12385) Reported by:
+ viraptor
+
+2008-04-09 14:40 +0000 [r113681] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: If Asterisk receives a 488 on an INVITE (not
+ a reinvite), then we should not send a BYE. (closes issue #12392)
+ Reported by: fnordian Patches: chan_sip.patch uploaded by
+ fnordian (license 110) with small modification from me
+
+2008-04-09 01:34 +0000 [r113596] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_iax2.c: Initialize fr->cacheable to make valgrind
+ happy
+
+2008-04-08 19:07 +0000 [r113507] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_parkandannounce.c: Fix potential buffer overflow that
+ could happen if more than 100 announce files were specified when
+ calling ParkAndAnnounce. This overflow is not exploitable
+ remotely and so there is no need for a security advisory. (closes
+ issue #12386) Reported by: davidw
+
+2008-04-08 18:48 +0000 [r113402-113504] Jason Parker <jparker@digium.com>
+
+ * channels/chan_skinny.c: Add a little more that is required for
+ previously added devices.
+
+ * channels/chan_skinny.c: Add support for several new(ish) devices
+ - most notably, 7942/7945, 7962/7965, 7975. Thanks to Greg Oliver
+ for providing me the required information.
+
+ * main/asterisk.c: Work around some silliness caused by
+ sys/capability.h - this should fix compile errors a number of
+ users have been experiencing.
+
+2008-04-08 16:51 +0000 [r113348-113399] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/astgenkey.8: Add security note on astgenkey's
+ manpage. (closes issue #12373) Reported by: lmamane Patches:
+ 20080406__bug12373.diff.txt uploaded by Corydon76 (license 14)
+
+ * channels/chan_sip.c: Move check for still-bridged channels out a
+ little further, to avoid possible deadlocks. (Closes issue
+ #12252) Reported by: callguy Patches: 20080319__bug12252.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: callguy
+
+2008-04-08 15:03 +0000 [r113296] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/slinfactory.h, main/slinfactory.c,
+ main/audiohook.c: If audio suddenly gets fed into one side of a
+ channel after a lapse of frames flush the other factory so that
+ old audio does not remain in the factory causing the sync code to
+ not execute. (closes issue #12296) Reported by: jvandal
+
+2008-04-07 21:34 +0000 [r113240] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_sip.c: (closes issue #12362) [redo of 113012] This
+ fixes a for loop (in realtime_peer) to check all the
+ ast_variables the loop was intending to test rather than just the
+ first one. The change exposed the problem of calling memcpy on a
+ NULL pointer, in this case the passed in sockaddr_in struct which
+ is now checked.
+
+2008-04-07 18:00 +0000 [r113118] Jason Parker <jparker@digium.com>
+
+ * channels/chan_skinny.c, configs/skinny.conf.sample: Allow
+ playback with noanswer (and add earlyrtp option). (closes issue
+ #9077) Reported by: pj Patches: earlyrtp.diff uploaded by wedhorn
+ (license 30) Tested by: pj, qwell, DEA, wedhorn
+
+2008-04-07 17:51 +0000 [r113117] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_strings.c: Force ast_mktime() to check for DST, since
+ strptime(3) does not. (Closes issue #12374)
+
+2008-04-07 16:08 +0000 [r113065] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c: This fix prevents a deadlock that was experienced
+ in chan_local. There was deadlock prevention in place in
+ chan_local, but it would not work in a specific case because the
+ channel was recursively locked. By unlocking the channel prior to
+ calling the generator's generate callback in
+ ast_read_generator_actions(), we prevent the recursive locking,
+ and therefore the deadlock. (closes issue #12307) Reported by:
+ callguy Patches: 12307.patch uploaded by putnopvut (license 60)
+ Tested by: callguy
+
+2008-04-07 15:16 +0000 [r113012] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_sip.c: (closes issue #12362) (closes issue #12372)
+ Reported by: vinsik Tested by: tecnoxarxa This one line change
+ makes an if inside a for loop (in realtime_peer) check all the
+ ast_variables the loop was intending to test rather than just the
+ first one.
+
+2008-04-04 19:26 +0000 [r112766-112820] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * channels/chan_gtalk.c: Free newly allocated channel before
+ returning
+
+ * channels/chan_gtalk.c: Prevent call connections when codecs don't
+ match. (closes issue #10604) Reported by: keepitcool Patches:
+ branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested
+ by: phsultan
+
+2008-04-04 00:52 +0000 [r112709-112711] Joshua Colp <jcolp@digium.com>
+
+ * main/Makefile: Pass in the path to Zaptel for systems that
+ install Zaptel headers in a separate location.
+
+ * main/asterisk.c: One thing at a time... let's get 1.4 building.
+
+2008-04-03 23:57 +0000 [r112689] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * main/asterisk.c: add a Zaptel timer check to verify the timer is
+ responding when Zaptel support is compiled into Asterisk and
+ Zaptel drivers are loaded. This will help people not waste their
+ valuable time debugging side effects.
+
+2008-04-03 14:32 +0000 [r112393-112599] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_zap.c: Fix the testing of the "res" variable so
+ that it is more logically correct and makes the correct warning
+ and debug messages print. (closes issue #12361) Reported by:
+ one47 Patches: chan_zap_deferred_digit.patch uploaded by one47
+ (license 23)
+
+ * main/manager.c: Fix a race condition in the manager. It is
+ possible that a new manager event could be appended during a
+ brief time when the manager is not waiting for input. If an event
+ comes during this period, we need to set an indicator that there
+ is an event pending so that the manager doesn't attempt to wait
+ forever for an event that already happened. (closes issue #12354)
+ Reported by: bamby Patches: manager_race_condition.diff uploaded
+ by bamby (license 430) (comments added by me)
+
+ * apps/app_queue.c: Ensure that there is no timeout if none is
+ specified. (closes issue #12349) Reported by: johnlange
+
+2008-04-01 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.19 released.
+
+2008-03-28 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.19-rc4 released.
+
+2008-03-28 16:19 +0000 [r111658] Jason Parker <jparker@digium.com>
+
+ * formats/format_wav_gsm.c: The file size of WAV49 does not need to
+ be an even number. (closes issue #12128) Reported by: mdu113
+ Patches: 12128-noevenlength.diff uploaded by qwell (license 4)
+ Tested by: qwell, mdu113
+
+2008-03-28 14:35 +0000 [r111442-111605] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/valgrind.txt: Update debugging text, since Valgrind
+ eliminated the --log-file-exactly option. (Closes issue #12320)
+
+ * main/acl.c: For FreeBSD, at least, the ifa_addr element could be
+ NULL. (closes issue #12300) Reported by: festr Patches:
+ acl.c.patch uploaded by festr (license 443)
+
+2008-03-27 13:03 +0000 [r111341-111391] Steve Murphy <murf@digium.com>
+
+ * apps/app_playback.c, main/pbx.c: These small documentation
+ updates made in response to a query in asterisk-users, where a
+ user was using Playback, but needed the features of Background,
+ and had no idea that Background existed, or that it might provide
+ the features he needed. I thought the best way to avert these
+ kinds of queries was to provide "See Also" references in all
+ three of "Background", "Playback", "WaitExten". Perhaps a project
+ to do this with all related apps is in order.
+
+ * pbx/pbx_ael.c, include/asterisk/ael_structs.h: (closes issue
+ #12302) Reported by: pj Tested by: murf These changes will set a
+ channel variable ~~EXTEN~~ just before generating code for a
+ switch, with the value of ${EXTEN}. The exten is marked as having
+ a switch, and ever after that, till the end of the exten, we
+ substitute any ${EXTEN} with ${~~EXTEN~~} instead in application
+ arguments; (and the ${EXTEN: also). The reason for this, is that
+ because switches are coded using separate extensions to provide
+ pattern matching, and jumping to/from these switch extensions
+ messes up the ${EXTEN} value, which blows the minds of users.
+
+2008-03-27 00:25 +0000 [r111245-111280] Jason Parker <jparker@digium.com>
+
+ * main/frame.c: Put this flag back so we don't change the API.
+
+ * main/frame.c: Remove excessive smoother optimization that was
+ causing audio glitches (small "pops") after (about 200ms later)
+ an "incorrectly" sized frame was received. While it would be very
+ nice to keep this as optimized as possible, it makes no sense for
+ the smoother to be dropping random bits of audio like this. Isn't
+ that the whole point of a smoother? Closes issue #12093.
+
+2008-03-26 19:55 +0000 [r111129] Joshua Colp <jcolp@digium.com>
+
+ * contrib/scripts/autosupport: Update autosupport script. (closes
+ issue #12310) Reported by: angler Patches: autosupport.diff
+ uploaded by angler (license 106)
+
+2008-03-26 19:51 +0000 [r111126] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, UPGRADE.txt: Merged revisions 111125 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar
+ 2008) | 2 lines update UPGRADE notes to document usage of the
+ script ........
+
+2008-03-26 19:37 +0000 [r111049-111121] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: This code change is made just for
+ clarification. It does exactly the same thing as before. It just
+ doesn't look as wrong.
+
+ * apps/app_voicemail.c: Add a lock to the vm_state structure and
+ use the lock around mail_open calls to prevent concurrent access
+ of the same mailstream. This, along with trunk's ability to
+ configure TCP timeouts for IMAP storage will help to prevent
+ crashes and hangs when using voicemail with IMAP storage. (closes
+ issue #10487) Reported by: ewilhelmsen
+
+2008-03-26 19:06 +0000 [r111024] Kevin P. Fleming <kpfleming@digium.com>
+
+ * codecs/ilbc, /, contrib/scripts/get_ilbc_source.sh (added):
+ Merged revisions 111019 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar
+ 2008) | 2 lines add a script to make getting the iLBC source code
+ simple for end users ........
+
+2008-03-26 19:04 +0000 [r111014-111020] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: If we are requested to authenticate a
+ reinvite make sure that it contains T38 SDP if need be. (closes
+ issue #11995) Reported by: fall
+
+ * channels/chan_iax2.c: Make sure that full video frames are sent
+ whenever the 15 bit timestamp rolls over. (closes issue #11923)
+ Reported by: mihai Patches: asterisk-fullvideo.patch uploaded by
+ mihai (license 94)
+
+2008-03-26 17:43 +0000 [r110880-110962] Kevin P. Fleming <kpfleming@digium.com>
+
+ * UPGRADE.txt: add note that the user will need to enable
+ codec_ilbc to get it to build
+
+ * codecs/ilbc/StateConstructW.h (removed),
+ codecs/ilbc/libilbc.vcproj (removed), codecs/ilbc/packing.h
+ (removed), codecs/ilbc/getCBvec.c (removed),
+ codecs/ilbc/LPCdecode.c (removed), codecs/ilbc/enhancer.c
+ (removed), codecs/ilbc/lsf.c (removed), codecs/ilbc/iLBC_encode.c
+ (removed), codecs/ilbc/getCBvec.h (removed),
+ codecs/ilbc/LPCdecode.h (removed), codecs/ilbc/enhancer.h
+ (removed), codecs/ilbc/FrameClassify.c (removed),
+ codecs/ilbc/iLBC_define.h (removed), codecs/ilbc/lsf.h (removed),
+ codecs/ilbc/iLBC_encode.h (removed), codecs/ilbc/FrameClassify.h
+ (removed), codecs/ilbc/helpfun.c (removed), codecs/ilbc/doCPLC.c
+ (removed), codecs/ilbc/anaFilter.c (removed),
+ codecs/ilbc/helpfun.h (removed), codecs/ilbc/createCB.c
+ (removed), codecs/ilbc/doCPLC.h (removed),
+ codecs/ilbc/anaFilter.h (removed), UPGRADE.txt,
+ codecs/ilbc/iLBC_decode.c (removed), codecs/ilbc/constants.c
+ (removed), codecs/ilbc/createCB.h (removed), CHANGES,
+ codecs/ilbc/iLBC_decode.h (removed), codecs/ilbc/constants.h
+ (removed), codecs/Makefile, codecs/ilbc/iCBSearch.c (removed),
+ codecs/ilbc/filter.c (removed), codecs/ilbc/hpInput.c (removed),
+ codecs/ilbc/gainquant.c (removed), codecs/ilbc/hpOutput.c
+ (removed), codecs/ilbc/iCBSearch.h (removed),
+ codecs/ilbc/filter.h (removed), codecs/ilbc/hpInput.h (removed),
+ codecs/ilbc/gainquant.h (removed), codecs/ilbc/LPCencode.c
+ (removed), codecs/ilbc/hpOutput.h (removed),
+ codecs/ilbc/StateSearchW.c (removed), codecs/codec_ilbc.c,
+ codecs/ilbc/LPCencode.h (removed), codecs/ilbc/StateSearchW.h
+ (removed), codecs/ilbc/iCBConstruct.c (removed),
+ codecs/ilbc/syntFilter.c (removed), /, codecs/ilbc/iCBConstruct.h
+ (removed), codecs/ilbc/syntFilter.h (removed),
+ codecs/ilbc/StateConstructW.c (removed), codecs/ilbc/packing.c
+ (removed): Merged revisions 110869 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar
+ 2008) | 2 lines due to licensing restrictions, we cannot
+ distribute the source code for iLBC encoding and decoding... so
+ remove it, and add instructions on how the user can obtain it
+ themselves ........
+
+2008-03-25 22:51 +0000 [r110779] Jason Parker <jparker@digium.com>
+
+ * cdr/cdr_custom.c: Make file access in cdr_custom similar to
+ cdr_csv. Fixes issue #12268. Patch borrowed from r82344
+
+2008-03-25 20:03 +0000 [r110727] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_sip.c: This one line change makes an if inside a
+ for loop (in realtime_peer) check all the ast_variables the loop
+ was intending to test rather than just the first one.
+
+2008-03-25 15:40 +0000 [r110635] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: When reverting a commit, I accidentally left
+ in this bit which was an experiment to see what would happen. It
+ passed the compile test, and I didn't notice I had left this
+ change in too. So this is a revert of a revert...sort of.
+
+2008-03-25 14:37 +0000 [r110628] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/options.h, main/asterisk.c, Makefile,
+ main/app.c: Add an option (transmit_silence) which transmits
+ silence during both Record() and DTMF generation. The reason this
+ is an option is that in order to transmit silence we have to
+ setup a translation path. This may not be needed/wanted in all
+ cases. (closes issue #10058) Reported by: tracinet
+
+2008-03-24 19:17 +0000 [r110618] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: This is a revert for revision 108288. The
+ reason is that that revision was not for an actual bug fix per
+ se, and so it really should not have been in 1.4 in the first
+ place. Plus, people who compile with DO_CRASH are more likely to
+ encounter a crash due to this change. While I think the usage of
+ DO_CRASH in ast_sched_del is a bit absurd, this sort of change is
+ beyond the scope of 1.4 and should be done instead in a developer
+ branch based on trunk so that all scheduler functions are fixed
+ at once. I also am reverting the change to trunk and 1.6 since
+ they also suffer from the DO_CRASH potential. (closes issue
+ #12272) Reported by: qq12345
+
+2008-03-24 17:34 +0000 [r110614] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Turn a NOTICE into a DEBUG message.
+
+2008-03-21 14:32 +0000 [r110474] Jason Parker <jparker@digium.com>
+
+ * codecs/gsm/Makefile: Don't attempt to do optimizations of gsm on
+ mips platforms either. (closes issue #12270) Reported by:
+ zandbelt Patches: 026-gsm-mips.patch uploaded by zandbelt
+ (license 33)
+
+2008-03-20 23:13 +0000 [r110163-110395] Russell Bryant <russell@digium.com>
+
+ * main/autoservice.c: Shorten the ast_waitfor() timeout from 500 ms
+ to 50 ms in the autoservice thread. This really should not make a
+ difference except in very rare cases. That case would be that all
+ of the channels in autoservice are not generating any frames. In
+ that case, this change reduces the potential amount of time that
+ a thread waits in ast_autoservice_stop() for the autoservice
+ thread to wrap back around to the beginning of its loop. (closes
+ issue #12266, reported by dimas)
+
+ * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
+ 110335 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008)
+ | 6 lines Fix some very broken code that was introduced in 1.2.26
+ as a part of the security fix. The dnsmgr is not appropriate
+ here. The dnsmgr takes a pointer to an address structure that a
+ background thread continuously updates. However, in these cases,
+ a stack variable was passed. That means that the dnsmgr thread
+ would be continuously writing to bogus memory. ........
+
+ * apps/app_meetme.c: Fix a bug where when calls on the trunk side
+ hang up while on hold, the state is not properly reflected.
+ (closes issue #11990, reported by anakaoka, patched by me)
+
+2008-03-19 20:33 +0000 [r110083] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c: Add a missing unlock in the case that memory
+ allocation fails in app_chanspy. Thanks to Russell for confirming
+ that this was an issue.
+
+2008-03-19 19:11 +0000 [r110019-110035] Joshua Colp <jcolp@digium.com>
+
+ * res/res_musiconhold.c: Add sanity checking for position resuming.
+ We *have* to make sure that the position does not exceed the
+ total number of files present, and we have to make sure that the
+ position's filename is the same as previous. These values can
+ change if a music class is reloaded and give unpredictable
+ behavior. (closes issue #11663) Reported by: junky
+
+ * main/rtp.c: Make sure that the mark bit does not incorrectly
+ cause video frame timestamps to be calculated as if they are
+ audio frames. (closes issue #11429) Reported by: sperreault
+ Patches: 11429-frametype.diff uploaded by qwell (license 4)
+
+2008-03-19 17:12 +0000 [r109973] Jason Parker <jparker@digium.com>
+
+ * Makefile, build_tools/cflags.xml, build_tools/cflags-devmode.xml
+ (added): People report bugs about Asterisk crashing with DO_CRASH
+ enabled was getting a little silly... Now we only show certain
+ cflags when you run configure with --enable-dev-mode
+ (corresponding menuselect change to follow)
+
+2008-03-19 15:41 +0000 [r109908] Steve Murphy <murf@digium.com>
+
+ * main/config.c: (closes issue #11442) Reported by: tzafrir
+ Patches: 11442.patch uploaded by murf (license 17) Tested by:
+ murf I didn't give tzafrir very much time to test this, but if he
+ does still have remaining issues, he is welcome to re-open this
+ bug, and we'll do what is called for. I reproduced the problem,
+ and tested the fix, so I hope I am not jumping by just going
+ ahead and committing the fix. The problem was with what file_save
+ does with templates; firstly, it tended to print out multiple
+ options: [my_category](!)(templateref) instead of
+ [my_category](!,templateref) which is fixed by this patch.
+ Nextly, the code to suppress output of duplicate declarations
+ that would occur because the reader copies inherited declarations
+ down the hierarchy, was not working. Thus: [master-template](!)
+ mastervar = bar [template](!,master-template) tvar = value
+ [cat](template) catvar = val would be rewritten as: ;! ;!
+ Automatically generated configuration file ;! Filename:
+ experiment.conf (/etc/asterisk/experiment.conf) ;! Generator:
+ Manager ;! Creation Date: Tue Mar 18 23:17:46 2008 ;!
+ [master-template](!) mastervar = bar
+ [template](!,master-template) mastervar = bar tvar = value
+ [cat](template) mastervar = bar tvar = value catvar = val This
+ has been fixed. Since the config reader 'explodes' inherited vars
+ into the category, users may, in certain circumstances, see
+ output different from what they originally entered, but it should
+ be both correct and equivalent.
+
+2008-03-19 04:06 +0000 [r109763-109838] Russell Bryant <russell@digium.com>
+
+ * main/utils.c: Tweak spacing in a recent change because I'm very
+ picky.
+
+ * apps/app_chanspy.c: Fix one place where the chanspy datastore
+ isn't removed from a channel. (issue #12243, reported by atis,
+ patch by me)
+
+2008-03-18 20:52 +0000 [r109713] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: This patch makes it so that all queue member
+ status changes are handled through device state code. This
+ removes several problems people were seeing where their queue
+ members would get into an "unknown" state. Huge props go to atis
+ on this one since he was the one who found the code section that
+ was causing the problem and proposed the solution. I just wrote
+ what he suggested :) (closes issue #12127) Reported by: atis
+ Patches: 12127v3.patch uploaded by putnopvut (license 60) Tested
+ by: atis, jvandal
+
+2008-03-18 19:23 +0000 [r109648] Jason Parker <jparker@digium.com>
+
+ * codecs/log2comp.h: Allow codecs that use log2comp (g726) to
+ compile correctly on x86 with gcc4 optimizations. (closes issue
+ #12253) Reported by: fossil Patches: log2comp.patch uploaded by
+ fossil (license 140)
+
+2008-03-18 17:58 +0000 [r109575] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_agent.c: Make sure an agent doesn't try to send
+ dtmf to a NULL channel closes issue #12242 Reported by Yourname
+
+2008-03-18 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.19-rc3 released.
+
+2008-03-18 16:25 +0000 [r109482] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/astobj.h: Fix character string being treated ad
+ format string
+
+2008-03-18 15:10 +0000 [r109393] Jason Parker <jparker@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 109391 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r109391 | qwell | 2008-03-18 10:08:41 -0500 (Tue, 18 Mar 2008) |
+ 3 lines Do not return with a successful authentication if the
+ From header ends up empty. (AST-2008-003) ........
+
+2008-03-18 14:58 +0000 [r109386] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c, channels/chan_sip.c: Put a maximum limit on the
+ number of payloads accepted, and also make sure a given payload
+ does not exceed our maximum value. (AST-2008-002)
+
+2008-03-18 06:37 +0000 [r109309] Steve Murphy <murf@digium.com>
+
+ * pbx/ael/ael-test/ael-ntest23 (added),
+ pbx/ael/ael-test/ael-ntest23/t1/a.ael (added),
+ pbx/ael/ael-test/ael-ntest23/t1/b.ael (added),
+ pbx/ael/ael-test/ael-ntest23/t1/c.ael (added),
+ pbx/ael/ael-test/ael-ntest23/t2/d.ael (added),
+ pbx/ael/ael-test/ael-ntest23/t2/e.ael (added),
+ pbx/ael/ael-test/ael-ntest23/t2/f.ael (added),
+ pbx/ael/ael-test/ref.ael-ntest23 (added), pbx/ael/ael_lex.c,
+ pbx/ael/ael-test/ael-ntest23/t3/g.ael (added),
+ pbx/ael/ael-test/ael-ntest23/t3/h.ael (added),
+ pbx/ael/ael-test/ael-ntest23/t3/i.ael (added), pbx/ael/ael.flex,
+ pbx/ael/ael-test/ael-ntest23/t3/j.ael (added),
+ pbx/ael/ael-test/ael-ntest23/qq.ael (added),
+ pbx/ael/ael-test/ael-ntest23/t1 (added),
+ pbx/ael/ael-test/ael-ntest23/t2 (added),
+ pbx/ael/ael-test/ael-ntest23/t3 (added),
+ pbx/ael/ael-test/ael-ntest23/extensions.ael (added): (closes
+ issue #11903) Reported by: atis Many thanks to atis for spotting
+ this problem and reporting it. The fix was to straighten out how
+ items are placed on and removed from the file stack. Regressions
+ as well as the provided test case helped to straighten out all
+ code paths. valgrind was used to make sure all memory allocated
+ was freed. Sorry for not solving this earlier. I got distracted.
+ Added the ntest23 regression test, which is mainly a copy of
+ ntest22, but with a few juicy errors thrown in, to replicate the
+ kind of error that atis spotted.
+
+2008-03-17 22:05 +0000 [r109226] Mark Michelson <mmichelson@digium.com>
+
+ * main/utils.c: Fix a logic flaw in the code that stores lock info
+ which is displayed via the "core show locks" command. The idea
+ behind this section of code was to remove the previous lock from
+ the list if it was a trylock that had failed. Unfortunately,
+ instead of checking the status of the previous lock, we were
+ referencing the index immediately following the previous lock in
+ the lock_info->locks array. The result of this problem, under the
+ right circumstances, was that the lock which we currently in the
+ process of attempting to acquire could "overwrite" the previous
+ lock which was acquired. While this does not in any way affect
+ typical operation, it *could* lead to misleading "core show
+ locks" output.
+
+2008-03-17 17:55 +0000 [r109171] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: Update the directory of placed calls on
+ skinny phones when dialing a channel that does not provide
+ progress (analog ZAP lines) The phone does handle the double
+ update on calls to channels that do provide progress and wont
+ insert duplicate items (closes issue #12239) Reported by: DEA
+ Patches: chan_skinny-call-log.txt uploaded by DEA (license 3)
+
+2008-03-17 16:24 +0000 [r109107] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: 200 OKs in response to a reinvite need to be
+ sent reliably. If the remote side does not receive one the dialog
+ will be torn down. (closes issue #12208) Reported by: atrash
+
+2008-03-17 15:15 +0000 [r109057] Jason Parker <jparker@digium.com>
+
+ * main/file.c: Backport revision 106439 from trunk. I didn't
+ realize this was broken in 1.4 as well. Closes issue #12222.
+
+2008-03-17 14:18 +0000 [r109012] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c: Make sure that we release the lock on the
+ spyee channel if the spyee or spy has hung up (closes issue
+ #12232) Reported by: atis
+
+2008-03-16 21:47 +0000 [r108961] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/dial.c: add missing break to case AST_CONTROL_SRCUPDATE
+ (closes issue #12228) Reported by: andrew Patches: SRC.patch
+ uploaded by andrew (license 240)
+
+2008-03-14 20:09 +0000 [r108792-108796] Russell Bryant <russell@digium.com>
+
+ * channels/chan_oss.c: Fix a channel name issue. chan_oss registers
+ the "Console" channel type, but it created channels with an "OSS"
+ prefix. (closes issue #12194, reported by davidw, patched by me)
+
+ * contrib/init.d/rc.suse.asterisk: Update the SuSE init script to
+ start networking before asterisk, as well. (closes issue #12200,
+ reported by and change suggested by reinerotto)
+
+2008-03-14 16:44 +0000 [r108737] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix a race condition in the SIP packet
+ scheduler which could cause a crash. chan_sip uses the scheduler
+ API in order to schedule retransmission of reliable packets (such
+ as INVITES). If a retransmission of a packet is occurring, then
+ the packet is removed from the scheduler and retrans_pkt is
+ called. Meanwhile, if a response is received from the packet as
+ previously transmitted, then when we ACK the response, we will
+ remove the packet from the scheduler and free the packet. The
+ problem is that both the ACK function and retrans_pkt attempt to
+ acquire the same lock at the beginning of the function call. This
+ means that if the ACK function acquires the lock first, then it
+ will free the packet which retrans_pkt is about to read from and
+ write to. The result is a crash. The solution: 1. If the ACK
+ function fails to remove the packet from the scheduler and the
+ retransmit id of the packet is not -1 (meaning that we have not
+ reached the maximum number of retransmissions) then release the
+ lock and yield so that retrans_pkt may acquire the lock and
+ operate. 2. Make absolutely certain that the ACK function does
+ not recursively lock the lock in question. If it does, then
+ releasing the lock will do no good, since retrans_pkt will still
+ be unable to acquire the lock. (closes issue #12098) Reported by:
+ wegbert (closes issue #12089) Reported by: PTorres Patches:
+ 12098-putnopvutv3.patch uploaded by putnopvut (license 60) Tested
+ by: jvandal
+
+2008-03-14 14:29 +0000 [r108682] Jason Parker <jparker@digium.com>
+
+ * res/res_musiconhold.c: Fix a potential segfault if chan (or
+ chan->music_state) is NULL. Closes issue #12210, credit to
+ edantie for pointing this out.
+
+2008-03-13 21:38 +0000 [r108469-108583] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c, main/channel.c, include/asterisk/channel.h:
+ Fix another issue that was causing crashes in chanspy. This
+ introduces a new datastore callback, called chan_fixup(). The
+ concept is exactly like the fixup callback that is used in the
+ channel technology interface. This callback gets called when the
+ owning channel changes due to a masquerade. Before this was
+ introduced, if a masquerade happened on a channel being spyed on,
+ the channel pointer in the datastore became invalid. (closes
+ issue #12187) (reported by, and lots of testing from atis) (props
+ to file for the help with ideas)
+
+ * channels/chan_sip.c: Make a tweak that gets the LEDs on polycom
+ phones to blink when an extension that has been subscribed to
+ goes on hold. Otherwise, they just stay on like it does when an
+ extension is in use. (closes issue #11263) Reported by: russell
+ Patches: notify_hold.rev1.txt uploaded by russell (license 2)
+ Tested by: russell
+
+ * apps/app_followme.c: Fix a couple uses of sprintf. The second one
+ could actually cause an overflow of a stack buffer. It's not a
+ security issue though, it only depends on your configuration.
+
+2008-03-12 21:53 +0000 [r108227-108288] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Change AST_SCHED_DEL use to ast_sched_del
+ for autocongestion in chan_sip. The scheduler callback will
+ always return 0. This means that this id is never rescheduled, so
+ it makes no sense to loop trying to delete the id from the
+ scheduler queue. If we fail to remove the item from the queue
+ once, it will fail every single time. (Yes I realize that in this
+ case, the macro would exit early because the id is set to -1 in
+ the callback, but it still makes no sense to use that macro in
+ favor of calling ast_sched_del once and being done with it) This
+ is the first of potentially several such fixes.
+
+ * include/asterisk/sched.h: Added a large comment before the
+ AST_SCHED_DEL macro to explain its purpose as well as when it is
+ appropriate and when it is not appropriate to use it. I also
+ removed the part of the debug message that mentions that this is
+ probably a bug because there are some perfectly legitimate places
+ where ast_sched_del may fail to delete an entry (e.g. when the
+ scheduler callback manually reschedules with a new id instead of
+ returning non-zero to tell the scheduler to reschedule with the
+ same idea). I also raised the debug level of the debug message in
+ AST_SCHED_DEL since it seems like it could come up quite
+ frequently since the macro is probably being used in several
+ places where it shouldn't be. Also removed the redundant line,
+ file, and function information since that is provided by ast_log.
+
+2008-03-12 19:57 +0000 [r108135] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c, main/channel.c: (closes issue #12187,
+ reported by atis, fixed by me after some brainstorming on the
+ issue with mmichelson) - Update copyright info on app_chanspy. -
+ Fix a race condition that caused app_chanspy to crash. The issue
+ was that the chanspy datastore magic that was used to ensure that
+ spyee channels did not disappear out from under the code did not
+ completely solve the problem. It was actually possible for
+ chanspy to acquire a channel reference out of its datastore to a
+ channel that was in the middle of being destroyed. That was
+ because datastore destruction in ast_channel_free() was done near
+ the end. So, this left the code in app_chanspy accessing a
+ channel that was partially, or completely invalid because it was
+ in the process of being free'd by another thread. The following
+ sort of shows the code path where the race occurred:
+ =============================================================================
+ Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy)
+ --------------------------------------||-------------------------------------
+ ast_channel_free() || - remove channel from channel list || -
+ lock/unlock the channel to ensure || that no references retrieved
+ from || the channel list exist. ||
+ --------------------------------------||-------------------------------------
+ || channel_spy() - destroy some channel data || - Lock chanspy
+ datastore || - Retrieve reference to channel || - lock channel ||
+ - Unlock chanspy datastore
+ --------------------------------------||-------------------------------------
+ - destroy channel datastores || - call chanspy datastore d'tor ||
+ which NULL's out the ds' || - Operate on the channel ...
+ reference to the channel || || - free the channel || || || -
+ unlock the channel
+ --------------------------------------||-------------------------------------
+ =============================================================================
+
+2008-03-12 19:16 +0000 [r108086] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: if we receive an INVITE with a
+ Content-Length that is not a valid number, or is zero, then don't
+ process the rest of the message body looking for an SDP closes
+ issue #11475 Reported by: andrebarbosa
+
+2008-03-12 18:26 +0000 [r108083] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_mixmonitor.c, include/asterisk/audiohook.h,
+ main/audiohook.c: Add a trigger mode that triggers on both read
+ and write. The actual function that returns the combined audio
+ frame though will wait until both sides have fed in audio, or
+ until one side stops (such as the case when you call Wait).
+ (closes issue #11945) Reported by: xheliox
+
+2008-03-12 16:59 +0000 [r108031] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Destroy the channel lock after the channel
+ datastores. (inspired by issue #12187)
+
+2008-03-12 01:52 +0000 [r107877] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/iax-friends.sql, contrib/scripts/sip-friends.sql:
+ Document all of the possible realtime fields
+
+2008-03-11 23:37 +0000 [r107714-107826] Jason Parker <jparker@digium.com>
+
+ * doc/voicemail_odbc_postgresql.txt: Update documentation for pgsql
+ ODBC voicemail. (closes issue #12186) Reported by: jsmith
+ Patches: vm_pgsql_doc_update.patch uploaded by jsmith (license
+ 15)
+
+ * channels/chan_gtalk.c: Copy voicemail dependency logic for
+ res_adsi to chan_gtalk (for jabber). (closes issue #12014)
+ Reported by: junky
+
+2008-03-11 20:48 +0000 [r107713] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile.rules, channels/Makefile: get chan_vpb to build properly
+ in dev mode
+
+2008-03-11 20:47 +0000 [r107712] Jason Parker <jparker@digium.com>
+
+ * apps/app_voicemail.c: Add a newline on a log
+
+2008-03-11 19:20 +0000 [r107582-107646] Joshua Colp <jcolp@digium.com>
+
+ * res/res_features.c: Make sure the visible indication is on the
+ right channel so when the masquerade happens the proper
+ indication is enacted. (closes issue #11707) Reported by: iam
+
+ * apps/app_meetme.c: Add an additional check for setting conference
+ parameter when using the marked user options. It was possible for
+ it to return to a no listen/no talk state if a masquerade
+ happened. (closes issue #12136) Reported by: aragon
+
+ * apps/app_exec.c: Fix a minor spelling error. (closes issue
+ #12183) Reported by: darrylc
+
+2008-03-11 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.19-rc2 released.
+
+2008-03-11 15:18 +0000 [r107352-107472] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_rpt.c: backport a fix from trunk
+
+ * channels/misdn/isdn_lib.c, codecs/Makefile,
+ channels/chan_misdn.c: fix various other problems found by gcc
+ 4.3
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ apps/app_sms.c: stop checking for mktime() in the configure
+ script... we don't use it, and the test is buggy under gcc 4.3
+
+ * configure, main/Makefile, configure.ac, makeopts.in: check for
+ compiler support for -fno-strict-overflow before using it (tested
+ with Debian's gcc 4.3, 4.1 and 3.4) (closes issue #12179)
+ Reported by: Netview
+
+ * configure, configure.ac: fix small bug in IMAP toolkit testing
+
+ * main/udptl.c, utils/Makefile, main/Makefile,
+ main/editline/readline.c, pbx/Makefile: fix up various compiler
+ warnings found with gcc-4.3: - the output of flex includes a
+ static function called 'input' that is not used, so for the
+ moment we'll stop having the compiler tell us about unused
+ variables in the flex source files (a better fix would be to
+ improve our flex post-processing to remove the unused function) -
+ main/stdtime/localtime.c makes assumptions about signed integer
+ overflow, and gcc-4.3's improved optimizer tries to take
+ advantage of handling potential overflow conditions at compile
+ time; for now, suppress these optimizations until we can fiure
+ out if the code needs improvement - main/udptl.c has some
+ references to uninitialized variables; in one case there was no
+ bug, but in the other it was certainly possibly for unexpected
+ behavior to occur - main/editline/readline.c had an unused
+ variable
+
+2008-03-11 00:59 +0000 [r107290] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: If we fail to alloc a channel, we should
+ re-lock the pvt structure before returning.
+
+2008-03-10 21:32 +0000 [r107230] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Use non-global storage for eswitch
+
+2008-03-10 20:27 +0000 [r107173] Jason Parker <jparker@digium.com>
+
+ * channels/chan_zap.c: Make sure to reenable echo can after a
+ "failed" (canceled, etc) three-way call. (closes issue #11335)
+ Reported by: rebuild
+
+2008-03-10 20:17 +0000 [r107099-107161] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c: Fix another bug specifically related to asynchronous
+ call origination. Once the PBX is started on the channel using
+ ast_pbx_start(), then the ownership of the channel has been
+ passed on to another thread. We can no longer access it in this
+ code. If the channel gets hung up very quickly, it is possible
+ that we could access a channel that has been free'd. (inspired by
+ BE-386)
+
+ * main/pbx.c: Fix some bugs related to originating calls. If the
+ code failed to start a PBX on the channel (such as if you set a
+ call limit based on the system's load average), then there were
+ cases where a channel that has already been free'd using
+ ast_hangup() got accessed. This caused weird memory corruption
+ and crashes to occur. (fixes issue BE-386) (much debugging credit
+ goes to twilson, final patch written by me)
+
+ * main/channel.c: Resolve a compiler warning.
+
+ * main/channel.c: Fix a race condition where the generator can go
+ away (closes issue #12175, reported by edantie, patched by me)
+
+2008-03-10 14:33 +0000 [r107016] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c, main/cdr.c, include/asterisk/cdr.h: Move where
+ unanswered CDRs are dropped to the CDR core, not everything uses
+ app_dial. (closes issue #11516) Reported by: ys Patches:
+ branch_1.4_cdr.diff uploaded by ys (license 281) Tested by:
+ anest, jcapp, dartvader
+
+2008-03-08 15:59 +0000 [r106945] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c: don't generate D-Channel "up" and "down"
+ messages unless the channel state is actually changing; also,
+ generate the "up" message when an implicit "up" occurs due to
+ reception of a normal event when we thought the channel was
+ "down"
+
+2008-03-07 22:51 +0000 [r106895] Russell Bryant <russell@digium.com>
+
+ * apps/app_meetme.c: Only start the SLA thread if SLA has actually
+ been configured.
+
+2008-03-07 22:14 +0000 [r106842] Jason Parker <jparker@digium.com>
+
+ * main/editline/Makefile.in: Fix hardcoded grep in editline, were
+ GNU grep is required. (closes issue #12124) Reported by: dmartin
+
+2008-03-07 19:32 +0000 [r106788] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c: Ignore source update control frame. (closes issue
+ #12168) Reported by: plack
+
+2008-03-07 17:16 +0000 [r106704] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/sched.h: Change a warning message to a debug
+ message. This is happening quite frequently, and it is not worth
+ spamming users with these messages unless we are pretty confident
+ that it should never happen. As it stands today, it _will_ and
+ _does_ happen and until that gets cleaned up a reasonable amount
+ on the development side, let's not spam the logs of everyone
+ else. (closes issue #12154)
+
+2008-03-07 16:22 +0000 [r106552-106635] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Warn the user when a temporary greeting
+ exists (Closes issue #11409)
+
+ * main/rtp.c: Properly initialize rtp->schedid (Closes issue
+ #12154)
+
+ * apps/app_chanspy.c, apps/app_rpt.c, main/asterisk.c,
+ apps/app_speech_utils.c, apps/app_voicemail.c, main/channel.c,
+ funcs/func_enum.c, channels/chan_misdn.c, main/frame.c,
+ main/manager.c: Safely use the strncat() function. (closes issue
+ #11958) Reported by: norman Patches: 20080209__bug11958.diff.txt
+ uploaded by Corydon76 (license 14)
+
+2008-03-06 22:10 +0000 [r106437] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c: Quell an annoying message that is likely to print
+ every single time that ast_pbx_outgoing_app is called. The reason
+ is that __ast_request_and_dial allocates the cdr for the channel,
+ so it should be expected that the channel will have a cdr on it.
+ Thanks to joetester on IRC for pointing this out
+
+2008-03-06 04:40 +0000 [r106328] Tilghman Lesher <tlesher@digium.com>
+
+ * sounds/Makefile: Upgrade to the next release of sounds
+
+2008-03-05 22:37 +0000 [r106237] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Fix a potential deadlock and a few
+ different potential crashes. (closes issue #12145, reported by
+ thiagarcia, patched by me)
+
+2008-03-05 22:32 +0000 [r106235] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_oss.c, main/rtp.c, channels/chan_mgcp.c,
+ apps/app_dial.c, main/channel.c, channels/chan_phone.c,
+ main/dial.c, channels/chan_zap.c, channels/chan_sip.c,
+ channels/chan_skinny.c, channels/chan_h323.c, main/file.c,
+ channels/chan_alsa.c, apps/app_followme.c,
+ include/asterisk/frame.h: Add a control frame to indicate the
+ source of media has changed. Depending on the underlying
+ technology it may need to change some things. (closes issue
+ #12148) Reported by: jcomellas
+
+2008-03-05 21:12 +0000 [r106178] Michiel van Baak <michiel@vanbaak.info>
+
+ * doc/realtime.txt: document var_metric so no bugreports will come
+ in when it's actually a configuration issue. (issue #12151)
+ Reported and patched by: caio1982 1.4 patch by me
+
+2008-03-05 15:32 +0000 [r106038] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c: when a PRI call must be moved to a different
+ B channel at the request of the other endpoint, ensure that any
+ DSP active on the original channel is moved to the new one
+ (closes issue #11917) Reported by: mavetju Tested by: mavetju
+
+2008-03-05 15:17 +0000 [r106015] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c, include/asterisk/sched.h: Correctly
+ initialize retransid in SIP, and ensure that the warning when
+ failing to delete a schedule entry can actually hit the log.
+ (closes issue #12140) Reported by: slavon Patches: sch2.patch
+ uploaded by slavon (license 288) (Patch slightly modified by me)
+
+2008-03-05 01:52 +0000 [r105932] Russell Bryant <russell@digium.com>
+
+ * main/rtp.c, main/translate.c, include/asterisk/frame.h: Fix a bug
+ that I just noticed in the RTP code. The calculation for setting
+ the len field in an ast_frame of audio was wrong when G.722 is in
+ use. The len field represents the number of ms of audio that the
+ frame contains. It would have set the value to be twice what it
+ should be.
+
+2008-03-04 18:10 +0000 [r105674-105676] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: In addition to setting the marker bit let's change
+ our ssrc so they know for sure it is a different source.
+
+ * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: When a
+ new source of audio comes in (such as music on hold) make sure
+ the marker bit gets set. (closes issue #10355) Reported by:
+ wdecarne Patches: 10355.diff uploaded by file (license 11)
+ (closes issue #11491) Reported by: kanderson
+
+2008-03-04 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.19-rc1 released.
+
+2008-03-04 04:31 +0000 [r105591] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c: Backport a minor bug fix from trunk that I found
+ while doing random code cleanup. Properly break out of the loop
+ when a context isn't found when verify that includes are valid.
+
+2008-03-03 18:06 +0000 [r105572] Jason Parker <jparker@digium.com>
+
+ * res/snmp/agent.c: Fix type for astNumChannels. (closes issue
+ #12114) Reported by: jeffg Patches: 12114.patch uploaded by jeffg
+ (license 192)
+
+2008-03-03 17:16 +0000 [r105563-105570] Russell Bryant <russell@digium.com>
+
+ * channels/chan_local.c: In the case of an ast_channel allocation
+ failure, take the local_pvt out of the pvt list before destroying
+ it.
+
+ * channels/chan_local.c: Fix a potential memory leak of the
+ local_pvt struct when ast_channel allocation fails. Also, in
+ passing, centralize the code necessary to destroy a local_pvt.
+
+ * main/autoservice.c: Update the copyright information for
+ autoservice. Most of the code in this file now is stuff that I
+ have written recently ...
+
+ * main/asterisk.c, main/channel.c, include/asterisk.h,
+ main/autoservice.c: Merge in some changes from
+ team/russell/autoservice-nochans-1.4 These changes fix up some
+ dubious code that I came across while auditing what happens in
+ the autoservice thread when there are no channels currently in
+ autoservice. 1) Change it so that autoservice thread doesn't keep
+ looping around calling ast_waitfor_n() on 0 channels twice a
+ second. Instead, use a thread condition so that the thread
+ properly goes to sleep and does not wake up until a channel is
+ put into autoservice. This actually fixes an interesting bug, as
+ well. If the autoservice thread is already running (almost always
+ is the case), then when the thread goes from having 0 channels to
+ have 1 channel to autoservice, that channel would have to wait
+ for up to 1/2 of a second to have the first frame read from it.
+ 2) Fix up the code in ast_waitfor_nandfds() for when it gets
+ called with no channels and no fds to poll() on, such as was the
+ case with the previous code for the autoservice thread. In this
+ case, the code would call alloca(0), and pass the result as the
+ first argument to poll(). In this case, the 2nd argument to
+ poll() specified that there were no fds, so this invalid pointer
+ shouldn't actually get dereferenced, but, this code makes it
+ explicit and ensures the pointers are NULL unless we have valid
+ data to put there. (related to issue #12116)
+
+2008-03-03 15:28 +0000 [r105557-105560] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c: It is possible for no audio to pass between the
+ current digit and next digit so expand logic that clears
+ emulation to AST_FRAME_NULL. (closes issue #11911) Reported by:
+ edgreenberg Patches: v1-11911.patch uploaded by dimas (license
+ 88) Tested by: tbsky
+
+ * channels/chan_sip.c: Add a comment to describe some logic.
+ (closes issue #12120) Reported by: flefoll Patches:
+ chan_sip.c.br14.patch-just-a-comment uploaded by flefoll (license
+ 244)
+
+2008-02-29 23:34 +0000 [r105409] Russell Bryant <russell@digium.com>
+
+ * main/autoservice.c: Fix a major bug in autoservice. There was a
+ race condition in the handling of the list of channels in
+ autoservice. The problem was that it was possible for a channel
+ to get removed from autoservice and destroyed, while the
+ autoservice thread was still messing with the channel. This led
+ to memory corruption, and caused crashes. This explains multiple
+ backtraces I have seen that have references to autoservice, but
+ do to the nature of the issue (memory corruption), could cause
+ crashes in a number of areas. (fixes the crash in BE-386) (closes
+ issue #11694) (closes issue #11940) The following issues could be
+ related. If you are the reporter of one of these, please update
+ to include this fix and try again. (potentially fixes issue
+ #11189) (potentially fixes issue #12107) (potentially fixes issue
+ #11573) (potentially fixes issue #12008) (potentially fixes issue
+ #11189) (potentially fixes issue #11993) (potentially fixes issue
+ #11791)
+
+2008-02-29 14:47 +0000 [r105326] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * res/res_jabber.c: Fix a potential memory leak
+
+2008-02-29 14:34 +0000 [r105296] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: If the message file does not exist, just
+ return harmlessly, instead of crashing. (Closes issue #12108)
+
+2008-02-29 13:48 +0000 [r105261] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_voicemail.c: Bump up the size of the uniqueid variable.
+ (closes issue #12107) Reported by: asgaroth
+
+2008-02-29 13:05 +0000 [r105209] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * res/res_jabber.c: Automatically create new buddy upon reception
+ of a presence stanza of type subscribed. (closes issue #12066)
+ Reported by: ffadaie Patches: branch-1.4-12066-1.diff uploaded by
+ phsultan (license 73) trunk-12066-1.diff uploaded by phsultan
+ (license 73) Tested by: ffadaie, phsultan
+
+2008-02-28 22:23 +0000 [r105116] Russell Bryant <russell@digium.com>
+
+ * main/utils.c, include/asterisk/lock.h: Fix a bug in the lock
+ tracking code that was discovered by mmichelson. The issue is
+ that if the lock history array was full, then the functions to
+ mark a lock as acquired or not would adjust the stats for
+ whatever lock is at the end of the array, which may not be
+ itself. So, do a sanity check to make sure that we're updating
+ lock info for the proper lock. (This explains the bizarre stats
+ on lock #63 in BE-396, thanks Mark!)
+
+2008-02-28 21:56 +0000 [r105113] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/rc.debian.asterisk: Update init script for LSB
+ compat (closes issue #9843) Reported by: ibc Patches:
+ rc.debian.asterisk.patch uploaded by ibc (license 211) Tested by:
+ paravoid
+
+2008-02-28 20:11 +0000 [r105059] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: When using autofill, members who are in use
+ should be counted towards the number of available members to call
+ if ringinuse is set to yes. Thanks to jmls who brought this issue
+ up on IRC
+
+2008-02-28 19:20 +0000 [r104920-105005] Jason Parker <jparker@digium.com>
+
+ * main/cdr.c, main/pbx.c: Make pbx_exec pass an empty string into
+ applications, if we get NULL. This protects against possible
+ segfaults in applications that may try to use data before
+ checking length (ast_strdupa'ing it, for example) (closes issue
+ #12100) Reported by: foxfire Patches: 12100-nullappargs.diff
+ uploaded by qwell (license 4)
+
+ * channels/chan_skinny.c: According to a video at www.cisco.com,
+ the 7921G supports 6 line appearances.
+
+2008-02-28 00:05 +0000 [r104868] Tilghman Lesher <tlesher@digium.com>
+
+ * main/Makefile, build_tools/strip_nonapi: Compatibility fix for
+ PPC64 (closes issue #12081) Reported by: jcollie Patches:
+ asterisk-1.4.18-funcdesc.patch uploaded by jcollie (license 412)
+ Tested by: jcollie, Corydon76
+
+2008-02-27 21:49 +0000 [r104841] Mark Michelson <mmichelson@digium.com>
+
+ * main/dial.c: Two fixes: 1. Make the list of ast_dial_channels a
+ lockable list. This is because in some cases, the ast_dial may
+ exist in multiple threads due to asynchronous execution of its
+ application, and I found some cases where race conditions could
+ exist. 2. Check in ast_dial_join to be sure that the channel
+ still exists before attempting to lock it, since it could have
+ gotten hung up but the is_running_app flag on the
+ ast_dial_channel may not have been cleared yet. (closes issue
+ #12038) Reported by: jvandal Patches: 12038v2.patch uploaded by
+ putnopvut (license 60) Tested by: jvandal
+
+2008-02-27 20:56 +0000 [r104787] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_chanspy.c: Don't loop around infinitely trying to spy on
+ our own channel, and don't forget to free/detach the datastore
+ upon hangup of the spy.
+
+2008-02-27 20:36 +0000 [r104783] Mark Michelson <mmichelson@digium.com>
+
+ * main/file.c: Bump a couple of more buffers up by 2 so that
+ annoying warnings aren't generated like crazy on every
+ fileexists_core call.
+
+2008-02-27 18:15 +0000 [r104704] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c: Ensure the session ID can't be 0.
+
+2008-02-27 17:41 +0000 [r104665] Joshua Colp <jcolp@digium.com>
+
+ * main/file.c: Bump up the buffer by 2.
+
+2008-02-27 17:33 +0000 [r104625] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c: Fix a problem in ChanSpy where it could get
+ stuck in an infinite loop without being able to detect that the
+ calling channel hung up. (closes issue #12076, reported by junky,
+ patched by me)
+
+2008-02-27 17:26 +0000 [r104598] Jason Parker <jparker@digium.com>
+
+ * res/res_features.c: Inherit language from the transfering channel
+ on a blind transfer. (closes issue #11682) Reported by: caio1982
+ Patches: local_atxfer_lang3-1.4.diff uploaded by caio1982
+ (license 22) Tested by: caio1982, victoryure
+
+2008-02-27 17:07 +0000 [r104596] Joshua Colp <jcolp@digium.com>
+
+ * main/loader.c: Use the lock (which already existed, it just
+ wasn't used) on the updaters list to protect the contents instead
+ of the overall module list lock. (closes issue #12080) Reported
+ by: ChaseVenters
+
+2008-02-27 16:53 +0000 [r104593] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/file.c: fallback to standard English prompts properly when
+ using new prompt directory layout (closes issue #11831) Reported
+ by: IgorG Patches: fallbacken.v1.diff uploaded by IgorG (license
+ 20) (modified by me to improve code and conform rest of function
+ to coding guidelines)
+
+2008-02-27 16:45 +0000 [r104591] Russell Bryant <russell@digium.com>
+
+ * channels/chan_zap.c: When we receive a known alarm, make sure
+ that the unknown alarm flag is not still set to make sure that
+ when we come back out of alarm, it gets reported in the log and
+ manager interface (after discussion with tzafrir on the -dev
+ list)
+
+2008-02-27 15:52 +0000 [r104536] Joshua Colp <jcolp@digium.com>
+
+ * res/res_smdi.c: Only stop the MWI monitor thread if it was
+ actually started. (closes issue #12086) Reported by: francesco_r
+
+2008-02-27 01:15 +0000 [r104332-104334] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c: Avoid some recursion in the cleanup code for
+ the chanspy datastore (closes issue #12076, reported by junky,
+ patched by me)
+
+ * channels/chan_zap.c: Zaptel 1.4 now exposes FXO battery state as
+ an alarm. However, Asterisk 1.4 does not know what to do with
+ these alarms. Only Asterisk 1.6 cares about it. So, if we get an
+ unknown alarm in chan_zap, don't generate confusing log messages
+ about it.
+
+2008-02-26 18:26 +0000 [r104132-104141] Jason Parker <jparker@digium.com>
+
+ * Makefile: Add badshell to .PHONY target (thanks Kevin)
+
+ * Makefile: Since all shells aren't as awesome as bash, we have to
+ fail if somebody tries to use a literal "~" in DESTDIR.
+
+ * sounds/Makefile: Revert previous abspath change. ...abspath is
+ new in GNU make 3.81. I feel so...defeated. Must find new fix!
+
+ * sounds/Makefile: Fix a very bizarre issue we were seeing with our
+ buildbot when using a DESTDIR that wasn't an absolute path (such
+ as DESTDIR=~/asterisk-1.4). Apparently what was happening, was
+ that some of the targets were being expanded to the full path, so
+ $@ ended up being /root/asterisk-1.4/[...]/ rather than
+ ~/asterisk-1.4/[...]/ It appears that this may be a new "feature"
+ in GNU make. (*cough*
+ http://en.wikipedia.org/wiki/Principle_of_least_surprise *cough*)
+
+2008-02-26 00:25 +0000 [r104119] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/smdi.h, apps/app_voicemail.c,
+ channels/chan_zap.c, res/res_smdi.c, configs/smdi.conf.sample:
+ Merge changes from team/russell/smdi-1.4 This commit brings in a
+ significant set of changes to the SMDI support in Asterisk. There
+ were a number of bugs in the current implementation, most notably
+ being that it was very likely on busy systems to pop off the
+ wrong message from the SMDI message queue. So, this set of
+ changes fixes the issues discovered as well as introducing some
+ new ways to use the SMDI support which are required to avoid the
+ bugs with grabbing the wrong message off of the queue. This code
+ introduces a new interface to SMDI, with two dialplan functions.
+ First, you get an SMDI message in the dialplan using
+ SMDI_MSG_RETRIEVE() and then you access details in the message
+ using the SMDI_MSG() function. A side benefit of this is that it
+ now supports more than just chan_zap. For example, with this
+ implementation, you can have some FXO lines being terminated on a
+ SIP gateway, but the SMDI link in Asterisk. Another issue with
+ the current implementation is that it is quite common that the
+ station ID that comes in on the SMDI link is not necessarily the
+ same as the Asterisk voicemail box. There are now additional
+ directives in the smdi.conf configuration file which let you map
+ SMDI station IDs to Asterisk voicemail boxes. Yet another issue
+ with the current SMDI support was related to MWI reporting over
+ the SMDI link. The current code could only report a MWI change
+ when the change was made by someone calling into voicemail. If
+ the change was made by some other entity (such as with IMAP
+ storage, or with a web interface of some kind), then the MWI
+ change would never be sent. The SMDI module can now poll for MWI
+ changes if configured to do so. This work was inspired by and
+ primarily done for the University of Pennsylvania. (also related
+ to issue #9260)
+
+2008-02-26 00:03 +0000 [r104111] Jason Parker <jparker@digium.com>
+
+ * channels/chan_h323.c: IPTOS_MINCOST is not defined on Solaris.
+ (closes issue #12050) Reported by: asgaroth Patches: 12050.patch
+ uploaded by putnopvut (license 60)
+
+2008-02-25 23:42 +0000 [r104102-104106] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c: This patch fixes some pretty significant
+ problems with how app_chanspy handles pointers to channels that
+ are being spied upon. It was very likely that a crash would occur
+ if the channel being spied upon hung up. This was because the
+ current ast_channel handling _requires_ that the object is locked
+ or else it could disappear at any time (except in the owning
+ channel thread). So, this patch uses some channel datastore magic
+ on the spied upon channel to be able to detect if and when the
+ channel goes away. (closes issue #11877) (patch written by me,
+ but thanks to kpfleming for the idea, and to file for review)
+
+ * main/utils.c: Improve the lock tracking code a bit so that a
+ bunch of old locks that threads failed to lock don't sit around
+ in the history. When a lock is first locked, this checks to see
+ if the last lock in the list was one that was failed to be
+ locked. If it is, then that was a lock that we're no longer
+ sitting in a trylock loop trying to lock, so just remove it.
+ (inspired by issue #11712)
+
+2008-02-25 21:37 +0000 [r104095] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Make it so a users.conf user creates both a
+ SIP peer and a SIP user. The user will be used for inbound
+ authentication for the device, and peer will be used for placing
+ calls to the device. (closes issue #9044) Reported by: queuetue
+ Patches: sip-gui-friend.diff uploaded by qwell (license 4)
+
+2008-02-25 21:31 +0000 [r104094] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: If the destination folder is full, don't
+ delete a message when exiting. (closes issue #12065) Reported by:
+ selsky Patch by: (myself)
+
+2008-02-25 20:49 +0000 [r104092] Jason Parker <jparker@digium.com>
+
+ * main/config.c: Allow the use of #include and #exec in situations
+ where the max include depth was only 1. Specifically, this fixes
+ using #include and #exec in extconfig.conf. This was basically
+ caused because the config file itself raises the include level to
+ 1. I opted not to raise the include limit, because recursion here
+ could cause very bizarre behavior. Pointed out, and tested by
+ jmls (closes issue #12064)
+
+2008-02-25 18:38 +0000 [r104086] Russell Bryant <russell@digium.com>
+
+ * channels/chan_agent.c: Ensure that the channel doesn't disappear
+ in agent_logoff(). If it does, it could cause a crash. (fixes the
+ crash reported in BE-396)
+
+2008-02-25 16:16 +0000 [r104082-104084] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: If a resubscription comes in for a dialog we
+ no longer know about tell the remote side that the dialog does
+ not exist so they subscribe again using a new dialog. (closes
+ issue #10727) Reported by: s0l4rb03 Patches: 10727-2.diff
+ uploaded by file (license 11)
+
+ * channels/chan_sip.c: Due to recent changes tag will no longer be
+ NULL if not present so we have to use ast_strlen_zero to see if
+ it's actually blank. (closes issue #12061) Reported by: flefoll
+ Patches: chan_sip.c.br14.patch_pedantic_no_totag uploaded by
+ flefoll (license 244)
+
+2008-02-22 22:45 +0000 [r104037] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Backwards debug message. (closes issue
+ #12052) Reported by: flefoll Patches:
+ chan_sip.c.br14.patch_found-notfound uploaded by flefoll (license
+ 244)
+
+2008-02-21 21:05 +0000 [r104026-104027] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_zap.c: And as a followup to revision 104026,
+ completely remove event-related calls from a section of code
+ where we know there was no event to handle or get.
+
+ * channels/chan_zap.c: Remove an incorrect debug message. It
+ reported that it had received a specific event and tried to
+ report which event was received. What actually was happening was
+ that it was reporting the number of bytes returned from a call to
+ read(). Thanks to Jared Smith for bringing the issue up on IRC
+
+2008-02-21 14:33 +0000 [r104015] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/manager.c: reduce the likelihood that HTTP Manager session
+ ids will consist of primarily '1' bits
+
+2008-02-20 22:32 +0000 [r103956] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Clear up confusion when viewing the
+ QUEUE_WAITING_COUNT of a "dead" realtime queue. Since from the
+ user's perspective, the queue does exist, we shouldn't tell them
+ we couldn't find the queue. Instead since it is a dead queue,
+ report a 0 waiting count This issue was brought up on IRC by jmls
+
+2008-02-20 22:06 +0000 [r103953] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_zap.c: Don't wait for additional digits when
+ overlap dialing is enabled if the setup message contains the
+ sending_complete information element. (closes issue #11785)
+ Reported by: klaus3000 Patches:
+ sending_complete_overlap_asterisk-1.4.17.patch.txt uploaded by
+ klaus3000 (license 65)
+
+2008-02-20 21:40 +0000 [r103904] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_local.c: Fix a crash if the channel becomes NULL
+ while attempting to lock it. (closes issue #12039) Reported by:
+ danpwi
+
+2008-02-20 17:53 +0000 [r103845] Tilghman Lesher <tlesher@digium.com>
+
+ * main/stdtime/localtime.c: Compat fix for Solaris (closes issue
+ #12022) Reported by: asgaroth Patches:
+ 20080219__bug12022.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: asgaroth
+
+2008-02-19 20:28 +0000 [r103823] Joshua Colp <jcolp@digium.com>
+
+ * channels/h323/ast_h323.cxx: Send CallerID Name in setup message.
+ (closes issue #11241) Reported by: tusar Patches:
+ h323id_as_callerid_name.patch uploaded by tusar (license 344)
+
+2008-02-19 20:02 +0000 [r103821] Russell Bryant <russell@digium.com>
+
+ * channels/chan_local.c: Account for the fact that the "other"
+ channel can disappear while the local pvt is not locked. (fixes a
+ problem introduced in rev 100581) (closes issue #12012) Reported
+ by: stevedavies Patch by me
+
+2008-02-19 17:31 +0000 [r103807-103812] Joshua Colp <jcolp@digium.com>
+
+ * configure, configure.ac: Don't look for launchd when cross
+ compiling. (closes issue #12029) Reported by: ovi
+
+ * channels/chan_sip.c: Fix building of chan_sip.
+
+2008-02-19 10:27 +0000 [r103806] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Make sure we send error replies correctly by
+ checking the via header.
+
+2008-02-18 23:56 +0000 [r103801] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c: Ensure that emulated DTMFs do not get interrupted
+ by another begin frame. (closes issue #11740) Reported by: gserra
+ Patches: v1-11740.patch uploaded by dimas (license 88) (closes
+ issue #11955) Reported by: tsearle (closes issue #10530) Reported
+ by: xmarksthespot
+
+2008-02-18 22:28 +0000 [r103790-103795] Jason Parker <jparker@digium.com>
+
+ * channels/chan_zap.c: Fix previous commit so that we actually
+ disable echocanbridged if echocancel is off.
+
+ * channels/chan_zap.c: Correct a message when echocancelwhenbridged
+ is on, but echocancel is not. Issue #12019
+
+2008-02-18 20:52 +0000 [r103786] Mark Michelson <mmichelson@digium.com>
+
+ * main/app.c: There was an invalid assumption when calculating the
+ duration of a file that the filestream in question was created
+ properly. Unfortunately this led to a segfault in the situation
+ where an unknown format was specified in voicemail.conf and a
+ voicemail was recorded. Now, we first check to be sure that the
+ stream was written correctly or else assume a zero duration.
+ (closes issue #12021) Reported by: jakep Tested by: putnopvut
+
+2008-02-18 17:31 +0000 [r103780] Tilghman Lesher <tlesher@digium.com>
+
+ * main/rtp.c, channels/chan_sip.c: When a SIP channel is being
+ auto-destroyed, it's possible for it to still be in bridge code.
+ When that happens, we crash. Delay the RTP destruction until the
+ bridge is ended. (closes issue #11960) Reported by: norman
+ Patches: 20080215__bug11960__2.diff.txt uploaded by Corydon76
+ (license 14) Tested by: norman
+
+2008-02-18 16:37 +0000 [r103770] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_zap.c: Fix a linked list corruption that under the
+ right circumstances could lead to a looped list, meaning it will
+ traverse forever. (closes issue #11818) Reported by: michael-fig
+ Patches: 11818.patch uploaded by putnopvut (license 60) Tested
+ by: michael-fig
+
+2008-02-18 16:11 +0000 [r103763-103768] Joshua Colp <jcolp@digium.com>
+
+ * main/asterisk.c: Backport fix from issue #9325. (closes issue
+ #11980) Reported by: rbrunka
+
+ * channels/chan_sip.c: Don't care if the extension given doesn't
+ exist for subscription based MWI.
+
+2008-02-15 23:31 +0000 [r103726-103741] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Fix a crash in chan_iax2 due to a race
+ condition (closes issue #11780) Reported by: guillecabeza
+ Patches: bug_iax2_jb_1.4.patch uploaded by guillecabeza (license
+ 380) bug_iax2_jb_trunk.patch uploaded by guillecabeza (license
+ 380)
+
+ * main/loader.c: In the case that you try to directly reload a
+ module has returned AST_MODULE_LOAD_DECLINE, log a message
+ indicating that the module is not fully initialized and must be
+ initialized using "module load".
+
+ * main/loader.c: Don't attempt to execute the reload callback for a
+ module that returned AST_MODULE_LOAD_DECLINE. This fixes a crash
+ that was reported against chan_console in trunk. (closes issue
+ #11953, reported by junky, fixed by me)
+
+2008-02-15 17:26 +0000 [r103688-103722] Mark Michelson <mmichelson@digium.com>
+
+ * doc/imapstorage.txt, configure, configure.ac: Final round of
+ changes for configure script logic for IMAP Now if a directory is
+ specified, then we will search that directory for a source
+ installation of the IMAP toolkit. If none is found, then we will
+ use that directory as the basis for detecting a package
+ installation of the IMAP c-client. If that check fails, then
+ configure will fail.
+
+ * configure, configure.ac: Fix a bit of wrong logic in the
+ configure script that caused problems when trying to configure
+ without IMAP. Patch suggestion from phsultan, but I modified it
+ slightly. (closes issue #12003) Reported by: pj Tested by:
+ putnopvut
+
+ * doc/imapstorage.txt, configure, configure.ac: I apparently
+ misunderstood one of the requirements of this configure change.
+ Now, if a source directory is specified with the --with-imap
+ option, and a valid source installation is not detected there,
+ then configure will fail and will not check for a package
+ installation.
+
+ * doc/imapstorage.txt: Make a small clarification in the
+ documentation
+
+ * doc/imapstorage.txt: Update documentation regarding configuration
+ of IMAP
+
+ * apps/app_voicemail.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Change to the
+ configure logic regarding IMAP. Prior to this commit, if you
+ wished to configure Asterisk with IMAP support, you would use the
+ --with-imap configure switch in one of the following two ways:
+ --with-imap=/some/directory would look in the directory specified
+ for a UW IMAP source installation --with-imap would assume that
+ you had imap-2004g installed in .. relative to the Asterisk
+ source With this set of changes the two above options still work
+ the same, but there are two new behaviors, too.
+ --with-imap=system will assume that you have -libc-client.so
+ where you store your shared objects and will attempt to find
+ c-client headers in your include path either in the imap or
+ c-client directory. If either of the two original methods of
+ specifying the imap option should fail, then the check for
+ --with-imap =system will be performed in addition. It is only
+ after this "system" check that failure can happen.
+
+ * apps/app_voicemail.c: Fix build for non-IMAP builds
+
+ * apps/app_voicemail.c: Fix the new message count if delete=yes
+ when using IMAP storage. (closes issue #11406) Reported by:
+ jaroth Patches: deleteflag_v2.patch uploaded by jaroth (license
+ 50) Tested by: jaroth
+
+2008-02-14 19:51 +0000 [r103683-103684] Jason Parker <jparker@digium.com>
+
+ * funcs/func_cdr.c: swap location for this..
+
+ * funcs/func_cdr.c: Document the 'l' option to the CDR() function.
+ (Thanks voipgate for pointing out the option, and Leif for
+ providing text for it.) Closes issue #11695.
+
+2008-02-13 06:25 +0000 [r103556-103607] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_agent.c: We aren't talking to ourselves; we're
+ talking to someone else. (closes issue #11771) Reported by:
+ msetim Patches: ami_agent_talkingto-1.4.diff uploaded by caio1982
+ (license 22) Tested by: caio1982, msetim
+
+ * apps/app_voicemail.c: Refuse to load app_voicemail if res_adsi is
+ not loaded (which is a symbol dependency) (closes issue #11760)
+ Reported by: non-poster Patches: 20080114__bug11760.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: Corydon76,
+ non-poster, jamesgolovich
+
+2008-02-12 22:24 +0000 [r103503-103504] Jason Parker <jparker@digium.com>
+
+ * main/asterisk.c: revert accidental change from last commit. oops
+
+ * contrib/scripts/safe_asterisk, main/asterisk.c: Remove condition
+ that was impossible.
+
+2008-02-12 15:09 +0000 [r103324-103385] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Even if no CallerID name or number has been
+ provided by the remote party still use the configured sip.conf
+ ones. (closes issue #11977) Reported by: pj
+
+ * apps/app_meetme.c: If entering a conference with the 'w' option
+ ensure that we can't listen or speak until the marked user
+ appears. (closes issue #11835) Reported by: alanmcmillan
+
+2008-02-11 17:05 +0000 [r103315] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configs/zapata.conf.sample: improve 2BCT documentation a bit
+ (thanks Jared)
+
+2008-02-09 06:23 +0000 [r103197] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Commit fix for being unable to send
+ voicemail from VoiceMailMain Reported by: William F Acker (via
+ the -users mailing list) Patch by: Corydon76 (license 14)
+
+2008-02-08 18:48 +0000 [r103070-103120] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Prevent a potential three-thread deadlock. Also
+ added a comment block to explicitly state the locking order
+ necessary inside app_queue. (closes issue #11862) Reported by:
+ flujan Patches: 11862.patch uploaded by putnopvut (license 60)
+ Tested by: flujan
+
+ * channels/chan_iax2.c: Yield the thread and return -1 if the ioctl
+ fails for Zaptel timing device. (closes issue #11891) Reported
+ by: tzafrir
+
+2008-02-08 15:08 +0000 [r102968] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: Make sure the presence of dbsecret is
+ factored into user scoring. (closes issue #11952) Reported by:
+ bbhoss
+
+2008-02-07 19:53 +0000 [r102858] Jason Parker <jparker@digium.com>
+
+ * res/res_features.c: Specify which digit string was matched in
+ debug message. (closes issue #11949) Reported by: dimas Patches:
+ v1-feature-debug.patch uploaded by dimas (license 88)
+
+2008-02-07 16:41 +0000 [r102807] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configs/zapata.conf.sample: document usage of 'transfer'
+ configuration option for ISDN PRI switch-side transfers
+
+2008-02-06 17:59 +0000 [r102653-102725] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Only consider a T.38-only INVITE compatible
+ if we have both a joint capability between us and them and if
+ they provided T.38.
+
+ * main/global_datastores.c: Add missing header file and
+ ASTERISK_FILE_VERSION usage. (closes issue #11936) Reported by:
+ snuffy
+
+2008-02-06 15:19 +0000 [r102651] Russell Bryant <russell@digium.com>
+
+ * configs/features.conf.sample: Clarify setting DYNAMIC_FEATURES so
+ that it gets inherited by outbound channels. (due to a discussion
+ between me and a user via email)
+
+2008-02-06 11:48 +0000 [r102627] Kevin P. Fleming <kpfleming@digium.com>
+
+ * pbx/Makefile, res/Makefile: ensure that all remaining
+ multi-object modules are built using their proper CFLAGS and
+ include directory paths
+
+2008-02-06 00:26 +0000 [r102576] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Move around some defines to unbreak ODBC
+ storage. (closes issue #11932) Reported by: snuffy
+
+2008-02-05 20:02 +0000 [r102453] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_mgcp.c: Clear the DTMF buffer on hangup. (closes
+ issue #11919) Reported by: eferro Patches:
+ mgcp_dtmfclean_on_hangup.diff uploaded by eferro (license 337)
+ Tested by: eferro
+
+2008-02-05 19:52 +0000 [r102450] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: If a REGISTER attempt comes in that is a
+ retransmission of a previous REGISTER do not create a new nonce
+ value. (issue #BE-381)
+
+2008-02-05 17:15 +0000 [r102425] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/Makefile: ensure that components of chan_misdn.so are
+ built using any special build options that the configure script
+ generated (reported by Philipp Kempgen on asterisk-dev)
+
+2008-02-05 15:09 +0000 [r102378] Joshua Colp <jcolp@digium.com>
+
+ * res/res_clioriginate.c: Perform dialing asynchronously when using
+ the originate CLI command so the CLI does not appear to block.
+ (closes issue #11927) Reported by: bbhoss
+
+2008-02-04 21:06 +0000 [r102214-102323] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, utils/muted.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Cross-platform
+ fix: OS X now deprecates the use of the daemon(3) API. (closes
+ issue #11908) Reported by: oej Patches:
+ 20080204__bug11908.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Corydon76
+
+ * funcs/func_strings.c: Missing braces. (closes issue #11912)
+ Reported by: dimas Patches: sprintf.patch uploaded by dimas
+ (license 88)
+
+2008-02-03 16:38 +0000 [r102090-102142] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Use the same CSEQ on CANCEL as on INVITE
+ (according to RFC 3261) (closes issue #9492) Reported by:
+ kryptolus Patches: bug9492.txt uploaded by oej (license 306)
+ Tested by: oej
+
+ * channels/chan_sip.c: Handle ACK and CANCEL in an invite
+ transaction - even if we get INFO transactions during the actual
+ call setup. (closes issue #10567) Reported by: jacksch Tested by:
+ oej Patch by: oej inspired by suggestions from neutrino88 in the
+ bug tracker
+
+2008-02-01 23:06 +0000 [r101989] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Change the SDP_SAMPLE_RATE macro. It turns
+ out that even though G.722 is 16 kHz, it is supposed to specified
+ as 8 kHz in the RTP, and RTP timestamps are supposed to be
+ calculated based on 8 kHz. (Apparently this is due to a bug in a
+ spec, but people follow it anyway, because it's the spec ...)
+
+2008-02-01 21:54 +0000 [r101894-101942] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Fix the VM_DUR variable for forwarded
+ voicemail, and fixed several other bugs while I'm in the area.
+ (closes issue #11615) Reported by: jamessan Patches:
+ 20071226__bug11615__2.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Corydon76, jamessan
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ acinclude.m4: Change detection of getifaddrs to use
+ AST_C_COMPILE_CHECK, backported from trunk (as suggested by
+ kpfleming)
+
+2008-02-01 17:41 +0000 [r101822] Jason Parker <jparker@digium.com>
+
+ * apps/app_authenticate.c: Remove a needless (and incorrect) call
+ to feof() after fgets(). This would have exited the loop early if
+ you had an authentication file with no newline at the end.
+
+2008-02-01 17:27 +0000 [r101818-101820] Russell Bryant <russell@digium.com>
+
+ * apps/app_authenticate.c: off by one error
+
+ * apps/app_authenticate.c: Don't overwrite the last character of a
+ line if it's not a newline. This would happen if the last line in
+ the file doesn't have a newline. (pointed out by Qwell)
+
+2008-02-01 15:55 +0000 [r101772] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ main/acl.c: Compatibility fix for OpenWRT (reported by Brian
+ Capouch via the mailing list)
+
+2008-02-01 00:32 +0000 [r101693] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Add some more sanity checking on IAX2 dial
+ strings for the case that no peer or hostname was provided, which
+ is the one part of the dial string that is absolutely required.
+ If it's not there, bail out. (closes issue #11897) Reported by
+ sokhapkin Patch by me
+
+2008-02-01 00:06 +0000 [r101649] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_amd.c: From bugtracker: "fix totalAnalysisTime to handle
+ periods of no channel activity" (closes issue #9256) Reported by:
+ cmaj Patches: amd-dont-wait-too-long-for-frames-take3.diff.txt
+ uploaded by cmaj (license 111) Tested by: cmaj, skygreg, ZX81,
+ rjain
+
+2008-01-31 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.18 released.
+
+2008-01-31 23:10 +0000 [r101601] Russell Bryant <russell@digium.com>
+
+ * main/translate.c, main/file.c: Fix a couple of places where
+ ast_frfree() was not called on a frame that came from a
+ translator. This showed itself by g729 decoders not getting
+ released. Since the flag inside the translator frame never got
+ unset by freeing the frame to indicate it was no longer in use,
+ the translators never got destroyed, and thus the g729 licenses
+ were not released. (closes issue #11892) Reported by: xrg
+ Patches: 11892.diff uploaded by russell (license 2) Tested by:
+ xrg, russell
+
+2008-01-31 21:00 +0000 [r101531] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_monitor.c: 1. Prevent the addition of an extra '/' to the
+ beginning of an absolute pathname. 2. If ast_monitor_change_fname
+ is called and the new filename is the same as the old, then exit
+ early and don't set the filename_changed field in the monitor
+ structure. Setting it in this case was causing ast_monitor_stop
+ to erroneously delete them. (closes issue #11741) Reported by:
+ garlew Tested by: putnopvut
+
+2008-01-31 19:52 +0000 [r101482] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c, channels/chan_iax2.c: Solaris compat fixes
+ for struct in_addr funkiness. Issue #11885, patch by snuffy.
+
+2008-01-31 19:30 +0000 [r101480] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c: closes issue #11845; that's the one where there's a
+ 1004 byte cdr leak with every AMI Redirect to a zap channel
+
+2008-01-31 19:17 +0000 [r101413-101433] Russell Bryant <russell@digium.com>
+
+ * channels/chan_agent.c: Add more missing locking of the agents
+ list ...
+
+ * channels/chan_agent.c: Move the locking from find_agent() into
+ the agent dialplan function handler to ensure that the agent
+ doesn't disappear while we're looking at it.
+
+ * channels/chan_agent.c: Add missing locking to the find_agent()
+ function.
+
+2008-01-30 15:41 +0000 [r101222] Joshua Colp <jcolp@digium.com>
+
+ * main/slinfactory.c: Fix an issue where if a frame of higher
+ sample size preceeded a frame of lower sample size and
+ ast_slinfactory_read was called with a sample size of the
+ combined values or higher a crash would happen. (closes issue
+ #11878) Reported by: stuarth
+
+2008-01-30 15:34 +0000 [r101219] Jason Parker <jparker@digium.com>
+
+ * configs/extensions.conf.sample: Change default config to use
+ descending channel order of groups, rather than ascending. Fixes
+ a potential source of confusion in glare-type situations. Issue
+ 11875, reported by JimVanM.
+
+2008-01-30 15:23 +0000 [r101216] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix a logic error with regards to autofill.
+ Prior to this change, it was possible for a caller to go out of
+ turn if autofill were enabled and callers ahead in the queue were
+ attempting to call a member. This change fixes this.
+
+2008-01-30 11:20 +0000 [r101152] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Stop musiconhold on attended transfer.
+ (closes issue #11872) Reported by: gareth Patches:
+ svn-101018.patch uploaded by gareth (license 208)
+
+2008-01-29 23:50 +0000 [r101080] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * build_tools/make_version: updated build_tools to handle the
+ autotag directory structure changes; changes related to BE-353.
+ Patch by The Russell and reviewed by The Me.
+
+2008-01-29 23:02 +0000 [r100973-101035] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Remove a memory leak from updating realtime
+ queues
+
+ * apps/app_queue.c: Fixing an erroneous return value returned when
+ attempting to pause or unpause a queue member fails. Fixes
+ BE-366, thanks to John Bigelow for writing the patch.
+
+2008-01-29 17:57 +0000 [r100934] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_mixmonitor.c: Don't forget to record the channel so we
+ know whether it is bridged or not later. (closes issue #11811)
+ Reported by: slavon
+
+2008-01-29 17:43 +0000 [r100932] Russell Bryant <russell@digium.com>
+
+ * main/Makefile: Fix the last couple of issues related to building
+ from a path that contains spaces. (closes issue #11834)
+
+2008-01-29 17:41 +0000 [r100930] Jason Parker <jparker@digium.com>
+
+ * channels/misdn_config.c: Initialize an array to 0s if config
+ option not specified. (closes issue #11860) Patches:
+ misdn_get_config.v1.diff uploaded by IgorG (license 20)
+
+2008-01-29 17:21 +0000 [r100882-100922] Russell Bryant <russell@digium.com>
+
+ * Makefile: Use GNU make magic instead of shell magic to escape
+ spaces in the working directory. (related to issue #11834)
+
+ * Makefile: Fix building Asterisk when the working path has spaces
+ in it. (closes issue #11834) Reported by: spendergrass Patched
+ by: me
+
+2008-01-29 16:10 +0000 [r100835] Jason Parker <jparker@digium.com>
+
+ * channels/chan_zap.c: Allow zap groups above 30 to work properly.
+ (closes issue #11590) Reported by: tbsky
+
+2008-01-29 10:36 +0000 [r100793] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: fixed potential segfault in misdn show
+ channels CLI command
+
+2008-01-29 08:26 +0000 [r100740] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: (closes issue #11736) Reported by: MVF
+ Patches: bug11736-2.diff uploaded by oej (license 306) Tested by:
+ oej, MVF, revolution (russellb: This was the showstopper for the
+ release.)
+
+2008-01-28 21:02 +0000 [r100675] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: WaitExten didn't handle AbsoluteTimeout properly
+ (went to 't' instead of 'T')
+
+2008-01-28 20:55 +0000 [r100673] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_vpb.cc, UPGRADE.txt: Undoing the deprecation of
+ chan_vpb. It is alive and well.
+
+2008-01-28 20:42 +0000 [r100672] Jason Parker <jparker@digium.com>
+
+ * apps/app_voicemail.c: When using ODBC_STORAGE, make sure we put
+ greeting files into the database like we do with the others.
+ Issue #11795 Reported by: dimas Patches: vmgreet.patch uploaded
+ by dimas (license 88)
+
+2008-01-28 18:34 +0000 [r100626-100629] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: For some reason, the use of this strdupa()
+ is leading to memory corruption on freebsd sparc64. This trivial
+ workaround fixes it. (closes issue #10300, closes issue #11857,
+ reported by mattias04 and Home-of-the-Brave)
+
+ * res/res_features.c: Fix a crash in ast_masq_park_call() (issue
+ #11342) Reported by: DEA Patches: res_features-park.txt uploaded
+ by DEA (license 3)
+
+2008-01-28 18:23 +0000 [r100624] Jason Parker <jparker@digium.com>
+
+ * channels/chan_zap.c: Correct a comment which made little/no
+ sense.
+
+2008-01-28 17:15 +0000 [r100581] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, channels/chan_local.c,
+ include/asterisk/channel.h: Make some deadlock related fixes.
+ These bugs were discovered and reported internally at Digium by
+ Steve Pitts. - Fix up chan_local to ensure that the channel lock
+ is held before the local pvt lock. - Don't hold the channel lock
+ when executing the timing function, as it can cause a deadlock
+ when using chan_local. This actually changes the code back to be
+ how it was before the change for issue #10765. But, I added some
+ other locking that I think will prevent the problem reported
+ there, as well.
+
+2008-01-27 21:59 +0000 [r100465] Tilghman Lesher <tlesher@digium.com>
+
+ * main/rtp.c, channels/chan_mgcp.c, main/cdr.c,
+ channels/chan_misdn.c, main/dnsmgr.c, channels/chan_sip.c,
+ channels/chan_h323.c, include/asterisk/sched.h, main/file.c,
+ pbx/pbx_dundi.c, channels/chan_iax2.c: When deleting a task from
+ the scheduler, ignoring the return value could possibly cause
+ memory to be accessed after it is freed, which causes all sorts
+ of random memory corruption. Instead, if a deletion fails, wait a
+ bit and try again (noting that another thread could change our
+ taskid value). (closes issue #11386) Reported by: flujan Patches:
+ 20080124__bug11386.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Corydon76, flujan, stuarth`
+
+2008-01-25 22:32 +0000 [r100418] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_vpb.cc, UPGRADE.txt: Deprecating chan_vpb. It is
+ now preferred that users of Voicetronix products use chan_zap in
+ combination with their zaptel drivers.
+
+2008-01-25 21:24 +0000 [r100378] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c: This would have never been true, since we're
+ passing (sizeof(req.data) - 1) as the len to recvfrom().
+
+2008-01-24 21:57 +0000 [r100264] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/app.h: make these macros not assume that the
+ only other field in the structure is 'argc'... this is true when
+ someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable
+ to define your own structure as long as it has the right fields
+
+2008-01-24 17:22 +0000 [r100164] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c: Update main Asterisk copyright info to 2008
+
+2008-01-24 16:41 +0000 [r100138] Jason Parker <jparker@digium.com>
+
+ * main/acl.c: Fix compilation on Solaris. (closes issue #11832)
+ Patches: bug-11832.diff uploaded by snuffy (license 35)
+
+2008-01-23 21:07 +0000 [r99977-99978] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Second attempt. Don't change invitestate
+ when receiving 18x messages in CANCEL state. (issue #11736)
+ Reported by: MVF Patch by oej.
+
+ * channels/chan_sip.c: Make sure we don't cancel destruction on
+ calls in CANCEL state, even if we get 183 while waiting for
+ answer on our CANCEL. (issue #11736) Reported by: MVF Patches:
+ bug11736.txt uploaded by oej (license 306) Tested by: MVF
+
+2008-01-23 20:25 +0000 [r99975] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_externalivr.c: Fixing a typo.
+
+2008-01-23 17:46 +0000 [r99923] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c: ChanSpy issues a beep when it starts at the
+ beginning of a list of channels to potentially spy on. However,
+ if there were no matching channels, it would beep at you over and
+ over, which is pretty annoying. Now, it will only beep once in
+ the case that there are no channels to spy on, but it will still
+ beep again once it reaches the beginning of the channel list
+ again. (closes issue #11738, patched by me)
+
+2008-01-23 16:18 +0000 [r99878] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: These flag tests were illogical. They were
+ testing sip_peer flags on a sip_pvt. Thanks to Russell for
+ helping to get this odd problem figured out.
+
+2008-01-23 04:31 +0000 [r99718-99777] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: When we reset the password via an external
+ command, we should also reset the password stored in the
+ in-memory list, too (otherwise it doesn't really take effect).
+ (closes issue #11809) Reported by: davetroy Patches:
+ fix_externpass.diff uploaded by davetroy (license 384)
+
+ * res/res_odbc.c: Oops, should have checked for a NULL obj, here,
+ too
+
+ * main/acl.c: Just confirmed that all current platforms need this
+ header file
+
+2008-01-22 20:56 +0000 [r99652] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Thanks to Russell's education I realize that
+ BUFSIZ has changed since I learned the C language over 20 years
+ ago... Resetting chan_sip to the size of BUFSIZ that I expected
+ in my old head to avoid to heavy memory allocations on some
+ systems.
+
+2008-01-22 20:34 +0000 [r99643] Tilghman Lesher <tlesher@digium.com>
+
+ * main/acl.c: Fix the defines for OS X (and Solaris, too)
+
+2008-01-22 17:41 +0000 [r99592-99594] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_local.c, res/res_features.c, channels/chan_agent.c,
+ apps/app_followme.c: Add more dependencies on chan_local and add
+ a note to the description of chan_local so that people don't
+ disable it in menuselect just to clean up.
+
+ * apps/app_dial.c: Add dependency on chan_local to app_dial. Dial
+ still runs without chan_local, but will be missing forwarding
+ functionality.
+
+2008-01-22 16:54 +0000 [r99540] Tilghman Lesher <tlesher@digium.com>
+
+ * main/acl.c: Ensure that we can get an address even when we don't
+ have a default route. (closes issue #9225) Reported by: junky
+ Patches: 20080122__bug9225.diff.txt uploaded by Corydon76
+ (license 14) Tested by: oej, loloski, sergee
+
+2008-01-22 15:08 +0000 [r99501] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Cleaning up some documentation that led to
+ confusion in a bug report
+
+2008-01-21 23:55 +0000 [r99426] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_local.c: Fixing an issue wherein monitoring local
+ channels was not possible. During a channel masquerade, the
+ monitors on the two channels involved are swapped. In 99% of the
+ cases this results in the desired effect. However, if monitoring
+ a local channel, this caused the monitor which was on the local
+ channel to get moved onto a channel which is immediately hung up
+ after the masquerade has completed. By swapping the monitors
+ prior to the masquerade, we avoid the problem by tricking the
+ masquerade into placing the monitor back onto the channel where
+ we want it. During the investigation of the issue, the channel's
+ monitor was the only thing that was swapped in such a manner
+ which did not make sense to have done. All other variable
+ swapping made sense.
+
+2008-01-21 18:11 +0000 [r99341] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_odbc.c, configs/res_odbc.conf.sample,
+ include/asterisk/res_odbc.h: Permit the user to specify number of
+ seconds that a connection may remain idle, which fixes a crash on
+ reconnect with the MyODBC driver. (closes issue #11798) Reported
+ by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: mvanbaak
+
+2008-01-21 16:01 +0000 [r99301] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Bump the buffer size for Via headers up to
+ 512. There are some exceptionally large Via headers out there.
+ (closes issue #11783) Reported by: ofirroval
+
+2008-01-19 10:05 +0000 [r99187] Russell Bryant <russell@digium.com>
+
+ * main/slinfactory.c: Fix a couple of memory leaks with frame
+ handling. Specifically, ast_frame_free() needed to be called on
+ the frame that came from the translator to signed linear.
+
+2008-01-18 22:57 +0000 [r99127] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/channel.h: Remove the __ in front of the unused
+ variable. This causes some compilers to freak out.
+
+2008-01-18 21:37 +0000 [r99079-99081] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/translate.h, main/frame.c: Revert adding the
+ packed attribute, as it really doesn't make sense why that would
+ do any good. Fix the real bug, which is to do the check to see if
+ the frame came from a translator at the beginning of
+ ast_frame_free(), instead of at the end. This ensures that it
+ always gets checked, even if none of the parts of the frame are
+ malloc'd, and also ensures that we aren't looking at free'd
+ memory in the case that it is a malloc'd frame. (closes issue
+ #11792, reported by explidous, patched by me)
+
+ * include/asterisk/translate.h: Since we're relying on the offset
+ between the frame and the beginning of the translator pvt struct,
+ set the packed attribute to make sure we get to the right place.
+ (potential fix for issue #11792)
+
+2008-01-18 17:13 +0000 [r99032] Terry Wilson <twilson@digium.com>
+
+ * res/res_features.c: This should at least temporarily fix a
+ problem where the 't' Dial option is incorrectly passed to the
+ transferee when built-in attended transfers are used. There is
+ still a problem with 'T', but better to fix some problems than no
+ problems while we work on it. (closes issue #7904) Reported by:
+ k-egg Patches: transfer-fix-b14-r97657.diff uploaded by sergee
+ (license 138) Tested by: sergee, otherwiseguy
+
+2008-01-17 23:42 +0000 [r99007-99014] Pari Nannapaneni <paripurnachand@digium.com>
+
+ * configs/cdr.conf.sample: doh! revert a revert of a revert
+ (changed by mistake in 99010)
+
+ * main/manager.c, configs/cdr.conf.sample: missed that one while
+ reverting
+
+ * main/manager.c: reverting 99001 - We need the Max-Age for
+ extending the life of cookie mansession_id
+
+2008-01-17 22:37 +0000 [r99004] Russell Bryant <russell@digium.com>
+
+ * main/frame.c, channels/chan_iax2.c, include/asterisk/frame.h:
+ Have IAX2 optimize the codec translation path just like chan_sip
+ does it. If the caller's codec is in our codec list, move it to
+ the top to avoid transcoding. (closes issue #10500) Reported by:
+ stevedavies Patches: iax-prefer-current-codec.patch uploaded by
+ stevedavies (license 184) iax-prefer-current-codec.1.4.patch
+ uploaded by stevedavies (license 184) Tested by: stevedavies, pj,
+ sheldonh
+
+2008-01-17 21:31 +0000 [r99001] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/manager.c: we should only send the Set-Cookie header to the
+ browser on the first response after creating a manager session,
+ not on every response (doing so causes the browser to clear any
+ local cookies it may have associated with the session)
+
+2008-01-17 16:19 +0000 [r98991] Jason Parker <jparker@digium.com>
+
+ * configs/zapata.conf.sample: Add a clarification about the
+ immediate= option of zapata.conf Issue 11784, patch by klaus3000.
+
+2008-01-16 22:36 +0000 [r98982] Russell Bryant <russell@digium.com>
+
+ * .cleancount, include/asterisk/channel.h: Add an unused pointer to
+ the ast_channel struct. This makes the ast_channel structure
+ retain the same size as it had in previous 1.4 releases. Also,
+ all of the offsets for members in the structure are still the
+ same (except for the two pointers that got replaced for the new
+ spy/whisper architecture.)
+
+2008-01-16 20:34 +0000 [r98966-98973] Joshua Colp <jcolp@digium.com>
+
+ * .cleancount: Bump up cleancount due to previous commit that
+ changed the channel structure.
+
+ * apps/app_chanspy.c, apps/app_mixmonitor.c, main/rtp.c,
+ main/channel.c, apps/app_meetme.c, include/asterisk/audiohook.h
+ (added), main/Makefile, include/asterisk/chanspy.h (removed),
+ include/asterisk/channel.h, main/audiohook.c (added): Replace
+ current spy architecture with backport of audiohooks. This should
+ take care of current known spy issues.
+
+ * channels/chan_iax2.c: Add missing NULLs at end of two
+ ast_load_realtimes. (closes issue #11769) Reported by: tequ
+ Patches: chaniax.patch uploaded by dimas (license 88)
+
+2008-01-16 17:20 +0000 [r98964] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_local.c: Fix a deadlock in chan_local in
+ local_hangup. There was contention because the local_pvt was held
+ and it was attempting to lock a channel, which is the incorrect
+ locking order. (closes issue #11730) Reported by: UDI-Doug
+ Patches: 11730.patch uploaded by putnopvut (license 60) Tested
+ by: UDI-Doug
+
+2008-01-16 15:08 +0000 [r98951-98960] Joshua Colp <jcolp@digium.com>
+
+ * main/dial.c: Introduce a lock into the dialing API that protects
+ it when destroying the structure. (closes issue #11687) Reported
+ by: callguy Patches: 11687.diff uploaded by file (license 11)
+
+ * main/rtp.c: Add two more SDP names for ulaw and alaw. (closes
+ issue #11777) Reported by: tootai
+
+ * channels/chan_sip.c: Don't drop the old record route information
+ when dealing with packets related to a reinvite. (closes issue
+ #11545) Reported by: kebl0155 Patches: reinvite-patch.txt
+ uploaded by kebl0155 (license 356)
+
+ * build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
+ configure.ac, makeopts.in: Add autoconf logic for speexdsp. Later
+ versions use a separate library for some things so we need to use
+ it if present in codec_speex. (closes issue #11693) Reported by:
+ yzg
+
+2008-01-15 23:50 +0000 [r98943-98946] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Change a buffer in check_auth() to be a
+ thread local dynamically allocated buffer, instead of a massive
+ buffer on the stack. This fixes a crash reported by Qwell due to
+ running out of stack space when building with LOW_MEMORY defined.
+ On a very related note, the usage of BUFSIZ in various places in
+ chan_sip is arbitrary and careless. BUFSIZ is a system specific
+ define. On my machine, it is 8192, but by definition (according
+ to google) could be as small as 256. So, this buffer in
+ check_auth was 16 kB. We don't even support SIP messages larger
+ than 4 kB! Further usage of this define should be avoided, unless
+ it is used in the proper context.
+
+ * main/rtp.c, include/asterisk/translate.h, main/frame.c,
+ main/translate.c, main/abstract_jb.c, channels/chan_iax2.c,
+ codecs/codec_zap.c, include/asterisk/frame.h: Commit a fix for
+ some memory access errors pointed out by the valgrind2.txt output
+ on issue #11698. The issue here is that it is possible for an
+ instance of a translator to get destroyed while the frame
+ allocated as a part of the translator is still being processed.
+ Specifically, this is possible anywhere between a call to
+ ast_read() and ast_frame_free(), which is _a lot_ of places in
+ the code. The reason this happens is that the channel might get
+ masqueraded during this time. During a masquerade, existing
+ translation paths get destroyed. So, this patch fixes the issue
+ in an API and ABI compatible way. (This one is for you,
+ paravoid!) It changes an int in ast_frame to be used as flag
+ bits. The 1 bit is still used to indicate that the frame contains
+ timing information. Also, a second flag has been added to
+ indicate that the frame came from a translator. When a frame with
+ this flag gets released and has this flag, a function is called
+ in translate.c to let it know that this frame is doing being
+ processed. At this point, the flag gets cleared. Also, if the
+ translator was requested to be destroyed while its internal frame
+ still had this flag set, its destruction has been deffered until
+ it finds out that the frame is no longer being processed.
+ Admittedly, this feels like a hack. But, it does fix the issue,
+ and I was not able to think of a better solution ...
+
+2008-01-15 20:08 +0000 [r98894-98934] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Based on the boundary found move over the
+ correct amount. (closes issue #11750) Reported by: tasker
+
+ * channels/chan_sip.c: Accept "; boundary=" not just ";boundary="
+ in the multipart mixed content type. (closes issue #11750)
+ Reported by: tasker
+
+2008-01-14 20:59 +0000 [r98849] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Adding in appropriate unlocks for the locks
+ I added. Thanks to joetester on IRC for pointing this out.
+
+2008-01-14 17:38 +0000 [r98774] Russell Bryant <russell@digium.com>
+
+ * main/translate.c: Revert a change that introduces an unacceptable
+ performance hit and is causing memory leaks ... (from rev 97973)
+
+2008-01-14 16:35 +0000 [r98733-98737] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fixing another compilation error. I'm a bit off
+ today :(
+
+ * apps/app_queue.c: Oops. Last commit had compilation error.
+
+ * apps/app_queue.c: Adding explicit defaults for missing options to
+ init_queue. This is necessary because if a user either removes or
+ comments one of these options and reloads their queues, the
+ option will not reset to its default, instead maintaining the
+ value from prior to the reload. Thanks to John Bigelow for
+ pointing this error out to me.
+
+2008-01-12 00:05 +0000 [r98467] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_odbc.c: Add a connection timeout attribute, as that was
+ what was intended with the login timeout, but ODBC divides it up
+ into 2 different timeouts. (Closes issue #11745)
+
+2008-01-11 22:46 +0000 [r98390] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_dundi.c: Fix up setting the EID on BSD based systems.
+ (closes issue #11646) Reported by: caio1982 Patches:
+ dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22)
+ dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22) Tested
+ by: caio1982, mvanbaak
+
+2008-01-11 21:28 +0000 [r98372] Pari Nannapaneni <paripurnachand@digium.com>
+
+ * main/http.c: Comment explaining how to force browser to always
+ read some html files from server.
+
+2008-01-11 19:51 +0000 [r98317-98325] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: If the incoming RTP stream changes codec force the
+ bridge to break if the other side does not support it. (closes
+ issue #11729) Reported by: tsearle Patches:
+ new_codec_patch_udiff.patch uploaded by tsearle (license 373)
+
+ * res/res_agi.c: If the channel is hungup during RECORD FILE send a
+ result code of -1 to be uniform with everything else. (closes
+ issue #11743) Reported by: davevg Patches: res_agi.diff uploaded
+ by davevg (license 209)
+
+2008-01-11 19:10 +0000 [r98315] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c: Properly report the hangup cause as no answer
+ when someone does not answer (closes issue #10574, reported by
+ boch, patched by moy)
+
+2008-01-11 18:25 +0000 [r98266] Tilghman Lesher <tlesher@digium.com>
+
+ * codecs/gsm/Makefile: Add another exception (which doesn't work)
+ for -march optimization flag. Reported by: thomasmebes Patch by:
+ tilghman (Closes issue #11563)
+
+2008-01-11 18:25 +0000 [r98265] Russell Bryant <russell@digium.com>
+
+ * doc/security.txt, main/asterisk.c, configure,
+ include/asterisk/autoconfig.h.in, main/Makefile, configure.ac,
+ makeopts.in: Backport the ability to set the ToS bits on Linux
+ when not running as root. Normally, we would not backport
+ features into 1.4, but, I was convinced by the justification
+ supplied by the supplier of this patch. He pointed out that this
+ patch removes a requirement for running as root, thus reducing
+ the potential impacts of security issues. (closes issue #11742)
+ Reported by: paravoid Patches: libcap.diff uploaded by paravoid
+ (license 200)
+
+2008-01-11 17:22 +0000 [r98219] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_followme.c: Ensure the return value of ast_bridge_call
+ is passed back up as the application return value. This is needed
+ for transfers to function so the PBX core knows to continue
+ execution. (closes issue #10327) Reported by: kkiely
+
+2008-01-11 15:52 +0000 [r98164] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Back out changes from revision 97077, since
+ it wasn't perfect
+
+2008-01-11 03:39 +0000 [r97976-98082] Russell Bryant <russell@digium.com>
+
+ * main/frame.c: Fix samples vs. length calculations for g722
+
+ * main/translate.c: Simplify this code with a suggestion from Luigi
+ on the asterisk-dev list. Instead of using is16kHz(), implement a
+ format_rate() function.
+
+ * main/translate.c: Fix various timing calculations that made
+ assumptions that the audio being processed was at a sample rate
+ of 8 kHz.
+
+2008-01-10 23:08 +0000 [r97973] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c, main/translate.c: 1) When we get a
+ translated frame out, clone it, because if the translator pvt is
+ freed before we use the frame, bad things happen. 2) Getting a
+ failure from ast_sched_delete means that the schedule ID is
+ currently running. Don't just ignore it. (Closes issue #11698)
+
+2008-01-10 21:57 +0000 [r97925] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Let us leave a voicemail for ourself if we
+ have logged into VoiceMailMain and chosen to leave a message.
+ (closes issue #11735, reported and patched by jamessan)
+
+2008-01-10 21:37 +0000 [r97849-97889] Steve Murphy <murf@digium.com>
+
+ * pbx/ael/ael_lex.c, pbx/Makefile, pbx/ael/ael.flex: Applied the
+ same fixes for ael.flex as was done in 97849 for ast_expr2.fl;
+ overrode the normally generate yyfree func with our own version
+ that checks the pointer for non-null before passing to free().
+ Also takes care of a little problem with 2.5.33 and the use of
+ the __STDC_VERSION__ macro.
+
+ * main/ast_expr2.fl, main/Makefile, main/ast_expr2f.c: This is a
+ fix for 2 things: a problem Terry was having in OSX with null
+ pointers, which was my fault, as I probably forgot to run the sed
+ script last time I made mods. So, I moved the fix into the flex
+ input itself. Then, I found when I used flex 2.5.33, that it was
+ using __STDC_VERSION__, and that's not real good; so I added back
+ in a DIFFERENT sed script to fix that little mess. Tested
+ everything, a couple different ways. Hope I did no harm, at the
+ least.
+
+2008-01-10 20:12 +0000 [r97847] Jason Parker <jparker@digium.com>
+
+ * include/asterisk/frame.h: Fix a comment that is no longer true.
+
+2008-01-10 16:19 +0000 [r97734-97753] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_kdeconsole.h (removed), configs/modules.conf.sample,
+ pbx/kdeconsole_main.cc (removed): Remove other remnants of
+ pbx_kdeconsole
+
+ * pbx/pbx_kdeconsole.cc (removed), build_tools/menuselect-deps.in,
+ configure, include/asterisk/autoconfig.h.in, configure.ac,
+ makeopts.in: Remove pbx_kdeconsole from the tree. It hasn't
+ worked in ages, and nobody has complained. (closes issue #11706,
+ reported by caio1982)
+
+2008-01-10 15:07 +0000 [r97697] Joshua Colp <jcolp@digium.com>
+
+ * funcs/func_groupcount.c: Don't try to copy the category from the
+ group if no category exists. (closes issue #11724) Reported by:
+ IgorG Patches: group_count.v1.patch uploaded by IgorG (license
+ 20)
+
+2008-01-09 23:01 +0000 [r97640-97645] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_gtkconsole.c: Strip terminal sequences from the verbose
+ messages
+
+ * pbx/pbx_gtkconsole.c: Make pbx_gtkconsole build ... but doesn't
+ actually load on my system still (related to issue #11706)
+
+2008-01-09 20:28 +0000 [r97618-97622] Jason Parker <jparker@digium.com>
+
+ * main/cli.c: Correctly display a message if a command could not be
+ found. Also fix a comment which may have led to this happening.
+ Issue 11718, reported by kshumard.
+
+ * main/cli.c: Fix some locking and return value funkiness. We
+ really shouldn't be unlocking this lock inside of a function,
+ unless we locked it there too.
+
+2008-01-09 18:48 +0000 [r97575] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Part 2 of app_queue doxygen improvements. Some
+ smaller functions this time
+
+2008-01-09 18:02 +0000 [r97529] Russell Bryant <russell@digium.com>
+
+ * res/res_features.c: Fix saying the parking space number to the
+ caller doing the parking ...
+
+2008-01-09 17:21 +0000 [r97491] Kevin P. Fleming <kpfleming@digium.com>
+
+ * codecs/codec_zap.c: report the same message whether Zaptel does
+ not have transcoder support loaded or no transcoders were found
+
+2008-01-09 16:44 +0000 [r97489] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * channels/chan_gtalk.c: Set the caller id within the gtalk_alloc
+ function. As underlined in issue #10437 by Josh, we need to
+ prevent a possible memory leak. We only set the name part of the
+ caller id, the number part is not relevant when dealing with
+ JIDs. Closes issue #11549.
+
+2008-01-09 16:11 +0000 [r97450] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_meetme.c: Don't do conferencing totally in Zaptel if
+ Monitor is running on the channel. (closes issue #11709) Reported
+ by: BigJimmy Patches: patch-meetmerec uploaded by BigJimmy
+ (license 371)
+
+2008-01-09 15:43 +0000 [r97410-97448] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c: pass the right variable to get an error
+ string... oops
+
+ * channels/chan_zap.c: add error number output to ioctl failure
+ messages to help with debugging
+
+2008-01-09 00:44 +0000 [r97350] Tilghman Lesher <tlesher@digium.com>
+
+ * main/cli.c, main/editline/readline.c: Allow filename completion
+ on zero-length modules, remove a memory leak, remove a file
+ descriptor leak, and make filename completion thread-safe.
+ Patched and tested by tilghman. (Closes issue #11681)
+
+2008-01-09 00:17 +0000 [r97206-97308] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: use the \retval doxygen command properly
+
+ * apps/app_queue.c: Part 1 of N of adding doxygen comments to
+ app_queue. I picked some of the most common functions used (which
+ also happen to be some the biggest/ugliest functions too) to
+ document first. I'm pretty new to doxygen so criticism is
+ welcome.
+
+ * apps/app_queue.c: Some coding guidelines-related cleanup
+
+2008-01-08 20:48 +0000 [r97195] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_mgcp.c: Fix various DTMF issues in chan_mgcp.
+ (closes issue #11443) Reported by: eferro Patches:
+ dtmf_control_hybrid-inband-mode.patch uploaded by eferro (license
+ 337)
+
+2008-01-08 20:47 +0000 [r97194] Tilghman Lesher <tlesher@digium.com>
+
+ * main/autoservice.c, main/utils.c: Increase constants to where
+ we're less likely to hit them while debugging. (Closes issue
+ #11694)
+
+2008-01-08 20:42 +0000 [r97192] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Making some changes designed to not allow
+ for a corrupted mailstream for a vm_state. 1. Add locking to the
+ vm_state retrieval functions so that no linked list corruption
+ occurs. 2. Make sure to always grab the persistent vm_state when
+ mailstream access is necessary. 3. Correct an incorrect return
+ value in the init_mailstream function. (closes issue #11304,
+ reported by dwhite)
+
+2008-01-08 19:53 +0000 [r97093-97152] Joshua Colp <jcolp@digium.com>
+
+ * funcs/func_groupcount.c: If no group has been provided to the
+ GROUP_COUNT dialplan function then use the first one specific to
+ the channel. (closes issue #11077) Reported by: m4him
+
+ * apps/app_queue.c: Make app_queue calls work with directed pickup.
+ (closes issue #11700) Reported by: jbauer
+
+2008-01-08 18:02 +0000 [r97077] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, channels/chan_sip.c: Apply multiple crash fixes,
+ found in issue #11386, but not completely closing that issue.
+
+2008-01-07 20:47 +0000 [r96884-96932] Russell Bryant <russell@digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 96931 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r96931 | russell | 2008-01-07 14:46:22 -0600 (Mon, 07 Jan 2008) |
+ 2 lines Change misery.digium.com to pbx.digium.com ........
+
+ * res/res_smdi.c: Don't crash if something happens when setting up
+ an SMDI interface and it gets destroyed before the SMDI port
+ handling thread gets created.
+
+2008-01-07 14:34 +0000 [r96797-96815] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * res/res_jabber.c: Indentation fix, makes the code easier to read
+
+ * res/res_jabber.c: Compute the base64 value over the
+ [authzid]\0authcid\0password string, thus excluding the trailing
+ NULL byte. This change has already been committed to trunk, see
+ #11644.
+
+2008-01-05 02:09 +0000 [r96644] Russell Bryant <russell@digium.com>
+
+ * main/devicestate.c: Don't pass an empty string as the device
+ name.
+
+20