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authorlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2010-05-13 17:35:54 +0000
committerlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2010-05-13 17:35:54 +0000
commit78ae8cc2c6dcab07645806d2ab34427c2bd5cf56 (patch)
tree809da0f7acdc55629bfdbc83299ea7df482959e3
parent45a811548b103d9e7b1f3a852416f9704e1092af (diff)
Importing files for 1.6.1.20-rc2 release.
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.1.20-rc2@262985 f38db490-d61c-443f-a65b-d21fe96a405b
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-rw-r--r--.version1
-rw-r--r--ChangeLog66231
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diff --git a/.lastclean b/.lastclean
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+36
diff --git a/.version b/.version
new file mode 100644
index 000000000..d5e4271bb
--- /dev/null
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+1.6.1.20-rc2
diff --git a/ChangeLog b/ChangeLog
new file mode 100644
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+++ b/ChangeLog
@@ -0,0 +1,66231 @@
+2010-05-13 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.1.20-rc2 Released
+
+ * This will be the last maintenance release in this branch. Please
+ move to the latest 1.6.2.x release for continued issue support.
+ See http://www.asterisk.org/asterisk-versions for more information.
+
+2010-05-12 20:02 +0000 [r262802] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * main/loader.c, main/cli.c, /: Merged revisions 262800 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r262800 | pabelanger | 2010-05-12 15:59:16 -0400 (Wed,
+ 12 May 2010) | 8 lines Notify CLI when modules is loaded /
+ unloaded (closes issue #17308) Reported by: pabelanger Patches:
+ cli.modules.patch uploaded by pabelanger (license 224) Tested by:
+ pabelanger, russell ........
+
+2010-05-12 18:07 +0000 [r262747] David Vossel <dvossel@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 262744 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r262744 | dvossel | 2010-05-12 13:01:20 -0500 (Wed, 12 May 2010)
+ | 17 lines Merged revisions 262662 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010)
+ | 11 lines fixes app_meetme dsp error We attempted to detect
+ silence after translating a frame from signed linear. This caused
+ a flooding of errors. To resolve this the code to detect silence
+ was moved before the translation. (closes issue #17133) Reported
+ by: jsdyer ........ ................
+
+2010-05-12 16:29 +0000 [r262515-262658] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_privacy.c: Merged revisions 262656 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r262656 |
+ tilghman | 2010-05-12 11:23:26 -0500 (Wed, 12 May 2010) | 8 lines
+ Ensure the arguments are initialized. Also miscellaneous CG
+ cleanup. (closes issue #16576) Reported by: uxbod Patches:
+ 20100505__issue16576.diff.txt uploaded by tilghman (license 14)
+ Tested by: uxbod ........
+
+ * /, include/asterisk/causes.h: Merged revisions 262513 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r262513 | tilghman | 2010-05-11 16:25:05 -0500 (Tue, 11
+ May 2010) | 7 lines Move cause 200 to cause 26, as specified in
+ Q.850. Also cleanup the formatting and add a few more that seem
+ like good candidates. (closes issue #16157) Reported by: wimpy
+ ........
+
+2010-05-11 20:02 +0000 [r262426] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 262419 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r262419 |
+ pabelanger | 2010-05-11 15:40:37 -0400 (Tue, 11 May 2010) | 8
+ lines Improve logging by displaying line number (closes issue
+ #16303) Reported by: dant Patches: issue16303.patch.v2 uploaded
+ by pabelanger (license 224) Tested by: dant, lmadsen, pabelanger
+ ........
+
+2010-05-11 19:58 +0000 [r262424] Jason Parker <jparker@digium.com>
+
+ * /, res/Makefile: Merged revisions 262422 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r262422 | qwell | 2010-05-11 14:57:24 -0500 (Tue, 11 May 2010) |
+ 18 lines Merged revisions 262421 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r262421 | qwell | 2010-05-11 14:55:42 -0500 (Tue, 11 May 2010) |
+ 11 lines Use a less silly method for modifying a flex-generated
+ file. The sed syntax that was used wasn't actually valid, causing
+ some versions to choke. This is the method that is used in 1.6.x+
+ for similar changes. (closes issue #16696) Reported by: bklang
+ Patches: 16696-sedfix.diff uploaded by qwell (license 4) Tested
+ by: qwell ........ ................
+
+2010-05-11 19:30 +0000 [r262416] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, channels/chan_sip.c: Merged revisions 262414 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r262414 |
+ pabelanger | 2010-05-11 15:26:17 -0400 (Tue, 11 May 2010) | 8
+ lines Improve logging information for misconfigured contexts
+ (closes issue #17238) Reported by: pprindeville Patches:
+ chan_sip-bug17238.patch uploaded by pprindeville (license 347)
+ Tested by: pprindeville ........
+
+2010-05-11 17:25 +0000 [r262339] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /, Makefile.rules: Merged revisions 262330
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r262330 | tilghman | 2010-05-11 12:23:51 -0500
+ (Tue, 11 May 2010) | 9 lines Merged revisions 262321 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r262321 | tilghman | 2010-05-11 12:22:07 -0500 (Tue, 11
+ May 2010) | 2 lines Fix issue #17302 a slightly different way
+ (mad props to Qwell) ........ ................
+
+2010-05-10 18:52 +0000 [r262238] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_console.c: Merged revisions 262236 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r262236 | dvossel | 2010-05-10 13:36:10 -0500 (Mon, 10 May 2010)
+ | 11 lines fixes crash in chan_console There is a race condition
+ between console_hangup() and start_stream(). It is possible for
+ console_hangup() to be called and then the stream thread to begin
+ after the hangup. To avoid this a check in start_stream() to make
+ sure the pvt-owner still exists while the pvt lock is held is
+ made. If the owner is gone that means the channel hung up and
+ start_stream should be aborted. ........
+
+2010-05-10 16:38 +0000 [r262154] Tilghman Lesher <tlesher@digium.com>
+
+ * /, Makefile.rules: Merged revisions 262152 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r262152 | tilghman | 2010-05-10 11:36:25 -0500 (Mon, 10 May 2010)
+ | 17 lines Merged revisions 262151 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r262151 | tilghman | 2010-05-10 11:34:21 -0500 (Mon, 10 May 2010)
+ | 10 lines Allow compilation on Mac OS X 10.4 (Tiger) (closes
+ issue #17297) Reported by: jcovert Patches:
+ 20100506__issue17297.diff.txt uploaded by tilghman (license 14)
+ (closes issue #17302) Reported by: jcovert ........
+ ................
+
+2010-05-09 02:17 +0000 [r261915-262104] Tilghman Lesher <tlesher@digium.com>
+
+ * autoconf/ast_ext_lib.m4, autoconf/ast_c_compile_check.m4,
+ autoconf/ast_c_define_check.m4, /, configure,
+ include/asterisk/autoconfig.h.in: Merged revisions 262102 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r262102 | tilghman | 2010-05-08 21:14:04 -0500 (Sat, 08
+ May 2010) | 5 lines Cleanup a bit more by getting rid of useless
+ version defines. Also make library detection use passed CFLAGS.
+ (closes issue #17309) Reported by: stuarth ........
+
+ * /, configure, configure.ac: Merged revisions 262048 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r262048 | tilghman | 2010-05-07 21:40:01 -0500 (Fri, 07 May 2010)
+ | 2 lines Use CPPFLAGS to pass PTHREAD_CFLAGS for vpb only
+ ........
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 261913 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r261913 |
+ tilghman | 2010-05-07 15:35:17 -0500 (Fri, 07 May 2010) | 14
+ lines Use the detected pthread building flags in every place,
+ instead of hardcoding -lpthread. We nicely detect the right flags
+ on each system for building Asterisk with pthreads, then ignore
+ it for every other build option that requires us to build with
+ pthreads. This caused some items to return a false negative. Also
+ cleanup some minor naming issues that caused "library library"
+ redundancy in the output. (closes issue #17303) Reported by:
+ stuarth Patches: 20100507__issue17303.diff.txt uploaded by
+ tilghman (license 14) Tested by: stuarth ........
+
+2010-05-06 20:13 +0000 [r261738] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 261736 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r261736 | jpeeler | 2010-05-06 15:11:53 -0500
+ (Thu, 06 May 2010) | 15 lines Merged revisions 261735 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r261735 | jpeeler | 2010-05-06 15:10:59 -0500 (Thu, 06 May 2010)
+ | 8 lines Only allow the operator key to be accepted after
+ leaving a voicemail. Or rather disallow the operator key from
+ being accepted when not offered, such as after finishing a
+ recording from within the mailbox options menu. ABE-2121 SWP-1267
+ ........ ................
+
+2010-05-06 17:08 +0000 [r261611] Jason Parker <jparker@digium.com>
+
+ * sounds/Makefile, /: Merged revisions 261609 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r261609 | qwell | 2010-05-06 12:06:40 -0500 (Thu, 06 May 2010) |
+ 11 lines Merged revisions 261608 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r261608 | qwell | 2010-05-06 11:56:02 -0500 (Thu, 06 May 2010) |
+ 4 lines Use the versioned MOH tarballs, now that we have them.
+ This makes for more reproducibility. Prompted by a discussion in
+ #asterisk-dev ........ ................
+
+2010-05-06 15:42 +0000 [r261562] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 261560 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r261560 |
+ tilghman | 2010-05-06 10:39:10 -0500 (Thu, 06 May 2010) | 8 lines
+ Permit more lines within a SIP body to be parsed. The example
+ given within the related issue showed 120 lines, which was mostly
+ a result of the body being XML. (closes issue #17179) Reported
+ by: khw ........
+
+2010-05-06 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.1.20-rc1 Released
+
+2010-05-06 14:02 +0000 [r261497] Russell Bryant <russell@digium.com>
+
+ * /, main/heap.c: Merged revisions 261496 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r261496 |
+ russell | 2010-05-06 08:58:07 -0500 (Thu, 06 May 2010) | 40 lines
+ Fix handling of removing nodes from the middle of a heap. This
+ bug surfaced in 1.6.2 and does not affect code in any other
+ released version of Asterisk. It manifested itself as SIP qualify
+ not happening when it should, causing peers to go unreachable.
+ This was debugged down to scheduler entries sometimes not getting
+ executed when they were supposed to, which was in turn caused by
+ an error in the heap code. The problem only sometimes occurs, and
+ it is due to the logic for removing an entry in the heap from an
+ arbitrary location (not just popping off the top). The scheduler
+ performs this operation frequently when entries are removed
+ before they run (when ast_sched_del() is used). In a normal pop
+ off of the top of the heap, a node is taken off the bottom,
+ placed at the top, and then bubbled down until the max heap
+ property is restored (see max_heapify()). This same logic was
+ used for removing an arbitrary node from the middle of the heap.
+ Unfortunately, that logic is full of fail. This patch fixes that
+ by fully restoring the max heap property when a node is thrown
+ into the middle of the heap. Instead of just pushing it down as
+ appropriate, it first pushes it up as high as it will go, and
+ _then_ pushes it down. Lastly, fix a minor problem in
+ ast_heap_verify(), which is only used for debugging. If a parent
+ and child node have the same value, that is not an error. The
+ only error is if a parent's value is less than its children. A
+ huge thanks goes out to cappucinoking for debugging this down to
+ the scheduler, and then producing an ast_heap test case that
+ demonstrated the breakage. That made it very easy for me to focus
+ on the heap logic and produce a fix. Open source projects are
+ awesome. (closes issue #16936) Reported by: ib2 Tested by:
+ cappucinoking, crjw (closes issue #17277) Reported by:
+ cappucinoking Patches: heap-fix.rev2.diff uploaded by russell
+ (license 2) Tested by: cappucinoking, russell ........
+
+2010-05-06 07:35 +0000 [r261452] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 261451 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r261451 | tzafrir | 2010-05-06 10:27:31 +0300 (ה', 06 מאי 2010) |
+ 4 lines When failing to configure, don't destroy 'cfg' twice
+ Fixes a crash when some config section had an incorrect channel
+ config. ........
+
+2010-05-05 19:14 +0000 [r261317] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * /, channels/chan_sip.c: Merged revisions 261314 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r261314 | pabelanger | 2010-05-05 14:43:03 -0400 (Wed, 05 May
+ 2010) | 19 lines Merged revisions 261274 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r261274 | pabelanger | 2010-05-05 12:42:22 -0400 (Wed, 05 May
+ 2010) | 12 lines Registration fix for SIP realtime. Make sure
+ realtime fields are not empty. (closes issue #17266) Reported by:
+ Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick
+ Lewis (license 657) Tested by: Nick_Lewis, sberney Review:
+ https://reviewboard.asterisk.org/r/643/ ........ ................
+
+2010-05-04 23:55 +0000 [r261097] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 261095 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r261095 | tilghman | 2010-05-04 18:51:52 -0500 (Tue, 04 May 2010)
+ | 18 lines Merged revisions 261093-261094 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r261093 | tilghman | 2010-05-04 18:36:53 -0500 (Tue, 04 May 2010)
+ | 7 lines Protect against overflow, when calculating how long to
+ wait for a frame. (closes issue #17128) Reported by: under
+ Patches: d.diff uploaded by under (license 914) ........ r261094
+ | tilghman | 2010-05-04 18:47:08 -0500 (Tue, 04 May 2010) | 2
+ lines Add a tiny corner case to the previous commit ........
+ ................
+
+2010-05-04 18:57 +0000 [r260926] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 260924 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260924 | jpeeler | 2010-05-04 13:51:28 -0500
+ (Tue, 04 May 2010) | 18 lines Merged revisions 260923 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260923 | jpeeler | 2010-05-04 13:46:46 -0500 (Tue, 04 May 2010)
+ | 12 lines Voicemail transfer to operator should occur
+ immediately, not after main menu. There were two scenarios in the
+ advanced options that while using the operator=yes and review=yes
+ options, the transfer occurred only after exiting the main menu
+ (after sending a reply or leaving a message for an extension).
+ Now after the audio is processed for the reply or message the
+ transfer occurs immediately as expected. ABE-2107 ABE-2108
+ ........ ................
+
+2010-05-04 15:51 +0000 [r260745-260804] Jason Parker <jparker@digium.com>
+
+ * /, build_tools/make_build_h: Merged revisions 260802 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260802 | qwell | 2010-05-04 10:49:57 -0500
+ (Tue, 04 May 2010) | 9 lines Merged revisions 260801 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r260801 | qwell | 2010-05-04 10:49:27 -0500 (Tue, 04 May
+ 2010) | 1 line Fix fallout from removing from configure script.
+ Pointed out by philipp64 on #asterisk-dev ........
+ ................
+
+ * /: Fix merge props
+
+2010-05-03 17:37 +0000 [r260741] Paul Belanger <paul.belanger@polybeacon.com>
+
+ * Makefile, /: Merged revisions 260661-260662 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260661 | pabelanger | 2010-05-03 12:41:30 -0400 (Mon, 03 May
+ 2010) | 10 lines non-root make install PREFIX=/tmp fails. Prepend
+ libdir when executing mkpkgconfig allowing non-root installs to
+ work. (closes issue #17268) Reported by: pabelanger Patches:
+ issue17268.patch uploaded by pabelanger (license 224) Tested by:
+ pabelanger ........ r260662 | pabelanger | 2010-05-03 12:54:41
+ -0400 (Mon, 03 May 2010) | 3 lines Should have removed /usr/lib/
+ part. Thanks Qwell. ........
+
+2010-05-03 14:59 +0000 [r260572] Leif Madsen <lmadsen@digium.com>
+
+ * doc/HOWTO_collect_debug_information.txt: Merged revisions 260570
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260570 | lmadsen | 2010-05-03 09:58:23 -0500
+ (Mon, 03 May 2010) | 9 lines Merged revisions 260569 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r260569 | lmadsen | 2010-05-03 09:57:39 -0500 (Mon, 03
+ May 2010) | 1 line Minor typo pointed out by pabelanger on IRC.
+ ........ ................
+
+2010-04-30 22:47 +0000 [r260440] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 260437 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260437 | jpeeler | 2010-04-30 17:36:49 -0500
+ (Fri, 30 Apr 2010) | 18 lines Merged revisions 260434 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010)
+ | 11 lines Ensure channel state is not incorrectly set in the
+ case of a very early answer. The needringing bit was being read
+ in dahdi_read after answering thereby setting the state to
+ ringing from up. This clears needringing upon answering so that
+ is no longer possible. (closes issue #17067) Reported by: tzafrir
+ Patches: needringing.diff uploaded by tzafrir (license 46)
+ ........ ................
+
+2010-04-30 20:18 +0000 [r260354] Mark Michelson <mmichelson@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 260346 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260346 | mmichelson | 2010-04-30 15:11:02 -0500
+ (Fri, 30 Apr 2010) | 24 lines Merged revisions 260345 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260345 | mmichelson | 2010-04-30 15:08:15 -0500 (Fri, 30 Apr
+ 2010) | 18 lines Fix potential crash from race condition due to
+ accessing channel data without the channel locked. In
+ res_musiconhold.c, there are several places where a channel's
+ stream's existence is checked prior to calling ast_closestream on
+ it. The issue here is that in several cases, the channel was not
+ locked while checking the stream. The result was that if two
+ threads checked the state of the channel's stream at
+ approximately the same time, then there could be a situation
+ where both threads attempt to call ast_closestream on the
+ channel's stream. The result here is that the refcount for the
+ stream would go below 0, resulting in a crash. I have added
+ proper channel locking to res_musiconhold.c to ensure that we do
+ not try to check chan->stream without the channel locked. A
+ Digium customer has been using this patch for several weeks and
+ has not had any crashes since applying the patch. ABE-2147
+ ........ ................
+
+2010-04-29 23:03 +0000 [r260233] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 260231 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260231 | rmudgett | 2010-04-29 17:44:14 -0500
+ (Thu, 29 Apr 2010) | 33 lines Merged revisions 260195 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010)
+ | 26 lines DTMF CallerID detection problems. The code handling
+ DTMF CallerID drops digits on long CallerID numbers and may
+ timeout waiting for the first ring with shorter numbers. The DTMF
+ emulation mode was not turned off when processing DTMF CallerID.
+ When the emulation code gets behind in processing the DTMF digits
+ it can skip a digit. For shorter numbers, the timeout may have
+ been too short. I increased it from 2 seconds to 4 seconds. Four
+ seconds is a typical time between rings for many countries.
+ (closes issue #16460) Reported by: sum Patches: issue16460.patch
+ uploaded by rmudgett (license 664) issue16460_v1.6.2.patch
+ uploaded by rmudgett (license 664) Tested by: sum, rmudgett
+ Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA
+ AST-334 JIRA SWP-901 ........ ................
+
+2010-04-29 18:18 +0000 [r260154] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 260148 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r260148 | tilghman | 2010-04-29 13:15:57 -0500 (Thu, 29
+ Apr 2010) | 2 lines Pattern match fail. ........
+
+2010-04-29 15:37 +0000 [r260052] David Vossel <dvossel@digium.com>
+
+ * /, include/asterisk/audiohook.h, main/audiohook.c: Merged
+ revisions 260050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r260050 | dvossel | 2010-04-29 10:33:27 -0500 (Thu, 29 Apr 2010)
+ | 21 lines Merged revisions 260049 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260049 | dvossel | 2010-04-29 10:31:02 -0500 (Thu, 29 Apr 2010)
+ | 14 lines Fixes crash in audiohook_write_list The middle_frame
+ in the audiohook_write_list function was being freed if a
+ audiohook manipulator returned a failure. This is incorrect
+ logic. This patch resolves this and adds detailed descriptions of
+ how this function should work and why manipulator failures must
+ be ignored. (closes issue #17052) Reported by: dvossel Tested by:
+ dvossel (closes issue #16196) Reported by: atis Review:
+ https://reviewboard.asterisk.org/r/623/ ........ ................
+
+2010-04-28 22:35 +0000 [r259958] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 259957 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r259957 |
+ mmichelson | 2010-04-28 17:34:15 -0500 (Wed, 28 Apr 2010) | 11
+ lines Don't override peer context with domain context. (closes
+ issue #17040) Reported by: pprindeville Patches:
+ asterisk-1.6-bugid17040.patch uploaded by pprindeville (license
+ 347) Tested by: pprindeville Review:
+ https://reviewboard.asterisk.org/r/565/ ........
+
+2010-04-28 21:33 +0000 [r259930] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, channels/chan_local.c, /: Merged revisions 259870
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259870 | dvossel | 2010-04-28 16:20:03 -0500
+ (Wed, 28 Apr 2010) | 39 lines Merged revisions 259858 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010)
+ | 33 lines resolves deadlocks in chan_local Issue_1. In the
+ local_hangup() 3 locks must be held at the same time... pvt,
+ pvt->chan, and pvt->owner. Proper deadlock avoidance is done when
+ the channel to hangup is the outbound chan_local channel, but
+ when it is not the outbound channel we have an issue... We
+ attempt to do deadlock avoidance only on the tech pvt, when both
+ the tech pvt and the pvt->owner are locked coming into that loop.
+ By never giving up the pvt->owner channel deadlock avoidance is
+ not entirely possible. This patch resolves that by doing deadlock
+ avoidance on both the pvt->owner and the pvt when trying to get
+ the pvt->chan lock. Issue_2. ast_prod() is used in
+ ast_activate_generator() to queue a frame on the channel and make
+ the channel's read function get called. This function is used in
+ ast_activate_generator() while the channel is locked, which
+ mean's the channel will have a lock both from the generator code
+ and the frame_queue code by the time it gets to chan_local.c's
+ local_queue_frame code... local_queue_frame contains some of the
+ same crazy deadlock avoidance that local_hangup requires, and
+ this recursive lock prevents that deadlock avoidance from
+ happening correctly. This patch removes ast_prod() from the
+ channel lock so only one lock is held during the
+ local_queue_frame function. (closes issue #17185) Reported by:
+ schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel
+ (license 671) issue_17185_v2.diff uploaded by dvossel (license
+ 671) Tested by: schmoozecom, GameGamer43 Review:
+ https://reviewboard.asterisk.org/r/631/ ........ ................
+
+2010-04-28 21:09 +0000 [r259855] Leif Madsen <lmadsen@digium.com>
+
+ * config.guess: Merged revisions 259853 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259853 | lmadsen | 2010-04-28 16:08:34 -0500 (Wed, 28 Apr 2010)
+ | 14 lines Merged revisions 259852 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259852 | lmadsen | 2010-04-28 16:07:48 -0500 (Wed, 28 Apr 2010)
+ | 6 lines Update config.guess. Updating config.guess because
+ after installing Ubuntu Server 9.10 and running all the update
+ scripts, running ./configure would not continue because it was
+ unable to determine what kind of system I had. After updating
+ config.guess things started working again. ........
+ ................
+
+2010-04-28 20:33 +0000 [r259776-259850] Jason Parker <jparker@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 259848 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259848 | qwell | 2010-04-28 15:32:14 -0500
+ (Wed, 28 Apr 2010) | 9 lines Merged revisions 259847 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r259847 | qwell | 2010-04-28 15:30:21 -0500 (Wed, 28 Apr
+ 2010) | 1 line Add AC_CONFIG_AUX_DIR to configure script, so
+ systems without install can use install-sh from our source dir.
+ ........ ................
+
+ * makeopts.in, /: Merged revisions 259837 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259837 | qwell | 2010-04-28 15:26:35 -0500 (Wed, 28 Apr 2010) |
+ 9 lines Merged revisions 259833 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259833 | qwell | 2010-04-28 15:25:36 -0500 (Wed, 28 Apr 2010) |
+ 1 line Missed this when removing $ID ........ ................
+
+ * Makefile, /, configure, configure.ac: Merged revisions 259760 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259760 | qwell | 2010-04-28 14:19:54 -0500
+ (Wed, 28 Apr 2010) | 14 lines Merged revisions 259748 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259748 | qwell | 2010-04-28 14:17:38 -0500 (Wed, 28 Apr 2010) |
+ 7 lines Remove usage of `id` since it isn't useful and was
+ causing breakge. Solaris `id` doesn't support the -u argument.
+ Instead of figuring out how to fix this to work on Solaris, I
+ decided to check why it was necessary and where else it was used.
+ It was only used in one place, and it hasn't been needed for a
+ very long time (I question whether it was ever needed). ........
+ ................
+
+2010-04-28 17:19 +0000 [r259679] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 259672 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259672 | jpeeler | 2010-04-28 12:18:43 -0500
+ (Wed, 28 Apr 2010) | 11 lines Merged revisions 259664 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259664 | jpeeler | 2010-04-28 12:13:29 -0500 (Wed, 28 Apr 2010)
+ | 4 lines Do not play goodbye prompt after timeout of message
+ review. ABE-2124 ........ ................
+
+2010-04-27 22:37 +0000 [r259615] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 259538 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259538 | rmudgett | 2010-04-27 17:18:09 -0500
+ (Tue, 27 Apr 2010) | 18 lines Merged revisions 259531 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27 Apr 2010)
+ | 11 lines DAHDI "WARNING" message is confusing and vague
+ "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed
+ failed: Success" Changed the warning to "Failed to decode
+ CallerID on channel 'name'". The message before it is likely more
+ specific about why the CallerID decode failed. SWP-501 AST-283
+ ........ ................
+
+2010-04-27 21:50 +0000 [r259529] Leif Madsen <lmadsen@digium.com>
+
+ * sounds/Makefile: Merged revisions 259527 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259527 | lmadsen | 2010-04-27 16:49:36 -0500 (Tue, 27 Apr 2010)
+ | 23 lines Merged revisions 259526 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259526 | lmadsen | 2010-04-27 16:48:47 -0500 (Tue, 27 Apr 2010)
+ | 15 lines Update sounds files. * Add additional sounds prompts
+ for say_enumeration * Update the English conference sounds
+ prompts so they are better quality and all sound more consistent
+ * Clean up the core-sounds-XX.txt and extra-sounds-XX.txt files
+ to include all present sound files Both core (en, fr, es) and
+ extra (en, fr) sounds files have been updated. (closes issue
+ #16200) Reported by: murf (closes issue #17137) Reported by:
+ lmadsen ........ ................
+
+2010-04-27 21:22 +0000 [r259355-259471] Jason Parker <jparker@digium.com>
+
+ * /, main/editline/configure, main/editline/Makefile.in,
+ main/editline/configure.in: Merged revisions 259439 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r259439 | qwell | 2010-04-27 16:13:01 -0500 (Tue, 27 Apr 2010) |
+ 5 lines Add gar to the check for AR for those silly OSes
+ (Solaris) that don't have ar. autoconf2.13 couldn't handle
+ AC_PROG_GREP, so I removed it. This is fine, since we don't need
+ to use anything that the configure script doesn't. ........
+
+ * /, configure, configure.ac: Merged revisions 259353 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259353 | qwell | 2010-04-27 14:31:55 -0500
+ (Tue, 27 Apr 2010) | 12 lines Merged revisions 259352 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259352 | qwell | 2010-04-27 14:29:26 -0500 (Tue, 27 Apr 2010) |
+ 5 lines Support the silly OSes that don't have ar and strip.
+ Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when path isn't
+ specified, and AC_PATH_TOOLS doesn't exist, we'll just switch to
+ AC_CHECK_TOOLS. ........ ................
+
+2010-04-27 18:53 +0000 [r259309] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
+ revisions 259307 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259307 | rmudgett | 2010-04-27 13:29:33 -0500 (Tue, 27 Apr 2010)
+ | 21 lines Merged revisions 259270 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010)
+ | 14 lines hidecalleridname parameter in chan_dahdi.conf Issue
+ #7321 implements a new chan_dahdi configuration option. However,
+ a change mentioned in the issue was never implemented. This is
+ the change that will allow the feature to work. I added a note to
+ chan_dahdi.conf.sample about the feature. (closes issue #17143)
+ Reported by: djensen99 Patches: diff.txt uploaded by djensen99
+ (license NA) (One line change) Tested by: djensen99 ........
+ ................
+
+2010-04-26 21:48 +0000 [r259078-259108] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 259105 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259105 | mmichelson | 2010-04-26 16:45:13 -0500 (Mon, 26 Apr
+ 2010) | 9 lines Merged revisions 259104 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259104 | mmichelson | 2010-04-26 16:44:43 -0500 (Mon, 26 Apr
+ 2010) | 3 lines Let compilation succeed warning-free when
+ DONT_OPTIMIZE is turned off. ........ ................
+
+ * main/channel.c, /: Merged revisions 259023 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259023 | mmichelson | 2010-04-26 16:13:35 -0500 (Mon, 26 Apr
+ 2010) | 19 lines Merged revisions 259018 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259018 | mmichelson | 2010-04-26 16:03:08 -0500 (Mon, 26 Apr
+ 2010) | 13 lines Prevent Newchannel manager events for dummy
+ channels. No Newchannel manager event will be fired for channels
+ that are allocated to not match a registered technology type.
+ Thus bogus channels allocated solely for variable substitution or
+ CDR operations do not result in a Newchannel event. (closes issue
+ #16957) Reported by: atis Review:
+ https://reviewboard.asterisk.org/r/601 ........ ................
+
+2010-04-25 18:14 +0000 [r258778] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 258776 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r258776 | tilghman | 2010-04-25 13:12:14 -0500 (Sun, 25 Apr 2010)
+ | 13 lines Merged revisions 258775 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010)
+ | 6 lines When StopMonitor is called, ensure that it will not be
+ restarted by a channel event. (closes issue #16590) Reported by:
+ kkm Patches: resmonitor-16590-trunk.239289.diff uploaded by kkm
+ (license 888) ........ ................
+
+2010-04-22 22:24 +0000 [r258705] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/cdr.c, main/channel.c, /, main/features.c: Merged revisions
+ 258671,258675 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r258671 | mnicholson | 2010-04-22 16:57:59 -0500 (Thu, 22 Apr
+ 2010) | 32 lines Merged revisions 193391,258670 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193391 | mnicholson | 2009-05-08 16:01:25 -0500 (Fri, 08 May
+ 2009) | 8 lines Set the proper disposition on originated calls.
+ (closes issue #14167) Reported by: jpt Patches:
+ call-file-missing-cdr2.diff uploaded by mnicholson (license 96)
+ Tested by: dlotina, rmartinez, mnicholson ........ r258670 |
+ mnicholson | 2010-04-22 16:49:07 -0500 (Thu, 22 Apr 2010) | 11
+ lines Fix broken CDR behavior. This change allows a CDR record
+ previously marked with disposition ANSWERED to be set as BUSY or
+ NO ANSWER. Additionally this change partially reverts r235635 and
+ does not set the AST_CDR_FLAG_ORIGINATED flag on CDRs generated
+ from ast_call(). To preserve proper CDR behavior, the
+ AST_CDR_FLAG_DIALED flag is now cleared from all brige CDRs in
+ ast_bridge_call(). (closes issue #16797) Reported by:
+ VarnishedOtter Tested by: mnicholson ........ (closes issue
+ #16222) Reported by: telles Tested by: mnicholson
+ ................ r258675 | mnicholson | 2010-04-22 17:11:23 -0500
+ (Thu, 22 Apr 2010) | 2 lines Fix previous commit.
+ ................
+
+2010-04-21 22:10 +0000 [r258435] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 258433 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r258433 | jpeeler | 2010-04-21 16:56:09 -0500
+ (Wed, 21 Apr 2010) | 15 lines Merged revisions 258432 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r258432 | jpeeler | 2010-04-21 16:45:36 -0500 (Wed, 21 Apr 2010)
+ | 8 lines Fix looping forever when no input received in certain
+ voicemail menu scenarios. Specifically, prompting for an
+ extension (when leaving or forwarding a message) or when
+ prompting for a digit (when saving a message or changing
+ folders). ABE-2122 SWP-1268 ........ ................
+
+2010-04-21 18:23 +0000 [r258334] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 258305 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r258305 |
+ dvossel | 2010-04-21 13:13:36 -0500 (Wed, 21 Apr 2010) | 12 lines
+ fixes issue with double "sip:" in header field This is a clear
+ mistake in logic. Future discussions about how to avoid having to
+ handle uri's like this should take place in the future, but this
+ fix needs to go in for now. (closes issue #15847) Reported by:
+ ebroad Patches: doublesip.patch uploaded by ebroad (license 878)
+ ........
+
+2010-04-20 17:51 +0000 [r258105] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 258065 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r258065 | jpeeler | 2010-04-20 12:06:19 -0500
+ (Tue, 20 Apr 2010) | 17 lines Merged revisions 258029 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r258029 | jpeeler | 2010-04-20 11:16:33 -0500 (Tue, 20 Apr 2010)
+ | 11 lines Play correct prompt when voicemail store failure
+ occurs after attempted forward. If a user's mailbox was full and
+ a message was attempted to be forwarded to said box, warnings on
+ the console would indicate failure. However, the played prompt
+ was that of success (vm-msgsaved). Now storage failure is taken
+ into account and the correct prompt (vm-mailboxfull) is played
+ when appropriate. ABE-2123 SWP-1262 ........ ................
+
+2010-04-19 18:22 +0000 [r257833] Terry Wilson <twilson@digium.com>
+
+ * /, main/features.c: Merged revisions 257810 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r257810 |
+ twilson | 2010-04-19 12:57:41 -0500 (Mon, 19 Apr 2010) | 5 lines
+ Fix incomplete CDR merge from r195881 Because res/res_features.c
+ was removed and main/cdr.c added, these changes didn't make it to
+ trunk and the 1.6.x branches ........
+
+2010-04-18 17:28 +0000 [r257770] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/cdr_odbc.conf.sample, /: Merged revisions 257768 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r257768 | tilghman | 2010-04-18 12:25:53 -0500 (Sun, 18
+ Apr 2010) | 2 lines Removing unused configuration parameters
+ ........
+
+2010-04-16 21:32 +0000 [r257739] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * apps/app_mixmonitor.c, /: Merged revisions 257713 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r257713 | dhubbard | 2010-04-16 16:22:30 -0500
+ (Fri, 16 Apr 2010) | 28 lines Merged revisions 257686 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16 Apr 2010)
+ | 21 lines Make the mixmonitor thread process audio frames faster
+ Mantis issue 17078 reports MixMonitor recordings have shorter
+ durations than the call duration. This was because the mixmonitor
+ thread was not processing frames from the audiohook fast enough.
+ The mixmonitor thread would slowly fall behind the most recent
+ audio frame and when the channel hangs up, the mixmonitor thread
+ would exit without processing the same number of frames as the
+ channel; leaving the mixmonitor recording shorter than actual
+ call duration. This revision fixes this issue by moving the
+ ast_audiohook_trigger_wait() and the subsequent audiohook.status
+ check into the block where the ast_audiohook_read_frame()
+ function returns NULL. (closes issue #17078) Reported by:
+ geoff2010 Patches: dw-M17078.patch uploaded by dhubbard (license
+ 733) Tested by: dhubbard, geoff2010 Review:
+ https://reviewboard.asterisk.org/r/611/ ........ ................
+
+2010-04-15 21:33 +0000 [r257508-257594] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/app.h, /, main/app.c: Merged revisions 257560
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r257560 | tilghman | 2010-04-15 16:26:19 -0500
+ (Thu, 15 Apr 2010) | 13 lines Merged revisions 257544 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257544 | tilghman | 2010-04-15 16:23:24 -0500 (Thu, 15 Apr 2010)
+ | 6 lines Allow application options with arguments to contain
+ parentheses, through a variety of escaping techniques. Fixes
+ SWP-1194 (ABE-2143). Review:
+ https://reviewboard.asterisk.org/r/604/ ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 257493 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r257493 | tilghman | 2010-04-15 15:30:15 -0500 (Thu, 15 Apr 2010)
+ | 20 lines Merged revisions 257467 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010)
+ | 13 lines Don't recreate peer, when responding to a repeated
+ deregistration attempt. When a reply to a deregistration is lost
+ in transmit, the client retries the deregistration. Previously,
+ this would cause a realtime/autocreate peer to be loaded back
+ into memory, after it had already been correctly purged. Instead,
+ we just want to resend the reply without loading the peer.
+ (closes issue #16908) Reported by: kkm Patches:
+ 20100412__issue16908.diff.txt uploaded by tilghman (license 14)
+ Tested by: kkm ........ ................
+
+2010-04-15 19:43 +0000 [r257345-257429] Leif Madsen <lmadsen@digium.com>
+
+ * doc/backtrace.txt: Merged revisions 257427 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r257427 | lmadsen | 2010-04-15 14:41:05 -0500 (Thu, 15 Apr 2010)
+ | 21 lines Merged revisions 257426 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257426 | lmadsen | 2010-04-15 14:40:33 -0500 (Thu, 15 Apr 2010)
+ | 13 lines Update backtrace.txt documentation. Update the
+ backtrace.txt documentation so it conforms to the same layout as
+ other documents we've been working on recently. Additionally, add
+ a bunch of new information about gathering backtraces for crashes
+ and deadlocks, along with ways of verifying your file before
+ uploading it. Create a couple of one line commands for people to
+ generate the files we need. (closes issue #17190) Reported by:
+ lmadsen Patches: backtrace.txt.patch-2 uploaded by lmadsen
+ (license 10) Tested by: lmadsen, pabelanger ........
+ ................
+
+ * doc/backtrace.txt: Merged revisions 257343 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r257343 | lmadsen | 2010-04-15 08:44:38 -0500 (Thu, 15 Apr 2010)
+ | 9 lines Merged revisions 257342 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257342 | lmadsen | 2010-04-15 08:41:45 -0500 (Thu, 15 Apr 2010)
+ | 1 line Update address of the bug tracker. ........
+ ................
+
+2010-04-14 23:00 +0000 [r257264] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/features.conf.sample, /, main/features.c: Merged
+ revisions 257262 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r257262 |
+ tilghman | 2010-04-14 17:57:35 -0500 (Wed, 14 Apr 2010) | 15
+ lines Yet another issue where the conversion of the application
+ delimiter to comma caused an issue. Application arguments within
+ the feature map could possibly contain a comma, which conflicts
+ with the syntax of the features.conf configuration file. This
+ patch allows the argument to be wrapped in parentheses or quoted,
+ to allow the application arguments to be interpreted as a single
+ configuration parameter. (closes issue #16646) Reported by:
+ pinga-fogo Patches: 20100414__issue16646.diff.txt uploaded by
+ tilghman (license 14) Tested by: tilghman Review:
+ https://reviewboard.asterisk.org/r/547/ ........
+
+2010-04-13 19:30 +0000 [r257214] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/manager.c, /, configs/manager.conf.sample: Merged revisions
+ 257146 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r257146 | mnicholson | 2010-04-13 13:10:30 -0500 (Tue, 13 Apr
+ 2010) | 16 lines Merged revisions 257070 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr
+ 2010) | 9 lines Add an option to restore past broken behavor of
+ the Events manager action Before r238915, certain values for the
+ EventMask parameter of the Events action would result in no
+ response being returned. This patch adds an option to restore
+ that broken behavior. Also while fixing this bug I discovered
+ that passing an empty EventMasks parameter would also result in
+ no response being returned, this has been fixed as well while
+ being preserved when the broken behavior is requested. (closes
+ issue #17023) Reported by: nblasgen Review:
+ https://reviewboard.asterisk.org/r/602/ ........ ................
+
+2010-04-13 19:20 +0000 [r257067-257208] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 257191 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r257191 |
+ tilghman | 2010-04-13 14:17:48 -0500 (Tue, 13 Apr 2010) | 10
+ lines Also unref the pvt when we delete the provisional keepalive
+ job. (closes issue #16774) Reported by: kowalma Patches:
+ 20100315__issue16774.diff.txt uploaded by tilghman (license 14)
+ Tested by: falves11, jamicque Review:
+ https://reviewboard.asterisk.org/r/591/ ........
+
+ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 257065 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r257065 | tilghman | 2010-04-13 11:33:21 -0500 (Tue, 13 Apr 2010)
+ | 8 lines Ensure that we can have commas within cdr values.
+ (closes issue #17001) Reported by: snuffy Patches:
+ 20100412__issue17001.diff.txt uploaded by tilghman (license 14)
+ Tested by: snuffy ........
+
+2010-04-12 17:31 +0000 [r256903] Leif Madsen <lmadsen@digium.com>
+
+ * doc/HOWTO_collect_debug_information.txt (added): Merged revisions
+ 256901 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r256901 | lmadsen | 2010-04-12 12:29:53 -0500 (Mon, 12 Apr 2010)
+ | 23 lines Merged revisions 256900 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r256900 | lmadsen | 2010-04-12 12:29:26 -0500 (Mon, 12 Apr 2010)
+ | 15 lines Add How-To document on collecting debugging info for
+ issues.asterisk.org Paul Belanger has been helping a lot with bug
+ tracking recently and created this document that we can now point
+ to when additional debugging information is required. This
+ document will help those filing issues to know how to get the
+ information required when filing their issues. This will make
+ things easier on the developers. Initial text and changes by
+ pabelanger. Tweaks and editing by myself. (closes issue #17159)
+ Reported by: pabelanger Patches:
+ HOWTO_collect_debug_information.txt.patch uploaded by lmadsen
+ (license 10) Tested by: tzafrir, pabelanger, lmadsen ........
+ ................
+
+2010-04-06 19:40 +0000 [r256372] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/lock.h: Merged revisions 256370 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r256370 | tilghman | 2010-04-06 14:28:42 -0500 (Tue, 06 Apr 2010)
+ | 2 lines Mac OS X does not support comparing a mutex to its
+ initializer. Create a test for this. ........
+
+2010-04-06 18:08 +0000 [r256267-256365] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Fix malformed if test. Regression of issue
+ 15883. Converted if statement to a switch statement for clarity.
+
+ * channels/chan_dahdi.c, /: Merged revisions 256265 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r256265 | rmudgett | 2010-04-05 19:39:44 -0500
+ (Mon, 05 Apr 2010) | 12 lines Merged revisions 256225 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05 Apr 2010)
+ | 5 lines DAHDI/PRI call to pri_channel_bridge() not protected by
+ PRI lock. SWP-1231 ABE-2163 ........ ................
+
+2010-05-03 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.1.19 Released
+
+2010-04-29 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.1.19-rc3 Released
+
+2010-04-29 10:31 +0000 [r260052] 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 [r259930] 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/
+
+2010-04-13 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.1.19-rc2 Released
+
+2010-04-13 [r257208] Tilghman Lesher <tlesher@digium.com
+
+ Also unref the pvt when we delete the provisional keepalive job.
+
+ (closes issue #16774)
+ Reported by: kowalma
+ Patches:
+ 20100315__issue16774.diff.txt uploaded by tilghman (license 14)
+ Tested by: falves11, jamicque
+
+ Review: https://reviewboard.asterisk.org/r/591/
+
+2010-04-05 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.1.19-rc1 Released
+
+2010-04-05 15:15 +0000 [r256163] Leif Madsen <lmadsen@digium.com>
+
+ * /, doc/tex/localchannel.tex: Merged revisions 256161 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r256161 | lmadsen | 2010-04-05 10:14:53 -0500 (Mon, 05 Apr 2010)
+ | 1 line Fix for localchannel.tex to allow PDFs to be generated
+ again. ........
+
+2010-04-02 23:47 +0000 [r256012-256017] Russell Bryant <russell@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 256015 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r256015 | russell | 2010-04-02 18:46:45 -0500
+ (Fri, 02 Apr 2010) | 16 lines Merged revisions 256014 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r256014 | russell | 2010-04-02 18:45:56 -0500 (Fri, 02 Apr 2010)
+ | 9 lines Resolve a deadlock that occurs due to a pointless call
+ to ast_bridged_channel() (closes issue #16840) Reported by:
+ bzing2 Patches: patch.txt uploaded by bzing2 (license 902)
+ issue_16840.rev1.diff uploaded by russell (license 2) Tested by:
+ bzing2, russell ........ ................
+
+ * main/channel.c, /: Merged revisions 256010 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r256010 | russell | 2010-04-02 18:30:58 -0500 (Fri, 02 Apr 2010)
+ | 9 lines Merged revisions 256009 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r256009 | russell | 2010-04-02 18:30:15 -0500 (Fri, 02 Apr 2010)
+ | 2 lines Remove extremely verbose debug message. ........
+ ................
+
+2010-04-02 20:20 +0000 [r255954] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 255952 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r255952 |
+ tilghman | 2010-04-02 15:19:01 -0500 (Fri, 02 Apr 2010) | 8 lines
+ Pass the PID of the Asterisk process, not the PID of the canary.
+ (closes issue #17065) Reported by: globalnetinc Patches:
+ astcanary.patch uploaded by makoto (license 38) Tested by: frawd,
+ globalnetinc ........
+
+2010-04-01 18:21 +0000 [r255675-255815] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 255796 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r255796 | tilghman | 2010-04-01 13:16:37 -0500 (Thu, 01 Apr 2010)
+ | 7 lines Fix DEBUG_THREADS build on Darwin. (closes issue
+ #16828) Reported by: oej Patches: 20100331__issue16828.diff.txt
+ uploaded by tilghman (license 14) ........
+
+ * apps/app_voicemail.c, /: Recorded merge of revisions 255592 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r255592 | tilghman | 2010-03-31 14:13:02 -0500
+ (Wed, 31 Mar 2010) | 22 lines Recorded merge of revisions 255591
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r255591 | tilghman | 2010-03-31 14:09:46 -0500 (Wed, 31 Mar 2010)
+ | 15 lines Ensure line terminators in email are consistent. Fixes
+ an issue with certain Mail Transport Agents, where attachments
+ are not interpreted correctly. (closes issue #16557) Reported by:
+ jcovert Patches: 20100308__issue16557__1.4.diff.txt uploaded by
+ tilghman (license 14) 20100308__issue16557__1.6.0.diff.txt
+ uploaded by tilghman (license 14)
+ 20100308__issue16557__trunk.diff.txt uploaded by tilghman
+ (license 14) Tested by: ebroad, zktech Reviewboard:
+ https://reviewboard.asterisk.org/r/544/ ........ ................
+
+2010-03-31 17:51 +0000 [r255506] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_dial.c, configs/sip.conf.sample: Merged revisions 255504
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r255504 | lmadsen | 2010-03-31 12:48:09 -0500 (Wed, 31
+ Mar 2010) | 5 lines 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 20:57 +0000 [r255325-255412] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_h323.c: Merged revisions 255410 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r255410 | russell | 2010-03-30 15:56:26 -0500
+ (Tue, 30 Mar 2010) | 9 lines Merged revisions 255409 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r255409 | russell | 2010-03-30 15:56:00 -0500 (Tue, 30
+ Mar 2010) | 2 lines Don't kill Asterisk if the H323 listener does
+ not start. ........ ................
+
+ * /, pbx/pbx_dundi.c: Merged revisions 255323 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r255323 | russell | 2010-03-30 11:07:49 -0500 (Tue, 30 Mar 2010)
+ | 9 lines Merged revisions 255322 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r255322 | russell | 2010-03-30 11:06:06 -0500 (Tue, 30 Mar 2010)
+ | 2 lines Don't make Asterisk not start if pbx_dundi fails to
+ initialize. ........ ................
+
+2010-03-26 19:29 +0000 [r255026-255068] Leif Madsen <lmadsen@digium.com>
+
+ * configs/sip.conf.sample: Finish syncing documentation for
+ tlsbindaddr. (issue #17054)
+
+ * /, configs/sip.conf.sample: Merged revisions 255021 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r255021 | lmadsen | 2010-03-26 14:07:38 -0500 (Fri, 26 Mar 2010)
+ | 8 lines Update confusing documentation for tlsbindaddr. Update
+ some confusing documentation for the tlsbindaddr option in
+ sip.conf.sample. Point at a link instead which has better
+ documentation. (closes issue #17054) Reported by: klaus3000
+ ........
+
+2010-03-25 20:43 +0000 [r254804] Jason Parker <jparker@digium.com>
+
+ * utils/Makefile, /: Merged revisions 254802 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r254802 | qwell | 2010-03-25 15:41:49 -0500 (Thu, 25 Mar 2010) |
+ 9 lines Merged revisions 254800 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r254800 | qwell | 2010-03-25 15:41:15 -0500 (Thu, 25 Mar 2010) |
+ 1 line Don't remove local copies of utils in uninstall. ........
+ ................
+
+2010-03-25 20:09 +0000 [r254720] Russell Bryant <russell@digium.com>
+
+ * channels/chan_usbradio.c, /: Merged revisions 254718 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r254718 | russell | 2010-03-25 15:08:40 -0500 (Thu, 25 Mar 2010)
+ | 2 lines chan_usbradio depends on alsa. ........
+
+2010-03-25 20:03 +0000 [r254717] Jason Parker <jparker@digium.com>
+
+ * 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 17:46 +0000 [r254555] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/acl.h, /: Merged revisions 254553 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r254553 | mmichelson | 2010-03-25 12:42:36 -0500
+ (Thu, 25 Mar 2010) | 11 lines Merged revisions 254552 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r254552 | mmichelson | 2010-03-25 12:33:35 -0500 (Thu, 25 Mar
+ 2010) | 5 lines Add doxygen for acl.h Review:
+ https://reviewboard.asterisk.org/r/528 ........ ................
+
+2010-03-25 17:21 +0000 [r254547] Sean Bright <sean@malleable.com>
+
+ * channels/chan_sip.c: Initialize stream to avoid a compilation
+ error.
+
+2010-03-25 17:10 +0000 [r254541] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix potential crashes from trying to
+ reference non-existent RTP streams.
+
+2010-03-25 16:23 +0000 [r254486] Terry Wilson <twilson@digium.com>
+
+ * /, main/file.c: Merged revisions 254453 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r254453 | twilson | 2010-03-25 11:03:51 -0500 (Thu, 25 Mar 2010)
+ | 9 lines Merged revisions 254451 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r254451 | twilson | 2010-03-25 10:57:29 -0500 (Thu, 25 Mar 2010)
+ | 2 lines Handle new SRCCHANGE control message here too ........
+ ................
+
+2010-03-25 16:15 +0000 [r254456] Mark Michelson <mmichelson@digium.com>
+
+ * main/rtp.c, /: Recorded merge of revisions 254454 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r254454 | mmichelson | 2010-03-25 11:04:48 -0500
+ (Thu, 25 Mar 2010) | 50 lines Recorded merge of revisions 254452
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r254452 | mmichelson | 2010-03-25 10:59:56 -0500 (Thu, 25 Mar
+ 2010) | 44 lines Several fixes regarding RFC2833 DTMF detection.
+ Here is a copy and paste of the details from my request on
+ reviewboard that dealt with these changes: Fix 1. The first
+ change in place is to fix Mantis issue 15811, which deals with a
+ situation where Asterisk will incorrectly interpret out of order
+ RFC2833 frames as duplicate DTMF digits. For instance, we would
+ receive a sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3:
+ DTMF 1 seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1
+ seqno 7: DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch
+ when we received the frame with seqno 5, we would interpret this
+ as a new DTMF 1. With this patch, we will check the seqno of the
+ incoming digit and not process the frame if the seqno is lower
+ than the last recorded seqno. Note that we do not record the
+ seqno of the dropped DTMF frame for future processing. While the
+ above situation is what was designed to be fixed, the patch is
+ written in such a way that the following would also be fixed too:
+ seqno 9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end)
+ seqno 13: DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno
+ 15: DTMF 2 (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In
+ this second situation, the beginning of the DTMF 2 arrives before
+ the final end frame of the DTMF 1. With the patch, seqno 12 is no
+ processed and thus we properly interpret the DTMF. Fix 2. The
+ second change in place is to fix an issue like the following:
+ seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) *packet
+ lost* seqno 4: DTMF 1 (end) *packet lost* seqno 5: DTMF 1 (end)
+ *packet lost* seqno 6: DTMF 2 When we receive seqno 6, we had
+ code in place that was supposed to properly end the previously
+ unended DTMF 1. The problem was that the code was essentially a
+ no-op. The code would set up an end frame for the DTMF 1 but
+ would immediately overwrite the frame with the begin for DTMF 2.
+ I changed process_dtmf_rfc2833() so that instead of returning a
+ single frame, it is given as an output parameter a list of
+ frames. Each frame that needs to be returned is appended to this
+ list. Fix 3. The final change is a minor one where an
+ AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco
+ DTMF or an RFC 3389 frame and no frame was returned, then we
+ would return &ast_null_frame. The problem is that earlier in the
+ function, we may have generated an AST_CONTROL_SRCCHANGE frame
+ and put it in the list of frames we wish to return. This frame
+ would be lost in such a case. The patch fixes this problem
+ ........ ................
+
+2010-03-25 15:22 +0000 [r254448] Leif Madsen <lmadsen@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 254446 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r254446 |
+ lmadsen | 2010-03-25 10:21:26 -0500 (Thu, 25 Mar 2010) | 9 lines
+ handle_speechset has 4 arguments. Update code to reflect that
+ handle_speechset has 4 arguments. (closes issue #17093) Reported
+ by: gpatri Patches: res_agi.patch uploaded by gpatri (license
+ 1014) Tested by: pabelanger, mmichelson ........
+
+2010-03-24 17:18 +0000 [r254283] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 254277 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r254277 | jpeeler | 2010-03-24 12:15:05 -0500 (Wed, 24 Mar 2010)
+ | 78 lines Merged revisions 254235 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r254235 | jpeeler | 2010-03-23 19:37:23 -0500 (Tue, 23 Mar 2010)
+ | 72 lines Ensure that monitor recordings are written to the
+ correct location (again) This is an extension to 248860. As such
+ the dialplan test has been extended: ; non absolute path, not
+ combined exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test)
+ exten => 5040, n, dial(sip/5001) ; absolute path, not combined
+ exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2) exten =>
+ 5041, n, dial(sip/5001) ; no path, not combined exten => 5042, 1,
+ monitor(wav,monitor_test3) exten => 5042, n, dial(sip/5001) ;
+ combined: changemonitor from non absolute to no path (leaves
+ tmp/jeff) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m)
+ exten => 5043, n, changemonitor(monitor_test5) exten => 5043, n,
+ dial(sip/5001) ; combined: changemonitor from no path to non
+ absolute path exten => 5044, 1, monitor(wav,monitor_test6,m)
+ exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this
+ wasn't possible before exten => 5044, n, dial(sip/5001) ; non
+ absolute path, combined exten => 5045, 1,
+ monitor(wav,tmp/jeff/monitor_test8,m) exten => 5045, n,
+ dial(sip/5001) ; absolute path, combined exten => 5046, 1,
+ monitor(wav,/tmp/jeff/monitor_test9,m) exten => 5046, n,
+ dial(sip/5001) ; no path, combined exten => 5047, 1,
+ monitor(wav,monitor_test10,m) exten => 5047, n, dial(sip/5001) ;
+ combined: changemonitor from non absolute to absolute (leaves
+ tmp/jeff) exten => 5048, 1,
+ monitor(wav,tmp/jeff/monitor_test11,m) exten => 5048, n,
+ changemonitor(/tmp/jeff/monitor_test12) exten => 5048, n,
+ dial(sip/5001) ; combined: changemonitor from absolute to non
+ absolute (leaves /tmp/jeff) exten => 5049, 1,
+ monitor(wav,/tmp/jeff/monitor_test13,m) exten => 5049, n,
+ changemonitor(tmp/jeff/monitor_test14) exten => 5049, n,
+ dial(sip/5001) ; combined: changemonitor from no path to absolute
+ exten => 5050, 1, monitor(wav,monitor_test15,m) exten => 5050, n,
+ changemonitor(/tmp/jeff/monitor_test16) exten => 5050, n,
+ dial(sip/5001) ; combined: changemonitor from absolute to no path
+ (leaves /tmp/jeff) exten => 5051, 1,
+ monitor(wav,/tmp/jeff/monitor_test17,m) exten => 5051, n,
+ changemonitor(monitor_test18) exten => 5051, n, dial(sip/5001) ;
+ not combined: changemonitor from non absolute to no path (leaves
+ tmp/jeff) exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19)
+ exten => 5052, n, changemonitor(monitor_test20) exten => 5052, n,
+ dial(sip/5001) ; not combined: changemonitor from no path to non
+ absolute exten => 5053, 1, monitor(wav,monitor_test21) exten =>
+ 5053, n, changemonitor(tmp/jeff/monitor_test22) exten => 5053, n,
+ dial(sip/5001) ; not combined: changemonitor from non absolute to
+ absolute (leaves tmp/jeff) exten => 5054, 1,
+ monitor(wav,tmp/jeff/monitor_test23) exten => 5054, n,
+ changemonitor(/tmp/jeff/monitor_test24) exten => 5054, n,
+ dial(sip/5001) ; not combined: changemonitor from absolute to non
+ absolute (leaves /tmp/jeff) exten => 5055, 1,
+ monitor(wav,/tmp/jeff/monitor_test24) exten => 5055, n,
+ changemonitor(tmp/jeff/monitor_test25) exten => 5055, n,
+ dial(sip/5001) ; not combined: changemonitor from no path to
+ absolute exten => 5056, 1, monitor(wav,monitor_test26) exten =>
+ 5056, n, changemonitor(/tmp/jeff/monitor_test27) exten => 5056,
+ n, dial(sip/5001) ; not combined: changemonitor from absolute to
+ no path (leaves /tmp/jeff) exten => 5057, 1,
+ monitor(wav,/tmp/jeff/monitor_test28) exten => 5057, n,
+ changemonitor(monitor_test29) exten => 5057, n, dial(sip/5001)
+ ........ ................
+
+2010-03-23 21:37 +0000 [r254130] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * tests/Makefile, /: Merged revisions 254001 via svnmerge from
+ http://svn.digium.com/svn/asterisk/trunk ........ r254001 |
+ tzafrir | 2010-03-23 21:19:52 +0200 (Tue, 23 Mar 2010) | 2 lines
+ Change the name of the category 'TEST' to match the name of the
+ subdir ........
+
+2010-03-23 21:20 +0000 [r254065] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 254050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r254050 |
+ jpeeler | 2010-03-23 16:17:23 -0500 (Tue, 23 Mar 2010) | 14 lines
+ Exit native bridging early for greater timing accuracy with
+ warnings This changes native bridging to break one millisecond
+ early so that the more accurate timeval calculations done in the
+ generic bridge can be performed using the bridge config.
+ Currently the time between exiting native bridging slightly late
+ can sometimes cause a large enough discrepancy for warnings to be
+ missed. For the record, 1.4 does not attempt to native bridge at
+ all when warnings are enabled. (closes issue #15815) Reported by:
+ adomjan Review: https://reviewboard.asterisk.org/r/577/ ........
+
+2010-03-22 19:56 +0000 [r253802] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/features.c: Merged revisions 253800 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r253800 | mnicholson | 2010-03-22 14:52:52 -0500 (Mon, 22 Mar
+ 2010) | 11 lines Merged revisions 253799 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r253799 | mnicholson | 2010-03-22 14:50:00 -0500 (Mon, 22 Mar
+ 2010) | 4 lines Unconditionally copy the caller's account code to
+ the called party. (related to issue #16331) ........
+ ................
+
+2010-03-22 19:06 +0000 [r253713-253759] Tilghman Lesher <tlesher@digium.com>
+
+ * /, contrib/scripts/dbsep.cgi: Merged revisions 253758 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r253758 | tilghman | 2010-03-22 14:05:27 -0500 (Mon, 22
+ Mar 2010) | 2 lines Update query should be an UPDATE, not a
+ SELECT. ........
+
+ * /, contrib/scripts/dbsep.cgi: Merged revisions 253755 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r253755 | tilghman | 2010-03-22 13:58:48 -0500 (Mon, 22
+ Mar 2010) | 4 lines Return the list for later manipulation. This
+ fixes an issue with the update procedure. Debugging with
+ mmichelson. ........
+
+ * configs/dbsep.conf.sample, /, contrib/scripts/dbsep.cgi: Merged
+ revisions 253712 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r253712 |
+ tilghman | 2010-03-22 11:59:35 -0500 (Mon, 22 Mar 2010) | 2 lines
+ Accomodate equal signs in DSNs and add documentation, based upon
+ mmichelson's feedback. ........
+
+2010-03-20 17:55 +0000 [r253621-253624] Russell Bryant <russell@digium.com>
+
+ * main/tcptls.c, /, main/features.c, cdr/cdr_pgsql.c,
+ main/stdtime/localtime.c: Merged revisions 253540 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r253540 | russell | 2010-03-20 07:03:07 -0500 (Sat, 20 Mar 2010)
+ | 2 lines Resolve more compiler warnings on FreeBSD. ........
+
+ * apps/app_followme.c, apps/app_dial.c, /: Merged revisions 253538
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r253538 | russell | 2010-03-20 06:43:08 -0500 (Sat, 20
+ Mar 2010) | 2 lines Resolve compiler warnings on FreeBSD.
+ ........
+
+ * /, pbx/pbx_dundi.c: Merged revisions 253537 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r253537 |
+ russell | 2010-03-20 06:39:39 -0500 (Sat, 20 Mar 2010) | 2 lines
+ Resolve a compiler warning on FreeBSD. ........
+
+ * channels/chan_dahdi.c, /: Merged revisions 253536 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r253536 | russell | 2010-03-20 06:33:30 -0500 (Sat, 20 Mar 2010)
+ | 4 lines Use SHRT_MAX instead of MAXSHORT. These changes fix
+ build issues I had with this module on FreeBSD. ........
+
+2010-03-19 07:59 +0000 [r253491] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/astobj2.c, /: Merged revisions 253490 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r253490 |
+ alecdavis | 2010-03-19 20:37:00 +1300 (Fri, 19 Mar 2010) | 19
+ lines prevent segfault if bad magic number is encountered.
+ internal_ao2_ref uses INTERNAL_OBJ which mzy report 'bad magic
+ number', but internal_ao2_ref continues on, causing segfault.
+ Although AO2_MAGIC number is checked by INTERNAL_OBJ before
+ internal_ao2_ref is called, A02_MAGIC is being destroyed (or a
+ wrong pointer) by the time internal_ao2_ref uses INTERNAL_OBJ.
+ internal_ao2_ref now returns -1 if INTERNAL_OBJ encouters a bad
+ magic number. (issue #17037) Reported by: alecdavis Patches:
+ bug17037.diff.txt uploaded by alecdavis (license 585) Tested by:
+ alecdavis ........
+
+2010-03-18 17:56 +0000 [r253258-253347] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_userevent.c: Slightly different fix for UserEvent docs
+ update. (issue #16961)
+
+ * doc/tex/localchannel.tex: Merged revisions 253256 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r253256 | lmadsen | 2010-03-18 10:46:52 -0500 (Thu, 18 Mar 2010)
+ | 9 lines Update to new Local channel documentation. Add same
+ changes as commit to 1.4, but convert to TeX. (issue #16963)
+ Reported by: kobaz Patches: localchannel-2.txt uploaded by kobaz
+ (license 834) ........
+
+2010-03-17 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 00:30 +0000 [r253030] Leif Madsen <lmadsen@digium.com>
+
+ * configs/say.conf.sample: Merged revisions 253028 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r253028 | lmadsen | 2010-03-16 19:29:06 -0500
+ (Tue, 16 Mar 2010) | 13 lines Merged revisions 253018 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r253018 | lmadsen | 2010-03-16 19:26:19 -0500 (Tue, 16 Mar 2010)
+ | 6 lines Add french snipset to say.conf. Add the french snipset
+ to say.conf. (Closes issue #15799) ........ ................
+
+2010-03-16 23:53 +0000 [r252977] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_stack.c, /: Merged revisions 252976 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r252976 |
+ tilghman | 2010-03-16 18:49:35 -0500 (Tue, 16 Mar 2010) | 8 lines
+ Mask out previous arguments on each nested invocation of Gosub.
+ (closes issue #16758) Reported by: wdoekes Patches:
+ 20100316__issue16758.diff.txt uploaded by tilghman (license 14)
+ Review: https://reviewboard.asterisk.org/r/561/ ........
+
+2010-03-16 19:01 +0000 [r252769] Russell Bryant <russell@digium.com>
+
+ * utils/Makefile, /: Merged revisions 252767 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r252767 | russell | 2010-03-16 14:01:04 -0500 (Tue, 16 Mar 2010)
+ | 13 lines Merged revisions 252766 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r252766 | russell | 2010-03-16 14:00:43 -0500 (Tue, 16 Mar 2010)
+ | 6 lines Don't treat warnings as errors for muted. muted
+ supports OS X, but uses functions marked as deprecated in 10.6.
+ However, the functions are still supported, so just ignore the
+ warnings for now and allow the build to proceed. ........
+ ................
+
+2010-03-16 18:49 +0000 [r252764] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/extensions.ael.sample: Merged revisions 252762 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r252762 | lmadsen | 2010-03-16 13:48:22 -0500
+ (Tue, 16 Mar 2010) | 15 lines Merged revisions 252761 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r252761 | lmadsen | 2010-03-16 13:46:20 -0500 (Tue, 16 Mar 2010)
+ | 7 lines Additional extensions.ael global variable fixes. Fixing
+ up a couple more overlapping global variable namespaces shared
+ with extensions.conf.sample. Also noticed a few of the lines that
+ were commented out didn't have the closing semi-colon so I added
+ that as well. (issue #17035) ........ ................
+
+2010-03-15 21:59 +0000 [r252625] Sean Bright <sean@malleable.com>
+
+ * /, apps/app_meetme.c: Merged revisions 252623 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r252623 |
+ seanbright | 2010-03-15 17:55:44 -0400 (Mon, 15 Mar 2010) | 4
+ lines Resolve a crash in SLATrunk when the specified trunk
+ doesn't exist. Reported by philipp64 in #asterisk-dev. ........
+
+2010-03-15 21:54 +0000 [r252621] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/org.asterisk.asterisk.plist, /: Merged revisions
+ 252619 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r252619 | tilghman | 2010-03-15 16:51:55 -0500 (Mon, 15 Mar 2010)
+ | 9 lines Merged revisions 252617 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r252617 | tilghman | 2010-03-15 16:43:14 -0500 (Mon, 15 Mar 2010)
+ | 2 lines Uh, yeah. Umask. I'm stupid. ........ ................
+
+2010-03-15 20:53 +0000 [r252536] Leif Madsen <lmadsen@digium.com>
+
+ * configs/extensions.ael.sample: Merged revisions 252534 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r252534 | lmadsen | 2010-03-15 15:52:32 -0500
+ (Mon, 15 Mar 2010) | 15 lines Merged revisions 252533 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r252533 | lmadsen | 2010-03-15 15:48:56 -0500 (Mon, 15 Mar 2010)
+ | 7 lines Update extensions.ael file to not overlap
+ extensions.conf. Updated the extensions.ael file so the global
+ variables don't overlap those that we have in extensions.conf
+ (sample files). This way unexpected things won't happed hopefully
+ if both pbx_ael and res_config are loaded. (closes issue #17035)
+ Reported by: pprindeville ........ ................
+
+2010-03-15 05:03 +0000 [r252364-252443] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 252442 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r252442 |
+ tilghman | 2010-03-14 23:25:35 -0500 (Sun, 14 Mar 2010) | 7 lines
+ THIS IS NOT PYTHON. Indentation doesn't matter, only braces do.
+ (closes issue #17025) Reported by: smurfix Patches: sip.patch
+ uploaded by smurfix (license 547) ........
+
+ * main/asterisk.c, Makefile,
+ contrib/init.d/org.asterisk.asterisk.plist (added), /: Merged
+ revisions 252362 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r252362 | tilghman | 2010-03-14 20:37:04 -0500 (Sun, 14 Mar 2010)
+ | 11 lines Merged revisions 252361 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r252361 | tilghman | 2010-03-14 20:33:50 -0500 (Sun, 14 Mar 2010)
+ | 4 lines Launch Asterisk on Mac OS X with launchd. Reviewboard:
+ https://reviewboard.asterisk.org/r/551/ ........ ................
+
+2010-03-14 17:47 +0000 [r252316] Sean Bright <sean@malleable.com>
+
+ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 252314 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r252314 | seanbright | 2010-03-14 13:43:46 -0400 (Sun, 14 Mar
+ 2010) | 8 lines Fix building CDR and CEL SQLite3 modules. They
+ added a sqlite3_log() function which was conflicting with our
+ function names. (closes issue #17017) Reported by: alephlg
+ ........
+
+2010-03-13 00:31 +0000 [r252135-252177] Terry Wilson <twilson@digium.com>
+
+ * main/rtp.c: Remove unusued field
+
+ * include/asterisk/frame.h, 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: 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 19:54 +0000 [r251996] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Forward declaring dahdi_pri was already
+ done.
+
+2010-03-12 19:50 +0000 [r251993] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 251989 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r251989 | tilghman | 2010-03-12 13:43:23 -0600 (Fri, 12 Mar 2010)
+ | 8 lines Don't override a user option with the global option.
+ (closes issue #16849) Reported by: ip-rob Patches:
+ 20100311__issue16849.diff.txt uploaded by tilghman (license 14)
+ Tested by: ip-rob ........
+
+2010-03-12 19:44 +0000 [r251990] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 251987 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r251987 | rmudgett | 2010-03-12 13:40:16 -0600
+ (Fri, 12 Mar 2010) | 9 lines Merged revisions 251986 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r251986 | rmudgett | 2010-03-12 13:33:22 -0600 (Fri, 12
+ Mar 2010) | 1 line Make chan_dahdi wakeup_sub() prototype not
+ conditional. ........ ................
+
+2010-03-11 21:08 +0000 [r251875-251886] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_exec.c, /: Merged revisions 251884 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r251884 |
+ tilghman | 2010-03-11 15:07:07 -0600 (Thu, 11 Mar 2010) | 8 lines
+ Because ExecIf needs to reprocess arguments, it's best if we
+ don't remove quotes during parsing. (closes issue #16905)
+ Reported by: ip-rob Patches: 20100303__issue16905.diff.txt
+ uploaded by tilghman (license 14) Tested by: ip-rob ........
+
+ * apps/app_system.c, /: Merged revisions 251877 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r251877 |
+ tilghman | 2010-03-11 14:25:02 -0600 (Thu, 11 Mar 2010) | 8 lines
+ If the argument to the system application is quoted, ensure we
+ remove the quotes before trying to execute. (closes issue #16842)
+ Reported by: ip-rob Patches: 20100310__issue16842.diff.txt
+ uploaded by tilghman (license 14) Tested by: ip-rob ........
+
+ * funcs/func_odbc.c: Verify whether the created buffer was actually
+ large enough to hold the expanded value. For certain types of
+ queries, where the size of the substituted query was much larger
+ than the template, it was possible for the substitution buffer to
+ be too small. This is only an issue in 1.6.1, as previously we
+ used a static buffer anyway, and we have a substitution routine
+ in 1.6.2 forward that automatically sizes itself appropriately to
+ handle larger expansions. (closes issue #17006) Reported by:
+ viniciusfontes Patches: 20100311__issue17006.diff.txt uploaded by
+ tilghman (license 14) Tested by: tilghman
+
+2010-03-11 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.1.18 released
+
+2010-03-05 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.1.18-rc2 released
+
+2010-03-05 Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 250913 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r250913 | tilghman
+ | 2010-03-04 22:37:36 -0600 (Thu, 04 Mar 2010) | 7 lines Missing quote in
+ ODBC query. (closes issue #16953) Reported by: elguero Patches:
+ app_voicemail-odbc-syntax-fix.diff uploaded by elguero (license 37)
+ ........
+
+2010-03-04 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.1.18-rc1 released
+
+2010-03-03 21:25 +0000 [r250611] Leif Madsen <lmadsen@digium.com>
+
+ * /, doc/tex/localchannel.tex: Merged revisions 250609 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r250609 | lmadsen | 2010-03-03 16:22:55 -0500 (Wed, 03 Mar 2010)
+ | 11 lines Update existing Local channel documentation. A
+ complete re-write of the Local channel documentation has been
+ performed, with the existing information from localchannel.txt
+ and localchannel.tex merged in. (closes issue #16637) Reported
+ by: kobaz Patches: localchannel.tex uploaded by lmadsen (license
+ 10) localchannel.txt uploaded by lmadsen (license 10) Tested by:
+ lmadsen, jsmith, mmichelson ........
+
+2010-03-03 19:09 +0000 [r250483] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 250481 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r250481 | jpeeler | 2010-03-03 13:06:06 -0600
+ (Wed, 03 Mar 2010) | 22 lines Merged revisions 250480 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r250480 | jpeeler | 2010-03-03 13:04:11 -0600 (Wed, 03 Mar 2010)
+ | 15 lines Make sure to clear red alarm after polarity reversal.
+ From the issue: The automatic overnight line tests (or manual
+ ones) used on UK (BT) lines causes a red alarm on a dahdi /
+ TDM400P connected channel. This is because the line uses voltage
+ tests (battery loss) and polarity reversal. The polarity reversal
+ causes chan_dahdi to initiate v23 CallerID processing but during
+ this the event DAHDI_EVENT_NOALARM is ignored so that the alarm
+ is never cleared. (closes issue #14163) Reported by: jedi98
+ Patches: chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license
+ 653) Tested by: mattbrown, Chainsaw, mikeeccleston ........
+ ................
+
+2010-03-03 18:05 +0000 [r250260-250397] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 250395 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r250395 | dvossel | 2010-03-03 12:03:19 -0600
+ (Wed, 03 Mar 2010) | 22 lines Merged revisions 250394 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r250394 | dvossel | 2010-03-03 12:02:27 -0600 (Wed, 03 Mar 2010)
+ | 16 lines fixes problem with duplicate TXREQ packets When
+ Asterisk receives an IAX2 TXREQ packet, try_transfer() will call
+ store_by_transfercallno() to link the chan_iax2_pvt struct into
+ iax_transfercallno_pvts. If a duplicate TXREQ packet is received
+ for the same call, the pvt struct will be linked into
+ iax_transfercallno_pvts multiple times. This patch fixes this.
+ Thanks rain for debugging this and providing a patch! (closes
+ issue #16904) Reported by: rain Patches:
+ iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested
+ by: rain, dvossel ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 250246 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r250246 |
+ dvossel | 2010-03-02 18:18:28 -0600 (Tue, 02 Mar 2010) | 2 lines
+ fixes signed to unsigned int comparision issue for FaxMaxDatagram
+ value. ........
+
+2010-03-02 21:11 +0000 [r250039-250053] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/imapstorage.tex, /: Merged revisions 250051 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r250051 | lmadsen | 2010-03-02 16:09:27 -0500 (Tue, 02 Mar 2010)
+ | 8 lines Update IMAP documentation. Update the IMAP
+ documentation to make it clear that storing voicemails in the
+ same folder as a large number of emails could potentially cause
+ significant slow downs when writing or retrieving voicemails.
+ (issue #16704) Reported by: TimeHider Tested by: lmadsen,
+ TimeHider ........
+
+ * configs/cdr.conf.sample: Merged revisions 250045 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r250045 | lmadsen | 2010-03-02 15:52:19 -0500
+ (Tue, 02 Mar 2010) | 15 lines Merged revisions 250043 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02 Mar 2010)
+ | 7 lines Update documentation to clarify purpose of unanswered
+ option. (closes issue #16267) Reported by: elsto Patches:
+ cdr.conf.sample.patch.txt uploaded by lmadsen (license 10) Tested
+ by: davidw, elsto ........ ................
+
+ * doc/tex/configuration.tex, /: Merged revisions 250037 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r250037 | lmadsen | 2010-03-02 15:36:10 -0500 (Tue, 02
+ Mar 2010) | 4 lines Update documentation to not imply we support
+ overriding options. (closes issue #16855) Reported by: davidw
+ ........
+
+2010-03-02 19:47 +0000 [r249949] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/editline/makelist.in, apps/app_echo.c, UPGRADE.txt: revert
+ ability to exit echo app caused a regression, as only supported
+ VOICE, not VIDEO etc. (issue #16880)
+
+2010-03-02 19:15 +0000 [r249896] David Vossel <dvossel@digium.com>
+
+ * channels/chan_oss.c, channels/misdn_config.c,
+ include/asterisk/abstract_jb.h, configs/alsa.conf.sample,
+ channels/chan_jingle.c, channels/chan_usbradio.c,
+ channels/chan_dahdi.c, channels/chan_skinny.c,
+ configs/mgcp.conf.sample, main/abstract_jb.c,
+ channels/chan_h323.c, channels/chan_alsa.c,
+ configs/sip.conf.sample, channels/chan_mgcp.c,
+ channels/chan_unistim.c, configs/console.conf.sample,
+ configs/chan_dahdi.conf.sample, channels/chan_local.c,
+ configs/oss.conf.sample, channels/chan_sip.c, /,
+ configs/usbradio.conf.sample, configs/misdn.conf.sample,
+ channels/chan_gtalk.c, channels/chan_console.c: Merged revisions
+ 249893 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r249893 |
+ dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines
+ fixes adaptive jitterbuffer configuration When configuring the
+ adaptive jitterbuffer, the target_extra value not only could not
+ be set from the configuration, but was not even being set to its
+ proper default. This value is required in order for the adaptive
+ jitterbuffer to work correctly. To resolve this a config option
+ has been added to expose this value to the conf files, and a
+ default value is provided when no config specific value is
+ present. ........
+
+2010-03-02 09:05 +0000 [r249843] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * apps/app_echo.c: fixes ability to exit echo app when called from
+ a ISDN channel, null frames prevent '#' exit. Now only echo back
+ VOICE and DTMF frames (issue #16880) Reported by: alecdavis
+ Patches: echo_exit_1-6-1.diff.txt uploaded by alecdavis (license
+ 585) Tested by: alecdavis
+
+2010-03-01 19:39 +0000 [r249674] Sean Bright <sean@malleable.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 249672 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r249672 | seanbright | 2010-03-01 14:36:30 -0500
+ (Mon, 01 Mar 2010) | 18 lines Merged revisions 249671 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r249671 | seanbright | 2010-03-01 14:35:01 -0500 (Mon, 01 Mar
+ 2010) | 11 lines Fix crash in app_voicemail related to message
+ counting. We were passing a 'struct inprocess **' and treating it
+ like a 'struct inprocess *' causing a segfault. (closes issue
+ #16921) Reported by: whardier Patches: 20100301_issue16921.patch
+ uploaded by seanbright (license 71) Tested by: whardier ........
+ ................
+
+2010-03-01 18:39 +0000 [r249624] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 249623 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r249623 | tilghman | 2010-03-01 12:36:06 -0600 (Mon, 01 Mar 2010)
+ | 2 lines Constify a bit of app_voicemail, to make ODBC and IMAP
+ compile once again. ........
+
+2010-03-01 17:19 +0000 [r249548] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 249538 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r249538 | jpeeler | 2010-03-01 11:11:31 -0600
+ (Mon, 01 Mar 2010) | 18 lines Merged revisions 249536 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r249536 | jpeeler | 2010-03-01 11:02:03 -0600 (Mon, 01 Mar 2010)
+ | 11 lines Modify queued frames from local channels to not set
+ the other side to up In this case, attended transfers were broken
+ due to ast_feature_request_and_dial detecting the channel being
+ set to up before the answer frame could be read and therefore
+ failing to mark the channel as ready. This fix is a regression
+ fix for 244785, which should continue to work properly as well.
+ (closes issue #16816) Reported by: jamhed Tested by: jamhed,
+ corruptor ........ ................
+
+2010-02-28 20:51 +0000 [r249406-249492] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 249491 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r249491 | tilghman | 2010-02-28 14:50:01 -0600 (Sun, 28 Feb 2010)
+ | 5 lines Fix unit test that Alec Davis broke. (closes issue
+ #16927) Reported by: alecdavis ........
+
+ * apps/app_voicemail.c, include/asterisk/app.h, /: Merged revisions
+ 249405 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r249405 |
+ tilghman | 2010-02-28 01:10:22 -0600 (Sun, 28 Feb 2010) | 2 lines
+ Properly document voicemail API documents. Also fix a crash
+ reported via the -dev list. ........
+
+2010-02-27 23:37 +0000 [r249363] 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: overlap_receiving_trunk.diff.txt uploaded
+ by alecdavis (license 585) Tested by: alecdavis
+
+2010-02-27 14:09 +0000 [r249237] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 249235 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r249235 | kpfleming | 2010-02-27 09:08:35 -0500
+ (Sat, 27 Feb 2010) | 9 lines Merged revisions 249234 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r249234 | kpfleming | 2010-02-27 09:07:59 -0500 (Sat, 27
+ Feb 2010) | 1 line add a reference to the now-published IAX2 RFC
+ ........ ................
+
+2010-02-26 18:48 +0000 [r249189] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 249187 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r249187 | tilghman | 2010-02-26 12:41:57 -0600 (Fri, 26 Feb 2010)
+ | 18 lines Cleanups to fix bugs in the VM count API functions. -
+ Urgent voicemails were not attached, because the attachment code
+ looked in the wrong folder. - Urgent voicemails were sometimes
+ counted twice when displaying the count of new messages. -
+ Backends were inconsistent as to which voicemails each API
+ counted. (closes issue #15654) Reported by: tomo1657 Patches:
+ 20100225__issue15654.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman (closes issue #16448) Reported by: hevad
+ Review: https://reviewboard.asterisk.org/r/525/ ........
+
+2010-02-26 17:05 +0000 [r249103] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 249101 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r249101 | mmichelson | 2010-02-26 11:04:58 -0600 (Fri, 26 Feb
+ 2010) | 14 lines Merged revisions 249100 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r249100 | mmichelson | 2010-02-26 11:04:29 -0600 (Fri, 26 Feb
+ 2010) | 8 lines For T.38 reINVITEs treat a 606 the same as a 488.
+ (closes issue #16792) Reported by: vrban Patches: t38_606.patch
+ uploaded by vrban (license 756) ........ ................
+
+2010-02-25 23:11 +0000 [r248954] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 248952 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r248952 | jpeeler | 2010-02-25 17:09:54 -0600 (Thu, 25 Feb 2010)
+ | 24 lines Merged revisions 248860 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r248860 | jpeeler | 2010-02-25 15:22:06 -0600 (Thu, 25 Feb 2010)
+ | 18 lines Ensure that monitor recordings are written to the
+ correct location (again) This is an extension to 248757. As such
+ the dialplan test has been extended: exten => 5040, 1,
+ monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
+ dial(sip/5001) exten => 5041, 1,
+ monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
+ dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
+ exten => 5042, n, dial(sip/5001) exten => 5043, 1,
+ monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n,
+ changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001)
+ exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n,
+ changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by
+ design and emits a warning exten => 5044, n, dial(sip/5001)
+ ........ ................
+
+2010-02-25 22:42 +0000 [r248948] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/acl.c: Merged revisions 248946 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r248946 |
+ mmichelson | 2010-02-25 16:41:48 -0600 (Thu, 25 Feb 2010) | 5
+ lines Fix incorrect ACL behavior when CIDR notation of "/0" is
+ used. AST-2010-003 ........
+
+2010-02-25 21:24 +0000 [r248863] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 248861 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r248861 | tilghman | 2010-02-25 15:22:39 -0600 (Thu, 25 Feb 2010)
+ | 22 lines Merged revisions 248859 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r248859 | tilghman | 2010-02-25 15:21:05 -0600 (Thu, 25 Feb 2010)
+ | 15 lines Some platforms clear /var/run at boot, which makes
+ connecting a remote console... difficult. Previously, we only
+ created the default /var/run/asterisk directory at install time.
+ While we could create it in the init script, that would not work
+ for those who start asterisk manually from the command line. So
+ the safest thing to do is to create it as part of the Asterisk
+ boot process. This also changes the ownership of the directory,
+ because the pid and ctl files are created after we setuid/setgid.
+ (closes issue #16802) Reported by: Brian Patches:
+ 20100224__issue16802.diff.txt uploaded by tilghman (license 14)
+ Tested by: tzafrir ........ ................
+
+2010-02-25 18:52 +0000 [r248796] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 248793 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r248793 | jpeeler | 2010-02-25 12:37:56 -0600 (Thu, 25 Feb 2010)
+ | 22 lines Merged revisions 248757 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010)
+ | 15 lines Ensure that monitor recordings are written to the
+ correct location. Recordings should be placed in the monitor
+ directory when a non-absolute path is used. Exact dialplan used
+ for testing: exten => 5040, 1,
+ monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
+ dial(sip/5001) exten => 5041, 1,
+ monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
+ dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
+ exten => 5042, n, dial(sip/5001) ABE-2101 ........
+ ................
+
+2010-02-24 21:29 +0000 [r248641] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/logger.c: Merged revisions 248584 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r248584 | tilghman | 2010-02-24 15:17:26 -0600 (Wed, 24 Feb 2010)
+ | 14 lines Merged revisions 248582 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r248582 | tilghman | 2010-02-24 15:02:18 -0600 (Wed, 24 Feb 2010)
+ | 7 lines Remove color code sequences from verbose messages that
+ go to logfiles. (closes issue #16786) Reported by: dodo Patches:
+ logger2.patch uploaded by dodo (license 989) Tested by: tilghman
+ ........ ................
+
+2010-02-23 16:48 +0000 [r248399] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 248397 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r248397 | dvossel | 2010-02-23 10:34:39 -0600 (Tue, 23 Feb 2010)
+ | 15 lines Merged revisions 248396 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010)
+ | 9 lines fixes invite with replaces deadlock (closes issue
+ #16862) Reported by: pwalker Patches: replaces_deadlock_1.4
+ uploaded by dvossel (license 671) Tested by: pwalker, dvossel
+ ........ ................
+
+2010-02-19 19:05 +0000 [r248009] Tilghman Lesher <tlesher@digium.com>
+
+ * main/loader.c, /, channels/chan_console.c: Merged revisions
+ 228798 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk (closes issue
+ #16470) Reported by: kjotte ........ r228798 | tilghman |
+ 2009-11-09 01:37:52 -0600 (Mon, 09 Nov 2009) | 14 lines Fix
+ various problems detected with Valgrind. * chan_console accessed
+ pvts after deallocation. * The module loader did not check
+ usecount on shutdown, which led to chan_iax2 reading a timer that
+ was already unloaded. (closes issue #16062) Reported by:
+ alexanderheinz Patches: 20091109__issue16062.diff.txt uploaded by
+ tilghman (license 14) Tested by: tilghman ........
+
+2010-02-19 18:22 +0000 [r247946] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 247914 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r247914 | rmudgett | 2010-02-19 11:33:33 -0600
+ (Fri, 19 Feb 2010) | 62 lines Merged revisions 247910 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600
+ (Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
+ .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri,
+ 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing
+ consistent with other channel technologies. The processing of
+ DTMF tones on the receiving side of an ISDN channel is
+ inconsistent with the way it is handled in other channels,
+ especially DAHDI analog. This causes DTMF tones sent from an ISDN
+ phone to be doubled at the connected party. We are using the
+ following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes
+ Option one is necessary because the asterisk DSP DTMF detection
+ is better than mISDN's internal DSP. Not as many false positives.
+ Option two is necessary to transmit DTMF tones end to end when
+ mISDN channels are connected to SIP channels with out of band
+ DTMF for example. The symptom is that DTMF tones sent by an ISDN
+ phone are doubled on the way through asterisk when two mISDN
+ channels are connected with a Local channel in between or if it
+ is bridged to an analog channel. The doubling of DTMF tones is
+ because DTMF is passed inband to asterisk by the mISDN channel
+ and passed out of band once again after the release of the DTMF
+ tone. Passing it inband is wrong. Neither an analog channel nor
+ SIP channel passes DTMF inband if configured to inband DTMF.
+ Analog and SIP channels filter out the DTMF tones because they
+ use the voice frames returned by ast_dsp_process. But chan_misdn
+ passes the unfiltered input voice frames instead. To overcome one
+ aspect of the problem, the doubling of DTMF tones when two mISDN
+ channels are directly bridged, someone made an 'optimization',
+ where in that case the DTMF tone passed out-of-band to the peer
+ channel is not translated to an inband tone at the transmit side.
+ This optimization is bad because it does not work in general. For
+ example, analog channels or mISDN channels when bridged through
+ an intermediary local channel will generate DTMF tones from
+ out-of-band information. Also, of course, it must not be done
+ when there is no inband DTMF available. This patch fixes the
+ issue. Now chan_misdn will filter the received inband DTMF signal
+ the same as other channel types. Another change included: No need
+ to build an extra translation path because ast_process_dsp does
+ it if required. Patches: misdn-dtmf.patch JIRA ABE-2080
+ ................ ................
+
+2010-02-18 23:15 +0000 [r247790-247843] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_speech.c, /: Merged revisions 247841 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r247841 |
+ tilghman | 2010-02-18 17:13:46 -0600 (Thu, 18 Feb 2010) | 7 lines
+ Revert an errant part of a previous cleanup, to fix a memory
+ corruption issue. (closes issue #16368) Reported by: thirionjwf
+ Patches: res_speech.c.patch uploaded by thirionjwf (license 955)
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 247787 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r247787 |
+ tilghman | 2010-02-18 15:42:53 -0600 (Thu, 18 Feb 2010) | 17
+ lines If the peer record is from realtime, it could be set to 0,
+ due to MySQL not representing NULL well in integer columns. NULL
+ means the value is not specified for the column, which normally
+ means the driver uses whatever is the default value. However, on
+ MySQL, placing a NULL in either a float or integer column results
+ in a retrieval of the 0 value. Hence, users get an errant error
+ on load. This patch suppresses that error and makes the value as
+ if it was not there. Note that this cannot be done in the
+ realtime driver, because the lack of difference between NULL and
+ 0 can only be intepreted correctly by the driver itself. If we
+ did it in the realtime driver, then it would be effectively
+ impossible to set any realtime field to 0, because it would act
+ as if the field were unspecified and possibly take on a different
+ value. (closes issue #16683) Reported by: wdoekes ........
+
+2010-02-18 19:45 +0000 [r247654] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/features.c: Merged revisions 247652 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r247652 | mnicholson | 2010-02-18 13:39:37 -0600 (Thu, 18 Feb
+ 2010) | 13 lines Merged revisions 247651 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r247651 | mnicholson | 2010-02-18 13:38:09 -0600 (Thu, 18 Feb
+ 2010) | 6 lines Copy the calling party's account code to the
+ called party if they don't already have one. (closes issue
+ #16331) Reported by: bluefox Tested by: mnicholson ........
+ ................
+
+2010-02-18 16:57 +0000 [r247505-247511] Leif Madsen <lmadsen@digium.com>
+
+ * README-SERIOUSLY.bestpractices.txt: Merged revisions 247509 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r247509 | lmadsen | 2010-02-18 11:54:43 -0500
+ (Thu, 18 Feb 2010) | 9 lines Merged revisions 247508 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r247508 | lmadsen | 2010-02-18 11:53:44 -0500 (Thu, 18
+ Feb 2010) | 1 line Add additional link to best practices document
+ per jsmith. ........ ................
+
+ * README-SERIOUSLY.bestpractices.txt (added): Merged revisions
+ 247503 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r247503 | lmadsen | 2010-02-18 11:41:04 -0500 (Thu, 18 Feb 2010)
+ | 18 lines Merged revisions 247502 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r247502 | lmadsen | 2010-02-18 11:38:17 -0500 (Thu, 18 Feb 2010)
+ | 10 lines Add best practices documentation. (issue #16808)
+ Reported by: lmadsen (issue #16810) Reported by: Nick_Lewis
+ Tested by: lmadsen Review:
+ https://reviewboard.asterisk.org/r/507/ ........ ................
+
+2010-02-18 04:21 +0000 [r247425] Russell Bryant <russell@digium.com>
+
+ * sounds/Makefile, Makefile, /: Merged revisions 247423 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r247423 | russell | 2010-02-17 22:20:11 -0600
+ (Wed, 17 Feb 2010) | 17 lines Merged revisions 247422 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r247422 | russell | 2010-02-17 22:19:01 -0600 (Wed, 17 Feb 2010)
+ | 10 lines Tweak argument handling for wget in the sounds
+ Makefile. 1) Fix the check to see if we are using wget to not be
+ full of fail. The configure script populates this variable with
+ the absolute path to wget if it is found, so it didn't work. 2)
+ Allow some extra arguments to be passed in for wget. This is just
+ a simple change to allow our Bamboo build script to tell wget to
+ be quiet and not fill up our logs with download status output.
+ ........ ................
+
+2010-02-17 21:28 +0000 [r246987-247336] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/utils.c, include/asterisk/strings.h: Merged revisions
+ 247335 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r247335 |
+ mmichelson | 2010-02-17 15:22:40 -0600 (Wed, 17 Feb 2010) | 20
+ lines Fix two problems in ast_str functions found while writing a
+ unit test. 1. The documentation for ast_str_set and
+ ast_str_append state that the max_len parameter may be -1 in
+ order to limit the size of the ast_str to its current allocated
+ size. The problem was that the max_len parameter in all cases was
+ a size_t, which is unsigned. Thus a -1 was interpreted as
+ UINT_MAX instead of -1. Changing the max_len parameter to be
+ ssize_t fixed this issue. 2. Once issue 1 was fixed, there was an
+ off-by-one error in the case where we attempted to write a string
+ larger than the current allotted size to a string when -1 was
+ passed as the max_len parameter. When trying to write more than
+ the allotted size, the ast_str's __AST_STR_USED was set to 1
+ higher than it should have been. Thanks to Tilghman for quickly
+ spotting the offending line of code. Oh, and the unit test that I
+ referenced in the top line of this commit will be added to
+ reviewboard shortly. Sit tight... ........
+
+ * apps/app_queue.c, /: Merged revisions 247169 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r247169 | mmichelson | 2010-02-17 10:24:54 -0600 (Wed, 17 Feb
+ 2010) | 9 lines Merged revisions 247168 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r247168 | mmichelson | 2010-02-17 10:24:17 -0600 (Wed, 17 Feb
+ 2010) | 3 lines Make sure that when autofill is disabled that
+ callers not in the front of the queue cannot place calls.
+ ........ ................
+
+ * /, main/utils.c: Merged revisions 247076 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r247076 |
+ mmichelson | 2010-02-16 17:44:33 -0600 (Tue, 16 Feb 2010) | 12
+ lines Add va_end calls to __ast_str_helper. According to the man
+ page for stdarg(3), "Each invocation of va_copy() must be matched
+ by a corresponding invocation of va_end() in the same function."
+ There were several cases in __ast_str_helper where va_copy was
+ not matched with a corresponding call to va_end. ........
+
+ * /, include/asterisk/strings.h: Merged revisions 246985 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r246985 | mmichelson | 2010-02-16 15:15:38 -0600 (Tue,
+ 16 Feb 2010) | 3 lines Add some clarifying documentation to the
+ ast_str_set and ast_str_append functions. ........
+
+2010-02-16 21:05 +0000 [r246902-246983] David Vossel <dvossel@digium.com>
+
+ * main/tcptls.c, /: Merged revisions 246980 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r246980 |
+ dvossel | 2010-02-16 14:54:48 -0600 (Tue, 16 Feb 2010) | 8 lines
+ warning message if openssl support is missing while attempting
+ tls connection (closes issue #16673) Reported by: michaesc
+ Patches: tls_error_msg.diff uploaded by dvossel (license 671)
+ ........
+
+ * main/channel.c, /: Merged revisions 246899 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r246899 |
+ dvossel | 2010-02-16 11:07:41 -0600 (Tue, 16 Feb 2010) | 16 lines
+ fixes sample rate conversion issue with Monitor application When
+ using ast_seekstream with the read/write streams of a monitor,
+ the number of samples we are seeking must be of the same rate as
+ the stream or the jump calculation will be incorrect. This patch
+ adds logic to correctly convert the number of samples to jump to
+ the sample rate the read/write stream is using. For example, if
+ the call is G722 (16khz) and the read/write stream is recording a
+ 8khz wav, seeking 320 samples of 16khz audio is not the same as
+ seeking 320 samples of 8khz audio when performing the
+ ast_seekstream on the stream. ABE-2044 ........
+
+2010-02-15 23:44 +0000 [r246712] Tilghman Lesher <tlesher@digium.com>
+
+ * Makefile, /: Merged revisions 246710 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r246710 | tilghman | 2010-02-15 17:43:28 -0600 (Mon, 15 Feb 2010)
+ | 12 lines Merged revisions 246709 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r246709 | tilghman | 2010-02-15 17:42:33 -0600 (Mon, 15 Feb 2010)
+ | 5 lines Make the menuselect instructions correct by allowing
+ 'make menuselect' to actually solve dependency problems.
+ (Previously, it would fail out again with the same message about
+ running 'make menuselect', which was NOT at all helpful.)
+ ........ ................
+
+2010-02-12 23:34 +0000 [r246548] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /: Merged revisions 246546 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r246546 | dvossel | 2010-02-12 17:32:33 -0600 (Fri, 12 Feb 2010)
+ | 21 lines Merged revisions 246545 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r246545 | dvossel | 2010-02-12 17:30:17 -0600 (Fri, 12 Feb 2010)
+ | 16 lines lock channel during datastore removal On channel
+ destruction the channel's datastores are removed and destroyed.
+ Since there are public API calls to find and remove datastores on
+ a channel, a lock should be held whenever datastores are removed
+ and destroyed. This resolves a crash caused by a race condition
+ in app_chanspy.c. (closes issue #16678) Reported by:
+ tim_ringenbach Patches: datastore_destroy_race.diff uploaded by
+ tim ringenbach (license 540) Tested by: dvossel ........
+ ................
+
+2010-02-12 19:08 +0000 [r246463] Jason Parker <jparker@digium.com>
+
+ * main/channel.c: Fix some silly formatting that made my head hurt.
+
+2010-02-10 21:28 +0000 [r246202-246206] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_strings.c: Merged revisions 246204 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r246204 | tilghman | 2010-02-10 15:24:10 -0600 (Wed, 10 Feb 2010)
+ | 2 lines Fussy compiler on another machine... ........
+
+ * /, funcs/func_strings.c: Merged revisions 246200 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r246200 | tilghman | 2010-02-10 15:19:35 -0600 (Wed, 10 Feb 2010)
+ | 2 lines Fix weird issue with unit tests on optimized build -
+ turned out to be a signing issue. ........
+
+2010-02-10 17:52 +0000 [r246119] David Vossel <dvossel@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 246116 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r246116 | dvossel | 2010-02-10 11:49:34 -0600 (Wed, 10 Feb 2010)
+ | 14 lines Merged revisions 246115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r246115 | dvossel | 2010-02-10 11:44:20 -0600 (Wed, 10 Feb 2010)
+ | 8 lines fixes random deadlock in app_queue with use_weight
+ during reload (closes issue #16677) Reported by: tim_ringenbach
+ Patches: app_queue_use_weight_deadlock.diff uploaded by tim
+ ringenbach (license 540) ........ ................
+
+2010-02-10 16:55 +0000 [r246072] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 246070 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r246070 | jpeeler | 2010-02-10 10:47:37 -0600 (Wed, 10 Feb 2010)
+ | 22 lines Change channel state on local channels for
+ busy,answer,ring. Previously local channels channel state never
+ changed. This became problematic when the state of the other side
+ of the local channel was lost, for example during a masquerade.
+ Changing the state of the local channel allows for the scenario
+ to be detected when the channel state is set to ringing, but the
+ peer isn't ringing. The specific problem scenario is described in
+ 164201. Although this was noted on one of the issues, here is the
+ tested dialplan verified to work: exten =>
+ 9700,1,Dial(Local/*9700@default&Local/0009700@default) exten =>
+ *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1)
+ exten => *9700,n,wait(3) ;3 works, 1 did not exten =>
+ *9700,n,Dial(SIP/5001) exten => 0009700,1,Wait(1) ;1 works, 3 did
+ not exten =>
+ 0009700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes
+ issue #14992) Reported by: davidw ........
+
+2010-02-10 15:38 +0000 [r245947-246024] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_strings.c: Merged revisions 246022 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r246022 | tilghman | 2010-02-10 09:36:57 -0600 (Wed, 10 Feb 2010)
+ | 2 lines Enable warnings on atypical conditions for the FILTER
+ function (suggested by mmichelson on the -dev list). ........
+
+ * configs/extensions.conf.sample, /, funcs/func_strings.c: Merged
+ revisions 245945 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r245945 | tilghman | 2010-02-10 08:06:12 -0600 (Wed, 10 Feb 2010)
+ | 9 lines Merged revisions 245944 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r245944 | tilghman | 2010-02-10 07:37:13 -0600 (Wed, 10 Feb 2010)
+ | 2 lines Include examples of FILTER usage in extension patterns
+ where a "." may be a risk. ........ ................
+
+2010-02-09 23:13 +0000 [r245795] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 245793 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r245793 | dvossel | 2010-02-09 17:07:17 -0600
+ (Tue, 09 Feb 2010) | 18 lines Merged revisions 245792 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r245792 | dvossel | 2010-02-09 16:55:38 -0600 (Tue, 09 Feb 2010)
+ | 12 lines Fixes iaxs and iaxsl size off by one issue. 2^15 =
+ 32768 which is the maximum allowed iax2 callnumber. Creating the
+ iaxs and iaxsl array of size 32768 means the maximum callnumber
+ is actually out of bounds. This causes a nasty crash. (closes
+ issue #15997) Reported by: exarv Patches: iax_fix.diff uploaded
+ by dvossel (license 671) ........ ................
+
+2010-02-09 18:09 +0000 [r245731] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 245729 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r245729 |
+ tilghman | 2010-02-09 12:06:30 -0600 (Tue, 09 Feb 2010) | 8 lines
+ Ensure frames are only freed once. (closes issue #16361) Reported
+ by: vlad Patches: 20100208__issue16361.diff.txt uploaded by
+ tilghman (license 14) Tested by: kenny, bloodoff, misaksen
+ ........
+
+2010-02-09 16:26 +0000 [r245682] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 245680 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r245680 |
+ kpfleming | 2010-02-09 10:24:52 -0600 (Tue, 09 Feb 2010) | 8
+ lines Don't offer MMR or JBIG transcoding during T.38
+ negotiation. After further discussion with Steve Underwood, we
+ should not (yet) be offering to receive MMR or JBIG transcoded
+ streams from T.38 endpoints. A future spandsp release will
+ support those features, and then they can be enabled during
+ negotiation ........
+
+2010-02-08 23:45 +0000 [r245625] Russell Bryant <russell@digium.com>
+
+ * /, main/event.c: Merged revisions 245624 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r245624 |
+ russell | 2010-02-08 17:43:00 -0600 (Mon, 08 Feb 2010) | 5 lines
+ Fix return value of get_ie_str() and get_ie_str_hash() for
+ non-existent IE. I found this bug while developing a unit test
+ for event allocation. Testing is awesome. ........
+
+2010-02-08 22:46 +0000 [r245580] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/Makefile, /, main/Makefile: Merged revisions 245578 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r245578 | tilghman | 2010-02-08 16:31:40 -0600 (Mon, 08
+ Feb 2010) | 12 lines Actually use _ASTLDFLAGS in the main/ and
+ channels/ Makefiles. They were previously passed correctly, but
+ they simply weren't used. This caused issues with various
+ platforms whose builds needed to pass special linker flags via
+ the configure script. (closes issue #16596) Reported by:
+ pprindeville Patches: asterisk-1.6-astldflags.patch uploaded by
+ pprindeville (license 347) Tested by: tilghman ........
+
+2010-02-08 20:42 +0000 [r245499] Jason Parker <jparker@digium.com>
+
+ * main/ast_expr2.fl, /, main/ast_expr2f.c: Merged revisions 245497
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r245497 | qwell | 2010-02-08 14:41:05 -0600
+ (Mon, 08 Feb 2010) | 11 lines Merged revisions 245496 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r245496 | qwell | 2010-02-08 14:39:50 -0600 (Mon, 08 Feb 2010) |
+ 4 lines Remove reference of documentation in source directory.
+ People don't always build Asterisk from source (distro packages,
+ anybody?). ........ ................
+
+2010-02-05 19:26 +0000 [r245095] Jeff Peeler <jpeeler@digium.com>
+
+ * contrib/firmware (removed), /, LICENSE: Merged revisions 245090
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r245090 | jpeeler | 2010-02-05 13:26:22 -0600
+ (Fri, 05 Feb 2010) | 11 lines Merged revisions 245044 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r245044 | kpfleming | 2010-02-05 12:32:29 -0600 (Fri, 05 Feb
+ 2010) | 5 lines Remove contrib/firmware directory as it is empty
+ Remove explicit license for IAXy firmware as it is no longer
+ included in the tree ........ ................
+
+2010-02-05 17:10 +0000 [r244929] Sean Bright <sean@malleable.com>
+
+ * main/asterisk.c, /: Merged revisions 244927 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r244927 | seanbright | 2010-02-05 12:05:32 -0500 (Fri, 05 Feb
+ 2010) | 9 lines Merged revisions 244926 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r244926 | seanbright | 2010-02-05 12:03:35 -0500 (Fri, 05 Feb
+ 2010) | 1 line Update main copyright date. ........
+ ................
+
+2010-02-03 19:27 +0000 [r244553] Mark Michelson <mmichelson@digium.com>
+
+ * main/sched.c, /: Merged revisions 244547 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r244547 |
+ mmichelson | 2010-02-03 13:26:53 -0600 (Wed, 03 Feb 2010) | 3
+ lines Initialize counters in ast_sched_report so that resulting
+ data is not bogus. ........
+
+2010-02-03 18:43 +0000 [r244507] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 244505 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r244505 | tilghman | 2010-02-03 12:34:29 -0600 (Wed, 03 Feb 2010)
+ | 8 lines The chanvar= setting should inherit the entire list of
+ variables, not just the first one. (closes issue #16359) Reported
+ by: raarts Patches: dahdi-setvars.diff uploaded by raarts
+ (license 937) Tested by: raarts ........
+
+2010-02-02 22:31 +0000 [r244446] David Vossel <dvossel@digium.com>
+
+ * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h:
+ Merged revisions 244443 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r244443 |
+ dvossel | 2010-02-02 16:27:23 -0600 (Tue, 02 Feb 2010) | 18 lines
+ fixes crash during T.38 negotiation caused by invalid or missing
+ FaxMaxDatagram field AST-2010-001 (closes issue #16634) Reported
+ by: krn (closes issue #16724) Reported by: barthpbx (closes issue
+ #16517) Reported by: bklang (closes issue #16485) Reported by:
+ elsto ........
+
+2010-02-02 20:35 +0000 [r244394] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 244393 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r244393 |
+ tilghman | 2010-02-02 14:32:29 -0600 (Tue, 02 Feb 2010) | 18
+ lines Properly respect GOSUB_RESULT as to what to do with the
+ master channel. Previously, we would parse GOSUB_RESULT, but not
+ actually do anything with it. (closes issue #16686) Reported by:
+ bklang Patches: app_dial-respect-gosub_result.patch uploaded by
+ bklang (license 919) (with modifications) ........
+
+2010-02-02 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.1.14
+
+ * AST-2010-001: An attacker attempting to negotiate T.38 over SIP can
+ remotely crash Asterisk by modifying the FaxMaxDatagram field of
+ the SDP to contain either a negative or exceptionally large value.
+ The same crash occurs when the FaxMaxDatagram field is omitted from
+ the SDP as well.
+
+2010-01-14 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.1.13
+
+2010-01-08 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.1.13-rc1
+
+2010-01-07 21:17 +0000 [r238497] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_oss.c, main/poll.c, channels/chan_usbradio.c,
+ include/asterisk/utils.h, /, channels/chan_sip.c,
+ channels/chan_alsa.c, channels/chan_console.c: Merged revisions
+ 209400 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209400 |
+ kpfleming | 2009-07-28 08:49:46 -0500 (Tue, 28 Jul 2009) | 3
+ lines Define side-effect-safe MIN and MAX macros and remove
+ duplicate definitions from various files. (closes issue #16251)
+ Reported by: asgaroth ........
+
+2010-01-07 20:20 +0000 [r238363-238430] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 238412 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r238412 | dvossel | 2010-01-07 14:15:27 -0600
+ (Thu, 07 Jan 2010) | 16 lines Merged revisions 238411 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07 Jan 2010)
+ | 10 lines fixes crash in "scheduled_destroy" in chan_iax A
+ signed short was used to represent a callnumber. This is makes it
+ possible to attempt to access the iaxs array with a negative
+ index. (closes issue #16565) Reported by: jensvb ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 238405 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r238405 |
+ dvossel | 2010-01-07 14:00:31 -0600 (Thu, 07 Jan 2010) | 8 lines
+ Change in sip show channels display format allowing more digits
+ for CID (closes issue #16459) Reported by: Rzadzins Patches:
+ chan_sip_longer_cid.patch uploaded by Rzadzins (license 953)
+ ........
+
+ * apps/app_queue.c, /: Merged revisions 238361 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r238361 |
+ dvossel | 2010-01-07 12:58:23 -0600 (Thu, 07 Jan 2010) | 8 lines
+ cli 'queue show' formatting fix. queue name was truncated over 12
+ characters (closes issue #16078) Reported by: RoadKill Patches:
+ quequename_limit.patch uploaded by ppyy (license 906) Tested by:
+ dvossel ........
+
+2010-01-07 12:09 +0000 [r238351] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * /, configs/sip.conf.sample: Merged revisions 238313 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r238313 | tzafrir | 2010-01-07 11:14:57 +0200 (ה', 07 ינו 2010) |
+ 2 lines Document the usefulness of explicit udp:// in the
+ register string ........
+
+2010-01-06 21:48 +0000 [r238233] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_cdr.c: Merged revisions 238231 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r238231 | tilghman | 2010-01-06 15:45:17 -0600 (Wed, 06 Jan 2010)
+ | 11 lines Merged revisions 238230 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r238230 | tilghman | 2010-01-06 15:41:55 -0600 (Wed, 06 Jan 2010)
+ | 4 lines Revise documentation on disposition values to the
+ actual values used. (closes issue #16289) Reported by: wdoekes
+ ........ ................
+
+2010-01-06 20:39 +0000 [r238136-238183] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_meetme.c: Merged revisions 238181 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r238181 |
+ jpeeler | 2010-01-06 14:37:18 -0600 (Wed, 06 Jan 2010) | 8 lines
+ Fix misreverting from 177158. (closes issue #15725) Reported by:
+ shanermn Patches: v1-15725.patch uploaded by dimas (license 88)
+ Tested by: shanermn ........
+
+ * /, main/features.c: Merged revisions 238134 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r238134 |
+ jpeeler | 2010-01-06 13:05:06 -0600 (Wed, 06 Jan 2010) | 10 lines
+ Fix channel name comparison for bridge application. The channel
+ name comparison was not comparing the whole string and therefore
+ if one channel name was a substring of the other, the bridge
+ would fail. (closes issue #16528) Reported by: telecos82 Patches:
+ res_features_r236843.diff uploaded by telecos82 (license 687)
+ ........
+
+2010-01-06 15:21 +0000 [r238012] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_mp3.c: Merged revisions 238010 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r238010 | russell | 2010-01-06 09:19:10 -0600 (Wed, 06 Jan 2010)
+ | 14 lines Merged revisions 238009 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r238009 | russell | 2010-01-06 09:18:22 -0600 (Wed, 06 Jan 2010)
+ | 7 lines Resolve a crash due to an ast_frame not being fully
+ initialized. (closes issue #16531) Reported by: john8675309
+ (closes SWP-615) ........ ................
+
+2010-01-06 06:52 +0000 [r237967] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: One duplicate setting here (dead code).
+
+2010-01-05 23:09 +0000 [r237841-237922] David Vossel <dvossel@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 237920 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r237920 |
+ dvossel | 2010-01-05 17:08:50 -0600 (Tue, 05 Jan 2010) | 16 lines
+ fixes holdtime playback issue in app_queue When reporting hold
+ time, the number of seconds should be mod 60. Otherwise audio
+ playback could be something like "2 minutes 123 seconds" rather
+ than "2 minutes 3 seconds". Also, the "minute" sound file is
+ missing, so for the moment until that file can be created the
+ "minutes" file is used instead. (closes issue #16168) Reported
+ by: nickilo Patches: patch-unified-trunk-rev-222176 uploaded by
+ nickilo (license ) Tested by: nickilo, wonderg ........
+
+ * main/pbx.c, /: Merged revisions 237839 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r237839 |
+ dvossel | 2010-01-05 13:29:47 -0600 (Tue, 05 Jan 2010) | 19 lines
+ fixes subscriptions being lost after 'module reload' During a
+ module reload if multiple extension configs are present, such as
+ both extensions.conf and extensions.ael, watchers for one
+ config's hints will be lost during the merging of the other
+ config. This happens because hint watchers are only preserved for
+ the current config being merged. The old context list is
+ destroyed after the merging takes place, meaning any watchers
+ that were not perserved will be removed. Now all hints are
+ preserved during merging regardless of what config file is being
+ merged. These hints are only restored if they are present within
+ the new context list. (closes issue #16093) Reported by: jlaroff
+ ........
+
+2010-01-05 17:22 +0000 [r237724] Russell Bryant <russell@digium.com>
+
+ * /, main/utils.c: Merged revisions 237699 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r237699 | russell | 2010-01-05 11:16:01 -0600 (Tue, 05 Jan 2010)
+ | 14 lines Merged revisions 237697 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237697 | russell | 2010-01-05 11:13:28 -0600 (Tue, 05 Jan 2010)
+ | 7 lines Change a NOTICE log message to DEBUG where it belongs.
+ (closes issue #16479) Reported by: alexrecarey (closes SWP-577)
+ ........ ................
+
+2010-01-04 21:52 +0000 [r237408-237576] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/say.c: Merged revisions 237574 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r237574 | tilghman | 2010-01-04 15:48:20 -0600 (Mon, 04 Jan 2010)
+ | 13 lines Merged revisions 237573 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237573 | tilghman | 2010-01-04 15:45:46 -0600 (Mon, 04 Jan 2010)
+ | 6 lines Bounds checking for input string (closes issue #16407)
+ Reported by: qwell Patches: 20100104__issue16407.diff.txt
+ uploaded by tilghman (license 14) ........ ................
+
+ * main/pbx.c, /: Merged revisions 237494 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r237494 | tilghman | 2010-01-04 14:59:01 -0600 (Mon, 04 Jan 2010)
+ | 15 lines Merged revisions 237493 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237493 | tilghman | 2010-01-04 14:57:35 -0600 (Mon, 04 Jan 2010)
+ | 8 lines Regression in issue #15421 - Pattern matching (closes
+ issue #16482) Reported by: wdoekes Patches:
+ astsvn-16482-betterfix.diff uploaded by wdoekes (license 717)
+ 20091223__issue16482.diff.txt uploaded by tilghman (license 14)
+ Tested by: wdoekes, tilghman ........ ................
+
+ * main/config.c, /: Merged revisions 237414 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r237414 |
+ tilghman | 2010-01-04 13:03:20 -0600 (Mon, 04 Jan 2010) | 2 lines
+ Oops, didn't compile (thanks, kpfleming) ........
+
+ * main/config.c, /: Merged revisions 237410 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r237410 |
+ tilghman | 2010-01-04 12:42:10 -0600 (Mon, 04 Jan 2010) | 7 lines
+ Further reduce the encoded blank values back to blank in the
+ realtime API. (closes issue #16533) Reported by: sergee Patches:
+ 200100104__issue16533.diff.txt uploaded by tilghman (license 14)
+ Tested by: sergee ........
+
+ * main/pbx.c, /, res/res_agi.c, include/asterisk/channel.h: Merged
+ revisions 237406 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r237406 | tilghman | 2010-01-04 12:28:28 -0600 (Mon, 04 Jan 2010)
+ | 23 lines Merged revisions 237405 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010)
+ | 16 lines Add a flag to disable the Background behavior, for AGI
+ users. This is in a section of code that relates to two other
+ issues, namely issue #14011 and issue #14940), one of which was
+ the behavior of Background when called with a context argument
+ that matched the current context. This fix broke FreePBX,
+ however, in a post-Dial situation. Needless to say, this is an
+ extremely difficult collision of several different issues. While
+ the use of an exception flag is ugly, fixing all of the issues
+ linked is rather difficult (although if someone would like to
+ propose a better solution, we're happy to entertain that
+ suggestion). (closes issue #16434) Reported by: rickead2000
+ Patches: 20091217__issue16434.diff.txt uploaded by tilghman
+ (license 14) 20091222__issue16434__1.6.1.diff.txt uploaded by
+ tilghman (license 14) Tested by: rickead2000 ........
+ ................
+
+2010-01-04 16:51 +0000 [r237329] David Vossel <dvossel@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 237327 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r237327 |
+ dvossel | 2010-01-04 10:39:11 -0600 (Mon, 04 Jan 2010) | 10 lines
+ app_queue segfaults if realtime field uniqueid is NULL (closes
+ issue #16385) Reported by: haakon Patches: app_queue.c.patch
+ uploaded by haakon (license 880) app_queue.c.patch_v2 uploaded by
+ dvossel (license 671) Tested by: haakon ........
+
+2010-01-04 16:26 +0000 [r237325] Jeff Peeler <jpeeler@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 237323 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r237323 |
+ jpeeler | 2010-01-04 10:24:51 -0600 (Mon, 04 Jan 2010) | 5 lines
+ Fix timeout for AGI command speech recognize. (closes issue
+ #16297) Reported by: semond ........
+
+2010-01-04 16:21 +0000 [r237321] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 237319 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r237319 | tilghman | 2010-01-04 10:20:03 -0600
+ (Mon, 04 Jan 2010) | 10 lines Merged revisions 237318 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237318 | tilghman | 2010-01-04 10:18:59 -0600 (Mon, 04 Jan 2010)
+ | 3 lines It's also possible for the Local channel to directly
+ execute an Application. Reviewboard:
+ https://reviewboard.asterisk.org/r/452/ ........ ................
+
+2010-01-02 10:01 +0000 [r237138] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 237136 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r237136 | oej | 2010-01-02 10:54:22 +0100 (Lör, 02 Jan 2010) | 10
+ lines Merged revisions 237135 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r237135 | oej | 2010-01-02 10:52:30 +0100 (Lör, 02 Jan 2010) | 2
+ lines Release memory of the contact acl before unloading module
+ ........ ................
+
+2009-12-30 22:00 +0000 [r236984] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 236982 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r236982 | tilghman | 2009-12-30 15:59:18 -0600
+ (Wed, 30 Dec 2009) | 16 lines Merged revisions 236981 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236981 | tilghman | 2009-12-30 15:57:10 -0600 (Wed, 30 Dec 2009)
+ | 9 lines Don't queue frames to channels that have no means to
+ process them. (closes issue #15609) Reported by: aragon Patches:
+ 20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by
+ tilghman (license 14) Tested by: aragon Review:
+ https://reviewboard.asterisk.org/r/452/ ........ ................
+
+2009-12-30 21:13 +0000 [r236904] Jeff Peeler <jpeeler@digium.com>
+
+ * /, utils/ael_main.c: Merged revisions 236902 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r236902 |
+ jpeeler | 2009-12-30 15:09:28 -0600 (Wed, 30 Dec 2009) | 2 lines
+ One more LOW_MEMORY compile fix. ........
+
+2009-12-30 17:57 +0000 [r236803-236851] Tilghman Lesher <tlesher@digium.com>
+
+ * /, cdr/cdr_adaptive_odbc.c: Merged revisions 236847 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r236847 | tilghman | 2009-12-30 11:53:29 -0600 (Wed, 30 Dec 2009)
+ | 4 lines When the field is blank, don't warn about the field
+ being unable to be coerced, just skip the column. (closes
+ http://lists.digium.com/pipermail/asterisk-dev/2009-December/041362.html)
+ Reported by Nic Colledge on the -dev list, fixed by me. ........
+
+ * /, channels/chan_sip.c: Merged revisions 236802 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r236802 |
+ tilghman | 2009-12-29 17:05:45 -0600 (Tue, 29 Dec 2009) | 7 lines
+ Shut down the SIP session timers more gracefully, in order to
+ prevent a possible crash. (closes issue #16452) Reported by:
+ corruptor Patches: 20091221__issue16452.diff.txt uploaded by
+ tilghman (license 14) Tested by: corruptor ........
+
+2009-12-28 22:11 +0000 [r236715] Jason Parker <jparker@digium.com>
+
+ * main/ast_expr2.c, /, main/ast_expr2.y: Merged revisions 236713
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r236713 | qwell | 2009-12-28 16:09:40 -0600 (Mon, 28 Dec
+ 2009) | 8 lines Allow "REMAINDER" to function properly in
+ expressions. (closes issue #16427) Reported by: wdoekes Patches:
+ ast16-reminder-remainder.patch uploaded by wdoekes (license 717)
+ Tested by: wdoekes ........
+
+2009-12-28 17:40 +0000 [r236669] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 236667 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r236667 | tilghman | 2009-12-28 11:37:46 -0600 (Mon, 28 Dec 2009)
+ | 4 lines Use recommended option, not deprecated option. (closes
+ issue #16515) Reported by: ManChicken ........
+
+2009-12-28 15:31 +0000 [r236512-236634] Sean Bright <sean@malleable.com>
+
+ * include/asterisk/threadstorage.h, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 236613 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r236613 | seanbright | 2009-12-28 10:22:54 -0500 (Mon, 28 Dec
+ 2009) | 14 lines Merged revisions 236585 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236585 | seanbright | 2009-12-28 10:12:08 -0500 (Mon, 28 Dec
+ 2009) | 7 lines Try a test compile to see if PTHREAD_ONCE_INIT
+ requires extra braces. There was conditional code (based on build
+ platform) to optioinally wrap PTHREAD_ONCE_INIT in braces that
+ was removed since it is fixed in newer versions of
+ Solaris/OpenSolaris, but I am still running into it on Solaris 10
+ x86 so add a configure-time check for it. ........
+ ................
+
+ * /, apps/app_meetme.c: Merged revisions 236510 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r236510 | seanbright | 2009-12-28 07:44:58 -0500 (Mon, 28 Dec
+ 2009) | 19 lines Merged revisions 236509 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec
+ 2009) | 12 lines Avoid a crash with large numbers of MeetMe
+ conferences. Similar to changes made to Queue(), when we have
+ large numbers of conferences in meetme.conf (1000s) and we use
+ alloca()/strdupa(), we can blow out the stack and crash, so
+ instead just use a single fixed buffer. (closes issue #16509)
+ Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded
+ by seanbright (license 71) Tested by: seanbright ........
+ ................
+
+2009-12-27 18:22 +0000 [r236436] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 236434 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r236434 | tilghman | 2009-12-27 12:20:53 -0600
+ (Sun, 27 Dec 2009) | 9 lines Merged revisions 236433 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r236433 | tilghman | 2009-12-27 12:19:38 -0600 (Sun, 27
+ Dec 2009) | 2 lines Turn on colors in the daemon, since there's
+ many requests for it on Ubuntu. ........ ................
+
+2009-12-26 15:31 +0000 [r236360] Kevin P. Fleming <kpfleming@digium.com>
+
+ * sounds/Makefile, /: Merged revisions 236358 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r236358 | kpfleming | 2009-12-26 09:27:44 -0600 (Sat, 26 Dec
+ 2009) | 9 lines Merged revisions 236357 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236357 | kpfleming | 2009-12-26 09:26:17 -0600 (Sat, 26 Dec
+ 2009) | 1 line update to latest releases with zero uid/gid
+ ........ ................
+
+2009-12-23 18:26 +0000 [r236188-236302] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_stack.c, /: Merged revisions 236300 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r236300 |
+ tilghman | 2009-12-23 12:25:27 -0600 (Wed, 23 Dec 2009) | 7 lines
+ AGI may be invoked from outside the dialplan (closes issue
+ #16510) Reported by: atis Patches: 20091223__issue16510.diff.txt
+ uploaded by tilghman (license 14) Tested by: atis ........
+
+ * /, res/res_agi.c: Merged revisions 236186 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r236186 | tilghman | 2009-12-22 21:07:48 -0600 (Tue, 22 Dec 2009)
+ | 11 lines Merged revisions 236184 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236184 | tilghman | 2009-12-22 20:55:24 -0600 (Tue, 22 Dec 2009)
+ | 4 lines If EXEC only gets a single argument, don't crash when
+ the second is used. (closes issue #16504) Reported by: bklang
+ ........ ................
+
+2009-12-22 17:06 +0000 [r236065] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 236063 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r236063 | dvossel | 2009-12-22 11:00:08 -0600 (Tue, 22 Dec 2009)
+ | 18 lines Merged revisions 236062 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r236062 | dvossel | 2009-12-22 10:58:19 -0600 (Tue, 22 Dec 2009)
+ | 11 lines fixes issue with p->method incorrectly set to ACK It
+ is possible for a second ACK to come in for a retransmitted
+ message. If an ack does not match an unacked message in our
+ queue, restore the previous p->method as this ACK is completely
+ ignored. (closes issue #16295) Reported by: omolenkamp Patches:
+ issue16295_v2.diff uploaded by dvossel (license 671) ........
+ ................
+
+2009-12-21 19:55 +0000 [r235943] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 235941 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r235941 | jpeeler | 2009-12-21 13:54:20 -0600 (Mon, 21 Dec 2009)
+ | 20 lines Merged revisions 235940 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r235940 | jpeeler | 2009-12-21 13:43:41 -0600 (Mon, 21 Dec 2009)
+ | 13 lines Change Monitor to not assume file to write to does not
+ contain pathing. 227944 changed the fname_base argument to always
+ append the configured monitor path. This change was necessary to
+ properly compare files for uniqueness. If a full path is given
+ though, nothing needs to be appended and that is handled
+ correctly now. (closes issue #16377) (closes issue #16376)
+ Reported by: bcnit Patches: res_monitor.c-issue16376-1.patch
+ uploaded by dant (license 670) ........ ................
+
+2009-12-21 17:11 +0000 [r235825] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/features.c: Merged revisions 235822 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r235822 | tilghman | 2009-12-21 11:00:46 -0600 (Mon, 21 Dec 2009)
+ | 15 lines Merged revisions 235821 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r235821 | tilghman | 2009-12-21 10:45:03 -0600 (Mon, 21 Dec 2009)
+ | 8 lines Send parking lot announcement to the channel which
+ parked the call, not the park-ee. (closes issue #16234) Reported
+ by: yeshuawatso Patches: 20091210__issue16234.diff.txt uploaded
+ by tilghman (license 14) 20091221__issue16234__1.4.diff.txt
+ uploaded by tilghman (license 14) Tested by: yeshuawatso ........
+ ................
+
+2009-12-20 09:07 +0000 [r235776] Alec L Davis <sivad.a@paradise.net.nz>
+
+ * main/dsp.c: restarts busydetector (if enabled) when DTMF is
+ received after call is bridged. (closes issue #16389) Reported
+ by: alecdavis Tested by: alecdavis Patch
+ dtmf_busydetector.diff2.txt uploaded by alecdavis (license 585)
+
+2009-12-18 23:03 +0000 [r235663] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /, include/asterisk/cdr.h: Merged revisions
+ 235660 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r235660 | jpeeler | 2009-12-18 16:51:37 -0600 (Fri, 18 Dec 2009)
+ | 55 lines Merged revisions 235635 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009)
+ | 48 lines Correct CDR dispositions for BUSY/FAILED This patch is
+ simple in that it reorders the disposition defines so that the
+ fix for issue 12946 works properly (the default CDR disposition
+ was changed to AST_CDR_NOANSWER). Also, the
+ AST_CDR_FLAG_ORIGINATED flag was set in ast_call to ensure all
+ CDR records are written. The side effects of CDR changes are
+ scary, so I'm documenting the test cases performed to attempt to
+ catch any regressions. The following tests were all performed
+ using 1.4 rev 195881 vs head (235571) + patch: A calls B C calls
+ B (busy) Hangup C Hangup A (Both SIP and features) A calls B A
+ blind transfers to C Hangup C (Both SIP and features) A calls B A
+ attended transfers to C Hangup C A calls B A attended transfers
+ to C (SIP) C blind transfers to A (features) Hangup A All of the
+ test scenario CDRs matched. The following tests were performed
+ just with the patch to ensure proper operation (with
+ unanswered=yes): exten =>s,1,Answer exten =>s,n,ResetCDR(w) exten
+ =>s,n,ResetCDR(w) exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w)
+ (closes issue #16180) Reported by: aatef Patches: bug16180.patch
+ uploaded by jpeeler (license 325) ........ ................
+
+2009-12-18 22:42 +0000 [r235575-235658] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 235656 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r235656 | tilghman | 2009-12-18 16:40:46 -0600
+ (Fri, 18 Dec 2009) | 9 lines Merged revisions 235652 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r235652 | tilghman | 2009-12-18 16:39:30 -0600 (Fri, 18
+ Dec 2009) | 2 lines Revise verbiage, per #asterisk-dev discussion
+ ........ ................
+
+ * /, configure, configure.ac: Merged revisions 235573 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r235573 | tilghman | 2009-12-18 15:19:43 -0600
+ (Fri, 18 Dec 2009) | 9 lines Merged revisions 235572 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r235572 | tilghman | 2009-12-18 15:18:16 -0600 (Fri, 18
+ Dec 2009) | 2 lines Point to the typical missing package, not the
+ cryptic "termcap support". ........ ................
+
+2009-12-17 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.1.12
+
+2009-12-09 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.1.12-rc1
+
+2009-12-08 18:31 +0000 [r233730] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 233718 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r233718 | tilghman | 2009-12-08 12:22:44 -0600 (Tue, 08 Dec 2009)
+ | 8 lines Find another ref leak and change how we manage module
+ references. (closes issue #16388) Reported by: parisioa Patches:
+ 20091208__issue16388.diff.txt uploaded by tilghman (license 14)
+ Tested by: parisioa, tilghman Review:
+ https://reviewboard.asterisk.org/r/442/ ........
+
+2009-12-08 18:02 +0000 [r233693] Russell Bryant <russell@digium.com>
+
+ * formats/format_ilbc.c, formats/format_vox.c,
+ formats/format_pcm.c, formats/format_g723.c,
+ formats/format_h263.c, formats/format_h264.c,
+ formats/format_g726.c, formats/format_jpeg.c,
+ formats/format_gsm.c, formats/format_g729.c, /,
+ formats/format_sln.c, formats/format_wav.c,
+ formats/format_ogg_vorbis.c, formats/format_sln16.c,
+ formats/format_wav_gsm.c: Merged revisions 233692 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r233692 | russell | 2009-12-08 12:00:16 -0600 (Tue, 08 Dec 2009)
+ | 16 lines Set a module load priority for format modules. A
+ recent change to app_voicemail made it such that the module now
+ assumes that all format modules are available while processing
+ voicemail configuration. However, when autoloading modules, it
+ was possible that app_voicemail was loaded before the format
+ modules. Since format modules don't depend on anything, set a
+ module load priority on them to ensure that they get loaded first
+ when autoloading. This fix applies to trunk, 1.6.1, and 1.6.2.
+ The fix for 1.4 and 1.6.0 will require a different approach since
+ the module load priority functionality is not present in the
+ module API. (issue #16412) Reported by: jiddings ........
+
+2009-12-08 07:40 +0000 [r233688] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c: Fixed compile error with OSP Toolkit 3.6.
+
+2009-12-07 23:56 +0000 [r233616] 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 23:29 +0000 [r233474-233613] David Vossel <dvossel@digium.com>
+
+ * /, main/utils.c: Merged revisions 233611 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r233611 |
+ dvossel | 2009-12-07 17:28:51 -0600 (Mon, 07 Dec 2009) | 4 lines
+ fixes incorrect logic in ast_uri_encode issue #16299 ........
+
+ * /, channels/chan_sip.c: Merged revisions 233472 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r233472 | dvossel | 2009-12-07 12:08:46 -0600 (Mon, 07 Dec 2009)
+ | 15 lines Merged revisions 233471 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r233471 | dvossel | 2009-12-07 12:07:38 -0600 (Mon, 07 Dec 2009)
+ | 9 lines fixes missing Contact header angle brackets (closes
+ issue #16298) Reported by: mgernoth Patches:
+ reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested
+ by: dvossel ........ ................
+
+2009-12-07 16:16 +0000 [r233395] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 233394 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r233394 |
+ mnicholson | 2009-12-07 10:14:42 -0600 (Mon, 07 Dec 2009) | 8
+ lines Do not reject SDP packets describing only non audio
+ streams. (closes issue #16387) Reported by: zalex1953 Patches:
+ media-level-c-fix1.diff uploaded by mnicholson (license 96)
+ Tested by: mnicholson, zalex1953 ........
+
+2009-12-04 21:55 +0000 [r233283] David Vossel <dvossel@digium.com>
+
+ * configs/iax.conf.sample, /: Merged revisions 233280 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r233280 | dvossel | 2009-12-04 15:54:44 -0600
+ (Fri, 04 Dec 2009) | 14 lines Merged revisions 233279 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r233279 | dvossel | 2009-12-04 15:54:01 -0600 (Fri, 04 Dec 2009)
+ | 7 lines clarify requirecalltoken option in iax.sample.conf
+ (closes issue #16223) Reported by: bklang Patches:
+ clarify-iax-requirecalltoken.patch uploaded by bklang (license
+ 919) ........ ................
+
+2009-12-04 21:00 +0000 [r233238] Matthias Nick <mnick@digium.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 233093 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r233093 |
+ mnick | 2009-12-04 11:15:47 -0600 (Fri, 04 Dec 2009) | 8 lines
+ Parse global variables or expressions in hint extensions Parse
+ global variables or expressions in hint extensions. Like: exten
+ => 400,hint,DAHDI/i2/${GLOBAL(var)} (closes issue #16166)
+ Reported by: rmudgett Tested by: mnick, rmudgett ........
+
+2009-12-04 17:37 +0000 [r233166] David Vossel <dvossel@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 233121 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r233121 | dvossel | 2009-12-04 11:22:31 -0600
+ (Fri, 04 Dec 2009) | 12 lines Merged revisions 233116 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r233116 | dvossel | 2009-12-04 11:21:34 -0600 (Fri, 04 Dec 2009)
+ | 6 lines document and rename strip_control() in app_voicemail
+ (closes issue #16291) Reported by: wdoekes ........
+ ................
+
+2009-12-04 17:22 +0000 [r233122] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 233100 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r233100 | russell | 2009-12-04 11:18:22 -0600 (Fri, 04 Dec 2009)
+ | 14 lines Merged revisions 233092 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r233092 | russell | 2009-12-04 11:12:47 -0600 (Fri, 04 Dec 2009)
+ | 7 lines Only do frame payload check for HOLD frames. This code
+ was added for helping to debug the source of invalid HOLD frames.
+ However, a side effect of this is that it will incorrectly report
+ errors for frames that have an integer payload. Make the check
+ for this block specific to the HOLD frame case. ........
+ ................
+
+2009-12-04 15:51 +0000 [r233048] Matthias Nick <mnick@digium.com>
+
+ * main/dsp.c, /: Merged revisions 233046 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r233046 | mnick | 2009-12-04 09:38:33 -0600 (Fri, 04 Dec 2009) |
+ 17 lines Merged revisions 233014 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r233014 | mnick | 2009-12-04 09:17:03 -0600 (Fri, 04 Dec 2009) |
+ 11 lines Warning message gets displayed only once Added
+ additional field 'int display_inband_dtmf_warning', which when
+ set to '1' displays the warning ('Inband DTMF is not supported on
+ codec %s. Use RFC2833'), and when set to '0' doesn't display the
+ warning. Otherwise you would get hundreds of warnings every
+ second. (closes issue #15769) Reported by: falves11 Patches:
+ patch_15769_14.txt uploaded by mnick (license 874) Tested by:
+ mnick, falves11 ........ ................
+
+2009-12-03 21:03 +0000 [r232865] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 232854 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r232854 | tilghman | 2009-12-03 14:47:07 -0600
+ (Thu, 03 Dec 2009) | 15 lines Merged revisions 232820 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232820 | tilghman | 2009-12-03 14:10:19 -0600 (Thu, 03 Dec 2009)
+ | 8 lines Deprecate "cz" in favor of "cs". Also, change the use
+ of language codes so that language registers as a prefix, rather
+ than an exact match. (closes issue #16272) Reported by: patrol-cz
+ Patches: 20091203__issue16272.diff.txt uploaded by tilghman
+ (license 14) ........ ................
+
+2009-12-03 15:03 +0000 [r232812] David Ruggles <thedavidfactor@gmail.com>
+
+ * apps/app_externalivr.c: Merged revisions 232587 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r232587 |
+ diruggles | 2009-12-02 17:17:22 -0500 (Wed, 02 Dec 2009) | 12
+ lines Prevent double closing of FDs by EIVR This caused a problem
+ when asterisk was under heavy load and running both AGI and EIVR
+ applications. EIVR would close an FD at which point it would be
+ considered freed and be used by a new AGI instance the second
+ close would then close the FD now in use by AGI. (closes issue
+ #16305) Reported by: diLLec Tested by: thedavidfactor, diLLec
+ Review: https://reviewboard.asterisk.org/r/436/ ........
+
+2009-12-03 00:18 +0000 [r232666] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_musiconhold.c: Recorded merge of revisions
+ 232660-232661 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r232660 |
+ tilghman | 2009-12-02 18:08:55 -0600 (Wed, 02 Dec 2009) | 19
+ lines Fix multiple issues with musiconhold, which led to classes
+ not getting destroyed properly. * Classes are now tracked past
+ removal from the core container, and module removal is actively
+ prevented until all references are freed. * A hanging reference
+ stored in the channel has been removed. This could have caused a
+ mismatch and the music state not properly cleared, if two or more
+ reloads occurred between MOH being stopped and MOH being
+ restarted. * In certain circumstances, duplicate classes were
+ possible. * A race existed at reload time between a process being
+ killed and the thread responsible for reading from the related
+ pipe respawning that process. * Several reference counts have
+ also been corrected. At least one could have caused deleted
+ classes to stick around forever, consuming resources. This
+ originally manifested as MOH external processes that were not
+ killed at reload time. (closes issue #16279, closes issue #16207)
+ Reported by: parisioa, dcabot Patches:
+ 20091202__issue16279__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: parisioa, tilghman ........ r232661 | tilghman |
+ 2009-12-02 18:09:36 -0600 (Wed, 02 Dec 2009) | 2 lines Remove
+ debugging line ........
+
+2009-12-02 22:04 +0000 [r232578-232584] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c, /: Merged revisions 232582 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r232582 | jpeeler | 2009-12-02 16:02:43 -0600 (Wed, 02 Dec 2009)
+ | 14 lines Merged revisions 232581 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232581 | jpeeler | 2009-12-02 15:57:42 -0600 (Wed, 02 Dec 2009)
+ | 7 lines Send ack (response/message) after receiving manager
+ action userevent (closes issue #16264) Reported by: dimas
+ Patches: event-ack.patch uploaded by dimas (license 88) ........
+ ................
+
+ * main/manager.c, /: Merged revisions 232576 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r232576 |
+ jpeeler | 2009-12-02 15:32:50 -0600 (Wed, 02 Dec 2009) | 8 lines
+ Make manager response to "Action: events" finish with empty line
+ (closes issue #16275) Reported by: vnovy Patches: manager.c.diff
+ uploaded by vnovy (license 922) ........
+
+2009-12-02 17:10 +0000 [r232358] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_amd.c: Merged revisions 232356 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r232356 | file | 2009-12-02 13:06:54 -0400 (Wed, 02 Dec 2009) |
+ 12 lines Merged revisions 232355 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232355 | file | 2009-12-02 13:04:52 -0400 (Wed, 02 Dec 2009) | 5
+ lines Fix a bug where if you hung up very quickly after calling
+ AMD it would overwrite the AMDSTATUS of HANGUP with TOOLONG.
+ (closes issue #16239) Reported by: CGMChris ........
+ ................
+
+2009-12-02 17:02 +0000 [r232353] David Vossel <dvossel@digium.com>
+
+ * /, main/acl.c: Merged revisions 232351 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r232351 | dvossel | 2009-12-02 11:00:15 -0600 (Wed, 02 Dec 2009)
+ | 12 lines Merged revisions 232350 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232350 | dvossel | 2009-12-02 10:59:18 -0600 (Wed, 02 Dec 2009)
+ | 6 lines ast_outaddrfor doesn't do htons() on port, looks odd in
+ strace. (closes issue #16290) Reported by: wdoekes ........
+ ................
+
+2009-12-02 16:42 +0000 [r232347] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 232345 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r232345 |
+ file | 2009-12-02 12:40:14 -0400 (Wed, 02 Dec 2009) | 7 lines Add
+ support for handling the 415 Unsupported media type response like
+ we do for a 488 Not acceptable here response. (closes issue
+ #16186) Reported by: atis Patches: sip_t38_response_415.patch
+ uploaded by atis (license 242) ........
+
+2009-12-02 15:43 +0000 [r232271] David Vossel <dvossel@digium.com>
+
+ * funcs/func_groupcount.c, /: Merged revisions 232269 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r232269 | dvossel | 2009-12-02 09:42:54 -0600
+ (Wed, 02 Dec 2009) | 15 lines Merged revisions 232268 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232268 | dvossel | 2009-12-02 09:41:36 -0600 (Wed, 02 Dec 2009)
+ | 9 lines fixes segfault in func_groupcount closes issue #16337)
+ Reported by: Parantido Patches: issue_16337.diff uploaded by
+ dvossel (license 671) Tested by: Parantido, dvossel ........
+ ................
+
+2009-12-02 14:55 +0000 [r232231] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 232230 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r232230 |
+ file | 2009-12-02 10:54:28 -0400 (Wed, 02 Dec 2009) | 5 lines Fix
+ a bug where a scheduled item ID would get retained on
+ registrations in a certain scenario causing code to execute
+ during reload that should not. (issue AST-263) ........
+
+2009-12-02 00:51 +0000 [r232093] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 232091 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r232091 | jpeeler | 2009-12-01 18:45:18 -0600
+ (Tue, 01 Dec 2009) | 17 lines Merged revisions 232090 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01 Dec 2009)
+ | 10 lines Do not modify the gain settings on data calls. (The
+ digital flag actually represents a data call.) (closes issue
+ #15972) Reported by: udosw Patches: transcap_digital_fix.diff.txt
+ uploaded by alecdavis (license 585) Tested by: alecdavis ........
+ ................
+
+2009-12-01 23:39 +0000 [r232010-232014] Russell Bryant <russell@digium.com>
+
+ * /, funcs/func_lock.c: Merged revisions 232012 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r232012 |
+ russell | 2009-12-01 17:38:34 -0600 (Tue, 01 Dec 2009) | 2 lines
+ Fix a build error on FreeBSD. ........
+
+ * /, main/file.c: Merged revisions 232008 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r232008 | russell | 2009-12-01 17:27:53 -0600 (Tue, 01 Dec 2009)
+ | 9 lines Merged revisions 232007 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r232007 | russell | 2009-12-01 17:25:36 -0600 (Tue, 01 Dec 2009)
+ | 2 lines Fix a warning pointed out by buildbot. ........
+ ................
+
+2009-12-01 22:00 +0000 [r231929] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 231927 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r231927 | jpeeler | 2009-12-01 15:54:21 -0600 (Tue, 01 Dec 2009)
+ | 19 lines Merged revisions 231911 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231911 | jpeeler | 2009-12-01 15:29:31 -0600 (Tue, 01 Dec 2009)
+ | 12 lines Fix crash with invalid frame data The crash was
+ happening as a result of a frame containing an invalid data
+ pointer, but was set with data length of zero. The few times the
+ issue was reproduced it _seemed_ that the frame was queued
+ properly, that is the data pointer was set to NULL. I never could
+ reproduce the crash so as a last resort the crash has been fixed,
+ but a check in __ast_read has been added to give as much
+ information about the source of problematic frames in the future.
+ (closes issue #16058) Reported by: atis ........ ................
+
+2009-12-01 21:21 +0000 [r231876] David Vossel <dvossel@digium.com>
+
+ * main/pbx.c, /: Merged revisions 231867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r231867 | dvossel | 2009-12-01 15:20:19 -0600 (Tue, 01 Dec 2009)
+ | 9 lines Merged revisions 231853 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231853 | dvossel | 2009-12-01 15:14:31 -0600 (Tue, 01 Dec 2009)
+ | 3 lines WaitExten m option with no parameters generates frame
+ with zero datalen but non-null data ptr ........ ................
+
+2009-12-01 15:48 +0000 [r231742] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/file.c: Merged revisions 231741 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r231741 | mnicholson | 2009-12-01 09:47:36 -0600 (Tue, 01 Dec
+ 2009) | 9 lines Merged revisions 231740 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231740 | mnicholson | 2009-12-01 09:34:57 -0600 (Tue, 01 Dec
+ 2009) | 2 lines Ignore unknown formats in ast_format_str_reduce()
+ and return an error if no know formats are found. ........
+ ................
+
+2009-11-30 21:55 +0000 [r231694] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h:
+ Merged revisions 231692 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r231692 |
+ kpfleming | 2009-11-30 15:47:42 -0600 (Mon, 30 Nov 2009) | 22
+ lines Another round of UDPTL stack fixes/improvements: 1) Allow
+ users of UDPTL stack to associate a character-string tag with a
+ UDPTL session, so that log/error/debug messages generated by the
+ UDPTL stack can be 'connected' to the endpoint that caused them
+ to be generated. 2) Improve comments (and process) of calculating
+ the far end's maximum IFP size when redundancy mode is in use for
+ error correction. 3) When an IFP larger than the calculated 'far
+ max IFP' size is presented for writing, truncate it rather than
+ putting in the buffer and allowing the buffer to overflow; this
+ will cause the ends to retrain to a lower bit rate that produces
+ IFPs of an appropriate size if possible, and if not possible, the
+ FAX transfer will fail completely. In these cases, it is due to
+ the one endpoint supplying a T38FaxMaxDatagram value that is
+ improperly calculated and is too low to be of use; we have
+ configuration options available to override this behavior. 4)
+ Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no
+ longer needed. ........
+
+2009-11-30 21:36 +0000 [r231690] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_voicemail.c, include/asterisk/file.h, /, main/file.c,
+ main/app.c: Merged revisions 231688 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r231688 | mnicholson | 2009-11-30 15:31:55 -0600 (Mon, 30 Nov
+ 2009) | 15 lines Merged revisions 231614 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov
+ 2009) | 8 lines Remove duplicate entries from voicemail format
+ lists. This prevents app_voicemail from entering an infinite loop
+ when the same format is specified twice in the format list.
+ (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson
+ Review: https://reviewboard.asterisk.org/r/429/ ........
+ ................
+
+2009-11-30 20:58 +0000 [r231608] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c: Turn off debug mode in 1.6.1; fix such that
+ debug mode and non-debug mode functions return the same types.
+ (Fixes an issue brought up in chat by twilson)
+
+2009-11-30 20:47 +0000 [r231604] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 231602 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r231602 |
+ file | 2009-11-30 16:44:30 -0400 (Mon, 30 Nov 2009) | 5 lines
+ When receiving SDP that matches the version of the last one do
+ not treat it as a fatal error. (closes issue #16238) Reported by:
+ seandarcy ........
+
+2009-11-30 18:57 +0000 [r231512-231559] David Vossel <dvossel@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 231556 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r231556 |
+ dvossel | 2009-11-30 12:55:07 -0600 (Mon, 30 Nov 2009) | 11 lines
+ app_queue crashes randomly, often during call-transfers This
+ patch adds a ref to the queue_ent object's parent call_queue in
+ queue_exec() so the call_queue won't be destroyed while the the
+ queue_ent still holds a pointer to it. (closes issue 0015686)
+ Tested by: dvossel, aragon ........
+
+ * main/rtp.c, /: Merged revisions 231491 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r231491 | dvossel | 2009-11-30 11:28:28 -0600 (Mon, 30 Nov 2009)
+ | 17 lines Merged revisions 231441 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231441 | dvossel | 2009-11-30 11:14:08 -0600 (Mon, 30 Nov 2009)
+ | 11 lines fixes crash caused by RTP comfort noise payload
+ greater than 24 bytes AST-2009-010 (closes issue #16242) Reported
+ by: amorsen Patches: issue16242.diff uploaded by oej (license
+ 306) Tested by: amorsen, oej, dvossel ........ ................
+
+2009-11-25 22:34 +0000 [r231301] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 231299 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r231299 | tilghman | 2009-11-25 16:33:02 -0600 (Wed, 25 Nov 2009)
+ | 9 lines Merged revisions 231298 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r231298 | tilghman | 2009-11-25 16:31:57 -0600 (Wed, 25 Nov 2009)
+ | 2 lines After a frame duplication failure, unlock the channel
+ before returning. ........ ................
+
+2009-11-25 15:44 +0000 [r231190] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, pbx/pbx_lua.c: Merged revisions 231189 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r231189 |
+ mnicholson | 2009-11-25 09:42:48 -0600 (Wed, 25 Nov 2009) | 4
+ lines Load pbx_lua with global symbols to allow linking with
+ other lua libraries. Found by Maxim Litnitskiy. ........
+
+2009-11-24 20:35 +0000 [r231135] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 231134 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r231134 |
+ tilghman | 2009-11-24 14:31:28 -0600 (Tue, 24 Nov 2009) | 7 lines
+ Found a few places where queue refcounts were counted
+ incorrectly. Also add debug statements. (closes issue #15982,
+ closes issue #15984) Reported by: atis Patches:
+ 20091111__issue15982.diff.txt uploaded by tilghman (license 14)
+ Tested by: atis ........
+
+2009-11-24 18:54 +0000 [r231097] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 231095 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r231095 |
+ jpeeler | 2009-11-24 12:50:36 -0600 (Tue, 24 Nov 2009) | 11 lines
+ Fix erroneous hangup extension execution ast_spawn_extension
+ behaves differently from 1.4 in that hangups and extensions that
+ do not exist do not return an error, whereas in 1.6 it does. This
+ is now taken into account so that the AST_FLAG_BRIDGE_HANGUP_RUN
+ flag gets set properly. (closes issue #16106) Reported by:
+ ajohnson Tested by: ajohnson ........
+
+2009-11-23 15:47 +0000 [r230883] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+ 230881 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r230881 |
+ file | 2009-11-23 09:45:45 -0600 (Mon, 23 Nov 2009) | 7 lines
+ Change fax detection in chan_sip so it behaves as one would
+ expect. Internally the way T.38 is negotiated has changed and the
+ option no longer reflects a behavior that is valid. It will now
+ look for a CNG tone on received calls and if present send the
+ call to the 'fax' extension. It is then up to the application or
+ channel to request the switch over to T.38. ........
+
+2009-11-23 15:36 +0000 [r230790-230879] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 230877 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r230877 | kpfleming | 2009-11-23 09:34:16 -0600 (Mon, 23 Nov
+ 2009) | 9 lines Merged revisions 230839 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r230839 | kpfleming | 2009-11-23 09:09:24 -0600 (Mon, 23 Nov
+ 2009) | 1 line Correct fix for issue #16268... the reporter's
+ original patch was very close to correct. ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 230773 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r230773 | kpfleming | 2009-11-23 08:15:48 -0600 (Mon, 23 Nov
+ 2009) | 12 lines Merged revisions 230772 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r230772 | kpfleming | 2009-11-23 08:13:56 -0600 (Mon, 23 Nov
+ 2009) | 5 lines Ensure that SDP parsing does not ignore the last
+ line of the SDP. (closes issue #16268) Reported by: sgimeno
+ ........ ................
+
+2009-11-20 22:37 +0000 [r230728] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 230726 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r230726 | dvossel | 2009-11-20 16:35:54 -0600 (Fri, 20 Nov 2009)
+ | 7 lines fixes iax2 show cache locking error, thanks alecdavis!
+ (closes issue #16094) Reported by: alecdavis Patches:
+ bug16094.diff.txt uploaded by alecdavis (license 585) Tested by:
+ alecdavis, dvossel ........
+
+2009-11-20 21:08 +0000 [r230630] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/features.c: Merged revisions 230628 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r230628 | mnicholson | 2009-11-20 15:01:10 -0600 (Fri, 20 Nov
+ 2009) | 15 lines Merged revisions 230627 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r230627 | mnicholson | 2009-11-20 14:53:06 -0600 (Fri, 20 Nov
+ 2009) | 8 lines Copy the peer CDR's userfield to the bridge CDR
+ if it exists. This is necessary for the recordagentcalls option
+ in chan_agent to store the recorded file name in the bridge CDR.
+ (closes issue #14590) Reported by: msetim Patches:
+ queue_agent_userfield.patch uploaded by Laureano (license 265)
+ Tested by: Laureano, mnicholson ........ ................
+
+2009-11-20 17:32 +0000 [r230511-230586] David Vossel <dvossel@digium.com>
+
+ * /, include/asterisk/audiohook.h, main/audiohook.c: Merged
+ revisions 230583 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r230583 |
+ dvossel | 2009-11-20 11:26:20 -0600 (Fri, 20 Nov 2009) | 6 lines
+ audiohook signal trigger on every status change (issue #14618)
+ Review: https://reviewboard.asterisk.org/r/434/ ........
+
+ * apps/app_mixmonitor.c, /: Merged revisions 230509 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r230509 | dvossel | 2009-11-19 15:26:21 -0600
+ (Thu, 19 Nov 2009) | 17 lines Merged revisions 230508 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r230508 | dvossel | 2009-11-19 15:22:46 -0600 (Thu, 19 Nov 2009)
+ | 10 lines fixes MixMonitor thread not exiting when
+ StopMixMonitor is used (closes issue #16152) Reported by: AlexMS
+ Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license
+ 671) Tested by: dvossel, AlexMS Review:
+ https://reviewboard.asterisk.org/r/424/ ........ ................
+
+2009-11-30 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.1.11
+
+ * AST-2009-010
+
+ * SDP parser regression fix (issue #16268, issue #16238)
+
+2009-11-18 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.1.10
+
+2009-11-13 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.1.10-rc3
+
+2009-11-13 15:57 +0000 [r229914] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 229912 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r229912 |
+ file | 2009-11-13 09:56:16 -0600 (Fri, 13 Nov 2009) | 2 lines Fix
+ T.38 negotiation regression introduced with the SDP parser
+ changes. ........
+
+2009-11-12 23:31 +0000 [r229751] Jason Parker <jparker@digium.com>
+
+ * channels/chan_oss.c, /: Merged revisions 229750 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r229750 |
+ qwell | 2009-11-12 17:30:10 -0600 (Thu, 12 Nov 2009) | 1 line Fix
+ mute toggling on OSS channels. ........
+
+2009-11-12 16:48 +0000 [r229672] David Vossel <dvossel@digium.com>
+
+ * funcs/func_audiohookinherit.c, /: Merged revisions 229670 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r229670 | dvossel | 2009-11-12 10:44:39 -0600
+ (Thu, 12 Nov 2009) | 12 lines Merged revisions 229669 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229669 | dvossel | 2009-11-12 10:41:49 -0600 (Thu, 12 Nov 2009)
+ | 6 lines fixes merging error, datastore was being freed in the
+ wrong function. (closes issue #16219) Reported by: aragon
+ ........ ................
+
+2009-11-11 20:48 +0000 [r229569] David Ruggles <thedavidfactor@gmail.com>
+
+ * doc/externalivr.txt: Merged revisions 229568 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r229568 |
+ diruggles | 2009-11-11 15:47:06 -0500 (Wed, 11 Nov 2009) | 9
+ lines Remove non-functional feature from ExternalIVR
+ documentation Remove non-functional socket implementation of
+ ExternalIVR from documentation (closes issue #16225) Reported by:
+ thedavidfactor Patches: externalivr.txt.20091111.1542.patch
+ uploaded by thedavidfactor (license 903) ........
+
+2009-11-11 19:54 +0000 [r229491-229501] David Brooks <dbrooks@digium.com>
+
+ * main/pbx.c, /: Merged revisions 229499 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r229499 | dbrooks | 2009-11-11 13:48:18 -0600 (Wed, 11 Nov 2009)
+ | 15 lines Merged revisions 229498 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229498 | dbrooks | 2009-11-11 13:46:19 -0600 (Wed, 11 Nov 2009)
+ | 8 lines Solaris doesn't like NULL going to ast_log Solaris will
+ crash if NULL is passed to ast_log. This simple patch simply uses
+ S_OR to get around this. (closes issue #15392) Reported by:
+ yrashk ........ ................
+
+ * /, apps/app_softhangup.c: Merged revisions 229460 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r229460 | dbrooks | 2009-11-11 12:13:56 -0600 (Wed, 11 Nov 2009)
+ | 7 lines Flags not initialized in app_softhangup.c, causing
+ undefined behavior Trivial patch [kobaz] to initialize an
+ ast_flags = {0} (closes issue #16129) Reported by: kobaz ........
+
+2009-11-10 22:17 +0000 [r229364] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 229361 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r229361 | tilghman | 2009-11-10 16:14:22 -0600 (Tue, 10 Nov 2009)
+ | 19 lines Merged revisions 229360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229360 | tilghman | 2009-11-10 16:09:16 -0600 (Tue, 10 Nov 2009)
+ | 12 lines If two pattern classes start with the same digit and
+ have the same number of characters, they will compare equal. The
+ example given in the issue report is that of [234] and [246],
+ which have these characteristics, yet they are clearly not
+ equivalent. The code still uses these two characteristics, yet
+ when the two scores compare equal, an additional check will be
+ done to compare all characters within the class to verify
+ equality. (closes issue #15421) Reported by: jsmith Patches:
+ 20091109__issue15421__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: jsmith, thedavidfactor ........ ................
+
+2009-11-10 22:04 +0000 [r229358] David Ruggles <thedavidfactor@gmail.com>
+
+ * doc/externalivr.txt: Merged revisions 229356 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r229356 | diruggles | 2009-11-10 17:01:50 -0500 (Tue, 10 Nov
+ 2009) | 16 lines Merged revisions 229355 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229355 | diruggles | 2009-11-10 16:45:15 -0500 (Tue, 10 Nov
+ 2009) | 9 lines Fix ExternalIVR Documentation Remove
+ documentation for event that doesn't function (closes issue
+ #16220) Reported by: thedavidfactor Patches:
+ externalivr.txt.20091110.1622.patch uploaded by thedavidfactor
+ (license 903) ........ ................
+
+2009-11-10 21:31 +0000 [r229353] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_stack.c, /: Merged revisions 229351 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r229351 |
+ tilghman | 2009-11-10 15:22:50 -0600 (Tue, 10 Nov 2009) | 7 lines
+ When GOSUB is invoked within an AGI, it may not exit correctly.
+ (closes issue #16216) Reported by: atis Patches:
+ 20091110__atis_work.diff.txt uploaded by tilghman (license 14)
+ Tested by: atis ........
+
+2009-11-10 20:09 +0000 [r229284] Joshua Colp <jcolp@digium.com>
+
+ * /, codecs/codec_g726.c: Merged revisions 229282 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r229282 | file | 2009-11-10 16:06:13 -0400 (Tue, 10 Nov 2009) |
+ 15 lines Merged revisions 229281 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229281 | file | 2009-11-10 16:03:14 -0400 (Tue, 10 Nov 2009) | 8
+ lines Remove broken support for direct transcoding between G.726
+ RFC3551 and G.726 AAL2. On some systems the translation core
+ would actually consider g726aal2 -> g726 -> signed linear to be a
+ quicker path then g726aal2 -> signed linear which exposed this
+ problem. (closes issue #15504) Reported by: globalnetinc ........
+ ................
+
+2009-11-10 17:53 +0000 [r229233] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 229168 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r229168 | dvossel | 2009-11-10 11:16:49 -0600
+ (Tue, 10 Nov 2009) | 15 lines Merged revisions 229167 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229167 | dvossel | 2009-11-10 11:15:57 -0600 (Tue, 10 Nov 2009)
+ | 9 lines don't crash on log message in solaris AST-2009-006
+ (closes issue #16206) Reported by: bklang Tested by: bklang
+ ........ ................
+
+2009-11-10 17:38 +0000 [r229230] David Ruggles <thedavidfactor@gmail.com>
+
+ * doc/externalivr.txt: Merged revisions 229228 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r229228 | diruggles | 2009-11-10 12:33:47 -0500 (Tue, 10 Nov
+ 2009) | 18 lines Merged revisions 229191 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r229191 | diruggles | 2009-11-10 12:23:59 -0500 (Tue, 10 Nov
+ 2009) | 11 lines Document ExternalIVR event tag collision
+ ExternalIVR uses the D tag for two different event types. This
+ documents that behavior and how to differentiate between the two
+ cases. Also includes a minor spelling fix and clarification
+ (closes issue #16211) Reported by: thedavidfactor Patches:
+ externalivr.txt.20091109.1507.patch uploaded by thedavidfactor
+ (license 903) ........ ................
+
+2009-11-10 15:38 +0000 [r229099] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Reverted revision 202008. (closes issue
+ #16175) Reported by: paul-tg
+
+2009-11-10 15:36 +0000 [r229095-229098] David Vossel <dvossel@digium.com>
+
+ * res/res_config_pgsql.c: reverting changes made by r229095 as they
+ are not applicable to 1.6.1
+
+ * res/res_config_pgsql.c, /: Merged revisions 229093 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r229093 | dvossel | 2009-11-10 09:27:45 -0600 (Tue, 10 Nov 2009)
+ | 11 lines fixes pgsql double free of threadstorage A thread
+ storage variable was being freed incorrectly, which resulted in a
+ double free if two queries were made in the same thread. (closes
+ issue #16011) Reported by: cristiandimache Patches:
+ issue16011.diff uploaded by dvossel (license 671) ........
+
+2009-11-10 11:22 +0000 [r229067] Gavin Henry <ghenry@suretecsystems.com>
+
+ * contrib/scripts/asterisk.ldap-schema, /: Merged revisions 229050
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r229050 | ghenry | 2009-11-10 11:16:10 +0000 (Tue, 10
+ Nov 2009) | 20 lines Schema file additions * Added
+ AsteriskDialplan, AsteriskAccount and AsteriskMailbox
+ objectClasses to allow standalone dialplan, account and mailbox
+ entries (STRUCTURAL) * Added new Fields: - AstAccountLanguage,
+ AstAccountTransport, AstAccountPromiscRedir, -
+ AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
+ - AstAccountVideoSupport, AstAccountIgnoreSDPVersion * Removed
+ redundant IPaddr (there's already IPAddress) - Gives more
+ configuration Flags for SIP-Users available (tested) - Allows to
+ create Asterisk Attributes in defined Asterisk ObjectClasses
+ without extensibleObject (which really should be the last
+ resort); gives also additional possibilities for LDAP-filter
+ (closes issue #15874) Reported by: Medozas Patches:
+ asterisk.ldap-schema.patch uploaded by Medozas (license 41)
+ Tested by: Medozas, suretec ........
+
+2009-11-09 22:52 +0000 [r229016] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 229015 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r229015 | twilson | 2009-11-09 16:50:22 -0600 (Mon, 09 Nov 2009)
+ | 8 lines Don't crash when bridge->tech_pvt == NULL This is a
+ similar solution to what is in place for chan_agent (closes issue
+ #16003) Reported by: atis Tested by: twilson ........
+
+2009-11-09 22:18 +0000 [r229014] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: fixes segfault when transferring a queue
+ caller In sip_hangup we attempted to lock p->owner after we set
+ it to NULL. Thanks to fhackenberger for reporting the issue and
+ submitting a patch. (closes issue 0015848) Reported by:
+ fhackenberger Patches: digium_bug_0015848 uploaded by
+ fhackenberger (license 592) Tested by: fhackenberger, lmadsen,
+ TomS, shin-shoryuken, dvossel
+
+2009-11-09 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.1.10-rc2
+
+2009-11-09 15:39 +0000 [r228899] Leif Madsen <lmadsen@digium.com>
+
+ * main/channel.c: Merged revisions 228897 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228897 | lmadsen | 2009-11-09 09:38:38 -0600 (Mon, 09 Nov 2009)
+ | 14 lines Merged revisions 228896 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228896 | lmadsen | 2009-11-09 09:37:43 -0600 (Mon, 09 Nov 2009)
+ | 6 lines Update WARNING message. Update a WARNING message to
+ give a suggested fix when encountered. (closes issue #16198)
+ Reported by: atis Tested by: atis ........ ................
+
+2009-11-09 14:54 +0000 [r228860] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 228858 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r228858 | mnicholson | 2009-11-09 08:37:07 -0600
+ (Mon, 09 Nov 2009) | 15 lines Merged revisions 228827 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228827 | mnicholson | 2009-11-09 08:16:03 -0600 (Mon, 09 Nov
+ 2009) | 8 lines Perform limited bounds checking when destroying
+ ast_mutex_t structures to make sure we don't try to use negative
+ indices. (closes issue #15588) Reported by: zerohalo Patches:
+ 20090820__issue15588.diff.txt uploaded by tilghman (license 14)
+ Tested by: zerohalo ........ ................
+
+2009-11-06 22:37 +0000 [r228695] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /: Merged revisions 228693 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228693 | dvossel | 2009-11-06 16:35:44 -0600 (Fri, 06 Nov 2009)
+ | 16 lines Merged revisions 228692 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228692 | dvossel | 2009-11-06 16:33:27 -0600 (Fri, 06 Nov 2009)
+ | 9 lines fixes audiohook write crash occuring in chan_spy
+ whisper mode. After writing to the audiohook list in ast_write(),
+ frames were being freed incorrectly. Under certain conditions
+ this resulted in a double free crash. (closes issue #16133)
+ Reported by: wetwired ........ ................
+
+2009-11-06 20:37 +0000 [r228650] Matthew Nicholson <mnicholson@digium.com>
+
+ * funcs/func_base64.c, /, main/utils.c: Merged revisions 228620 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r228620 | mnicholson | 2009-11-06 13:47:11 -0600
+ (Fri, 06 Nov 2009) | 15 lines Merged revisions 228378 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228378 | mnicholson | 2009-11-06 10:26:59 -0600 (Fri, 06 Nov
+ 2009) | 8 lines Properly handle '=' while decoding base64
+ messages and null terminate strings returned from BASE64_DECODE.
+ (closes issue #15271) Reported by: chappell Patches:
+ base64_fix.patch uploaded by chappell (license 8) Tested by:
+ kobaz ........ ................
+
+2009-11-06 18:41 +0000 [r228550] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 228548 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228548 | file | 2009-11-06 14:37:59 -0400 (Fri, 06 Nov 2009) |
+ 11 lines Merged revisions 228547 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228547 | file | 2009-11-06 14:32:58 -0400 (Fri, 06 Nov 2009) | 4
+ lines Don't overwrite caller ID name on a trunk with the
+ configured fullname when using users.conf (issue ABE-1989)
+ ........ ................
+
+2009-11-06 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.1.10-rc1
+
+2009-11-06 17:53 +0000 [r228502] Joshua Colp <jcolp@digium.com>
+
+ * /, doc/tex/localchannel.tex: Merged revisions 228499 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r228499 | file | 2009-11-06 13:52:00 -0400 (Fri, 06 Nov 2009) | 2
+ lines Fix the localchannel.tex file. ........
+
+2009-11-06 17:24 +0000 [r228422-228451] David Vossel <dvossel@digium.com>
+
+ * /, codecs/codec_ilbc.c: Merged revisions 228441 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r228441 |
+ dvossel | 2009-11-06 11:22:31 -0600 (Fri, 06 Nov 2009) | 3 lines
+ Fixes merging issue from 1.4, frame data is held in data.ptr in
+ trunk ........
+
+ * /, codecs/codec_ilbc.c: Merged revisions 228420 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228420 | dvossel | 2009-11-06 11:09:01 -0600 (Fri, 06 Nov 2009)
+ | 19 lines Merged revisions 228418 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228418 | dvossel | 2009-11-06 11:07:13 -0600 (Fri, 06 Nov 2009)
+ | 13 lines fixes segfault in iLBC For reasons not yet known, it
+ appears possible for an ast_frame to have a datalen greater than
+ zero while the actual data is NULL during Packet Loss
+ Concealment. Most codecs don't support PLC so this doesn't affect
+ them. This patch catches the malformed frame and prevents the
+ crash from occuring. Additional efforts to determine why it is
+ possible for a frame to look like this are still being
+ investigated. (issue #16979) ........ ................
+
+2009-11-06 16:44 +0000 [r228412] Joshua Colp <jcolp@digium.com>
+
+ * /, main/abstract_jb.c: Merged revisions 228410 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228410 | file | 2009-11-06 12:42:23 -0400 (Fri, 06 Nov 2009) |
+ 14 lines Merged revisions 228409 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228409 | file | 2009-11-06 12:41:20 -0400 (Fri, 06 Nov 2009) | 7
+ lines Fix a bug caused by a partially invalid frame (from the
+ jitterbuffer) passing through the Asterisk core. (closes issue
+ #15560) Reported by: jvandal (closes issue #15709) Reported by:
+ covici ........ ................
+
+2009-11-06 15:44 +0000 [r228267-228341] David Vossel <dvossel@digium.com>
+
+ * /, main/astfd.c: Merged revisions 228339 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228339 | dvossel | 2009-11-06 09:42:46 -0600 (Fri, 06 Nov 2009)
+ | 12 lines Merged revisions 228338 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228338 | dvossel | 2009-11-06 09:41:41 -0600 (Fri, 06 Nov 2009)
+ | 5 lines fixes crash in astfd.c (closes issue #15981) Reported
+ by: slavon ........ ................
+
+ * funcs/func_audiohookinherit.c, /: Merged revisions 228268 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r228268 | dvossel | 2009-11-06 09:04:24 -0600 (Fri, 06
+ Nov 2009) | 9 lines fixes memory leak in func_audiohookinherit.c
+ (closes issue #15394) Reported by: boroda Patches:
+ bug15394_memoryleak_diff2.txt uploaded by dbrooks (license 790)
+ Tested by: dbrooks, boroda ........
+
+ * /, channels/chan_sip.c: Merged revisions 227238 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r227238 |
+ dvossel | 2009-11-03 11:12:52 -0600 (Tue, 03 Nov 2009) | 5 lines
+ user.conf entries in SIP were not having their peer type set.
+ (closes issue #16120) Reported by: jsmith ........
+
+2009-11-05 22:13 +0000 [r228193-228197] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 228196 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r228196 |
+ tilghman | 2009-11-05 16:12:45 -0600 (Thu, 05 Nov 2009) | 2 lines
+ Yet another error message in the dialplan (thanks,
+ rmudgett/russellb) ........
+
+ * /, apps/app_meetme.c: Merged revisions 228191 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r228191 |
+ tilghman | 2009-11-05 15:24:21 -0600 (Thu, 05 Nov 2009) | 7 lines
+ MEETME_INFO should not return a literal error message to the
+ dialplan. (closes issue #15450) Reported by: JimVanM Patches:
+ meetmeinfopatch.diff.txt uploaded by dbrooks (license 790) Tested
+ by: JimVanM ........
+
+2009-11-05 21:24 +0000 [r228192] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 228189 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r228189 |
+ jpeeler | 2009-11-05 15:23:06 -0600 (Thu, 05 Nov 2009) | 11 lines
+ Fix the fix for chanspy option o In 224178, I assumed the
+ uploaded patch was correct as it had received positive feedback.
+ The flags were being checked in the incorrect location. Upon
+ testing the fix this time it was also found that the flags from
+ the dialplan weren't being copied to the
+ chanspy_translation_helper. (closes issue #16167) Reported by:
+ marhbere ........
+
+2009-11-05 19:41 +0000 [r228147] David Brooks <dbrooks@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 228145 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r228145 | dbrooks | 2009-11-05 13:34:50 -0600
+ (Thu, 05 Nov 2009) | 16 lines Merged revisions 228078 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05 Nov 2009)
+ | 9 lines chan_misdn Asterisk 1.4.27-rc2 crash Crash related to
+ chan_misdn connection. Patch submitted by gknispel_proformatique,
+ tested by francesco_r. "I have many crash since i have upgraded
+ to Asterisk 1.4.27-rc2. Attached a full bt." This patch zeros out
+ an ast_frame. (closes issue #16041) Reported by: francesco_r
+ ........ ................
+
+2009-11-05 19:19 +0000 [r228090] Jason Parker <jparker@digium.com>
+
+ * channels/chan_vpb.cc, /: Merged revisions 228080 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r228080 | qwell | 2009-11-05 13:16:29 -0600
+ (Thu, 05 Nov 2009) | 15 lines Merged revisions 228079 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228079 | qwell | 2009-11-05 13:14:25 -0600 (Thu, 05 Nov 2009) |
+ 8 lines Fix crash on VPB exception when no hardware is present.
+ (closes issue #14970) Reported by: tzafrir Patches:
+ vpb_exception.diff uploaded by tzafrir (license 46) Tested by:
+ markwaters ........ ................
+
+2009-11-05 17:10 +0000 [r228016] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_externalivr.c: Merged revisions 228015 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r228015 | tilghman | 2009-11-05 11:08:02 -0600 (Thu, 05 Nov 2009)
+ | 4 lines Don't crash if no arguments are passed. (closes issue
+ #16119) Reported by: thedavidfactor ........
+
+2009-11-04 23:56 +0000 [r227948] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 227945 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r227945 | jpeeler | 2009-11-04 17:50:59 -0600 (Wed, 04 Nov 2009)
+ | 21 lines Merged revisions 227944 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227944 | jpeeler | 2009-11-04 17:47:08 -0600 (Wed, 04 Nov 2009)
+ | 14 lines Fix incorrect filename comparsion after monitor file
+ change The logic to detect if a requested file is indeed a
+ different file from the current file was incorrect. The main
+ issue being confusion of the use of filename_base which was
+ previously set without pathing information and then compared to
+ another full path. Robust file comparison logic has been added to
+ properly check if two files are the same even if symlinks are
+ used. (closes issue #15313) Reported by: caspy Patches:
+ 20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license
+ 325) but mostly tilghman's work ........ ................
+
+2009-11-04 21:15 +0000 [r227761-227832] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 227829 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r227829 | mnicholson | 2009-11-04 15:03:33 -0600 (Wed, 04 Nov
+ 2009) | 17 lines Merged revisions 227827 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov
+ 2009) | 10 lines This patch modifies the Dial application to
+ monitor the calling channel for hangups while playing back
+ announcements. (closes issue #16005) Reported by: falves11
+ Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson
+ (license 96) Tested by: mnicholson, falves11 Review:
+ https://reviewboard.asterisk.org/r/407/ ........ ................
+
+ * 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
+
+2009-11-04 19:27 +0000 [r227723-227745] Joshua Colp <jcolp@digium.com>
+
+ * /, static-http/prototype.js: Merged revisions 227739 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r227739 | file | 2009-11-04 15:26:19 -0400 (Wed,
+ 04 Nov 2009) | 12 lines Merged revisions 227735 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227735 | file | 2009-11-04 15:25:37 -0400 (Wed, 04 Nov 2009) | 5
+ lines Fix a security issue where it may be possible for someone
+ to execute a cross-site AJAX request exploit. (AST-2009-009)
+ ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 227712 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r227712 | file | 2009-11-04 15:20:46 -0400 (Wed, 04 Nov 2009) |
+ 12 lines Merged revisions 227700 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227700 | file | 2009-11-04 15:17:39 -0400 (Wed, 04 Nov 2009) | 5
+ lines Fix a security issue where sending a REGISTER with a
+ differing username in the From URI and Authorization header would
+ reveal whether it was valid or not. (AST-2009-008) ........
+ ................
+
+2009-11-03 20:01 +0000 [r227374] Jason Parker <jparker@digium.com>
+
+ * Makefile, /, main/Makefile: Merged revisions 227372 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r227372 | qwell | 2009-11-03 13:59:46 -0600 (Tue, 03 Nov 2009) |
+ 9 lines Fix some build issues on Solaris. (closes issue #14517)
+ (SWP-109) Reported by: asgaroth Patches: bug_14517.diff uploaded
+ by snuffy (license 35) Tested by: asgaroth, snuffy, dougm, qwell
+ ........
+
+2009-11-03 19:49 +0000 [r227363-227370] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_controlplayback.c, /: Merged revisions 227368 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r227368 | lmadsen | 2009-11-03 13:48:53 -0600 (Tue, 03
+ Nov 2009) | 8 lines Change warning message to debug message.
+ app_controlplayback outputs a warning, when in fact it is normal.
+ (closes issue #16071) Reported by: atis Patches:
+ controlplayback_warning.patch uploaded by atis (license 242)
+ ........
+
+ * configs/extensions.conf.sample, /: Merged revisions 227361 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r227361 | lmadsen | 2009-11-03 13:25:18 -0600 (Tue, 03
+ Nov 2009) | 11 lines Additional fixes to the
+ extensions.conf.sample file. Update the extensions.conf.sample
+ [stdexten] context so that we use the variable instead of
+ requiring it to be passed explicitly. Also updated uses of the
+ [stdexten] context throughout. (closes issue #15858) Reported by:
+ pprindeville Patches: stdexten-context-update.txt uploaded by
+ lmadsen (license 10) Tested by: pprindeville ........
+
+2009-11-03 18:11 +0000 [r227279] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Merged revisions 227275 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227275 | rmudgett | 2009-11-03 11:55:47 -0600 (Tue, 03 Nov 2009)
+ | 4 lines Make sure the outgoing flag is cleared if a new channel
+ fails to get created for outgoing calls. This is the relevant
+ portion of asterisk/trunk -r226648 ........
+
+2009-11-03 15:38 +0000 [r227169] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 227167 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r227167 | file | 2009-11-03 11:37:08 -0400 (Tue, 03 Nov 2009) |
+ 12 lines Merged revisions 227166 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227166 | file | 2009-11-03 11:36:16 -0400 (Tue, 03 Nov 2009) | 5
+ lines Fix a bug where an RPID header could be generated with a
+ blank username in the URI. (closes issue #15909) Reported by:
+ kobaz ........ ................
+
+2009-11-03 15:24 +0000 [r227164] Leif Madsen <lmadsen@digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 227162 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r227162 | lmadsen | 2009-11-03 09:19:47 -0600 (Tue, 03
+ Nov 2009) | 7 lines Update extensions.conf.sample file to fix
+ incorrect extensions. (closes issue #15857) Reported by:
+ pprindeville Patches: stdexten.patch#2 uploaded by pprindeville
+ (license 347) Tested by: pprindeville ........
+
+2009-11-03 13:32 +0000 [r227155] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 227091 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r227091 | oej | 2009-11-03 12:11:15 +0100 (Tis, 03 Nov 2009) | 15
+ lines Merged revisions 227088 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7
+ lines Use proper response code when violating Contact ACL's.
+ https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a
+ quick review. (EDVX-003) ........ ................
+
+2009-11-02 21:05 +0000 [r226977] 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 18:11 +0000 [r226892] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 226890 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r226890 | file | 2009-11-02 14:08:54 -0400 (Mon, 02 Nov 2009) |
+ 18 lines Merged revisions 226889 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) |
+ 11 lines Fix a bug where the recorded privacy introduction file
+ would not get removed if the caller hung up while the called
+ party had not yet answered. This was fixed by introducing an
+ argument to the 'n' option which, when enabled, removes the
+ introduction file under all scenarios. This was done to preserve
+ the behavior that has existed for quite some time. (closes issue
+ #14674) Reported by: ulogic Patches: bug14674.patch uploaded by
+ jpeeler (license 325) ........ ................
+
+2009-11-02 17:17 +0000 [r226814] Tilghman Lesher <tlesher@digium.com>
+
+ * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 226812 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r226812 | tilghman | 2009-11-02 11:15:31 -0600
+ (Mon, 02 Nov 2009) | 15 lines Merged revisions 226811 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226811 | tilghman | 2009-11-02 11:14:20 -0600 (Mon, 02 Nov 2009)
+ | 8 lines Don't allow two separate instances of safe_asterisk
+ when restarting from the init script. (closes issue #14562)
+ Reported by: davidw Patches: Initially
+ 20091022__issue14562.diff.txt uploaded by tilghman (license 14)
+ Modified to 20091030__Issue14562_diff.txt uploaded by davidw
+ (license 780) Tested by: davidw ........ ................
+
+2009-10-29 18:15 +0000 [r226534] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_local.c, /, doc/tex/localchannel.tex: Merged
+ revisions 226532 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r226532 | file | 2009-10-29 15:13:42 -0300 (Thu, 29 Oct 2009) |
+ 13 lines Merged revisions 226531 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6
+ lines Add an option to enabling passing music on hold start and
+ stop requests through instead of acting on them in chan_local.
+ (closes issue #14709) Reported by: dimas ........
+ ................
+
+2009-10-28 20:15 +0000 [r226380-226386] Leif Madsen <lmadsen@digium.com>
+
+ * configs/sip.conf.sample: Merged revisions 226384 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r226384 | lmadsen | 2009-10-28 15:11:07 -0500
+ (Wed, 28 Oct 2009) | 17 lines Merged revisions 226382 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009)
+ | 9 lines Update documentation in sip.conf.sample. Update the
+ documentation in sip.conf.sample in order to make it more clear
+ that directmedia/canreinvite do not cause Asterisk to ignore
+ reINVITEs. It is only used to stop Asterisk from generating a
+ reINVITE, but does not stop it from accepting them if necessary.
+ (closes issue #15644) Reported by: lmadsen ........
+ ................
+
+ * /, doc/tex/channelvariables.tex: Merged revisions 226378 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r226378 | lmadsen | 2009-10-28 14:50:00 -0500
+ (Wed, 28 Oct 2009) | 15 lines Merged revisions 226377 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009)
+ | 7 lines Update CALLINGSUBADDR channel variable documentation.
+ (closes issue #15734) Reported by: alecdavis Patches:
+ channelvariables.tex.diff.txt uploaded by alecdavis (license 585)
+ Tested by: alecdavis ........ ................
+
+2009-10-28 18:05 +0000 [r226169-226307] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/linkedlists.h: Merged revisions 226305 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r226305 | tilghman | 2009-10-28 13:04:05 -0500
+ (Wed, 28 Oct 2009) | 9 lines Merged revisions 226304 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28
+ Oct 2009) | 2 lines Fix documentation (pointed out by
+ TheDavidFactor on #-dev) ........ ................
+
+ * main/manager.c, /: Merged revisions 226159 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r226159 | tilghman | 2009-10-27 15:22:07 -0500 (Tue, 27 Oct 2009)
+ | 14 lines Merged revisions 226138 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009)
+ | 7 lines Manager output is not always NULL-terminated, so force
+ a NULL at the end of the filestream. (closes issue #15495)
+ Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded
+ by tilghman (license 14) Tested by: pdf ........ ................
+
+2009-10-27 17:04 +0000 [r226100] Terry Wilson <twilson@digium.com>
+
+ * /, res/res_http_post.c: Merged revisions 226099 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r226099 |
+ twilson | 2009-10-27 11:48:54 -0500 (Tue, 27 Oct 2009) | 2 lines
+ Don't prepend the URI prefix to the post directory ........
+
+2009-10-26 23:48 +0000 [r226053] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * /, configure, configure.ac: detect ARM Linux EABI OSARCH as
+ linux-gnu instead of linux-gnueabi * Set OSARCH to linux-gnu even
+ if host_os is linux-gnueabi * When checking if we are Linux,
+ check OSARCH rather than host_os The newer ARM ABI ("EABI") shows
+ the OS name 'linux-gnueabi' rather than 'linux-gnu' . This patch
+ sets OSARCH to be 'linux-gnu' even in such a case. OSARCH is
+ tested for the value of 'linux-gnu' in one or two places in the
+ tree. This patch also fixes the check libcap to check for $OSARCH
+ rather than $host_os . See also:
+ http://wiki.debian.org/ArmEabiPort Merged revisions 225957 via
+ svnmerge from http://svn.digium.com/svn/asterisk/branches/1.4
+ Merged revisions 226018 via svnmerge from
+ http://svn.digium.com/svn/asterisk/trunk
+
+2009-10-26 19:41 +0000 [r225913] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 225912 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r225912 |
+ jpeeler | 2009-10-26 14:40:26 -0500 (Mon, 26 Oct 2009) | 12 lines
+ ACL check not present for verifying SIP INVITEs The ACL check in
+ check_peer_ok was missing and has now been restored. The missing
+ check allowed for calls to be made on prohibited networks where
+ an ACL was defined in sip.conf and the allowguest option was set
+ to off. See the AST security advisory below for more information.
+ Merge code associated with AST-2009-007. (closes issue #16091)
+ Reported by: thom4fun ........
+
+2009-10-26 15:51 +0000 [r225870] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_fax.c: Backport audio handling loop fixes from trunk
+ version of app_fax. This backport resolves some issues handling
+ audio frames during FAX processing, and ensures that the FAX
+ application doesn't accidentally get notified of a T.38
+ switchover at the end of a successful FAX. (issue #16127)
+
+2009-10-23 14:50 +0000 [r225652] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 225650 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r225650 |
+ dvossel | 2009-10-23 09:41:50 -0500 (Fri, 23 Oct 2009) | 3 lines
+ Fixes an iterator memory leak and uninitialized memory ........
+
+2009-10-23 14:07 +0000 [r225584] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, /: Merged revisions 225582 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r225582 | kpfleming | 2009-10-23 09:02:42 -0500 (Fri, 23 Oct
+ 2009) | 17 lines Merged revisions 225581 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct
+ 2009) | 10 lines Don't force menuselect.makeopts to be rebuilt on
+ every build. For some reason the menuselect.makeopts file was
+ listed as PHONY in the Makefile, resulting in 'make' needing to
+ rebuild it for every build. This then resulted in the embedded
+ module rules being rebuilt on every build, which can be slow and
+ is unnecessary. This patch fixes the problem by properly allowing
+ 'make' to know when the menuselect.makeopts file needs to be
+ rebuilt (defining the proper dependencies). ........
+ ................
+
+2009-10-22 22:07 +0000 [r225490] David Vossel <dvossel@digium.com>
+
+ * main/tcptls.c, /, channels/chan_sip.c, apps/app_externalivr.c,
+ include/asterisk/tcptls.h: Merged revisions 225445 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r225445 | dvossel | 2009-10-22 14:55:51 -0500 (Thu, 22 Oct 2009)
+ | 50 lines SIP TCP/TLS: move client connection setup/write into
+ tcp helper thread, various related locking/memory fixes. What
+ this patch fixes 1.Moves sip TCP/TLS connection setup into the
+ TCP helper thread: Connection setup takes awhile and before this
+ it was being done while holding the monitor lock. 2.Moves TCP/TLS
+ writing to the TCP helper thread: Through the use of a packet
+ queue and an alert pipe, the TCP helper thread can now be woken
+ up to write data as well as read data. 3.Locking error: sip_xmit
+ returned an XMIT_ERROR without giving up the tcptls_session lock.
+ This lock has been completely removed from sip_xmit and placed in
+ the new sip_tcptls_write() function. 4.Memory leak: When creating
+ a tcptls_client the tls_cfg was alloced but never freed unless
+ the tcptls_session failed to start. Now the session_args for a
+ sip client are an ao2 object which frees the tls_cfg on
+ destruction. 5.Pointer to stack variable: During
+ sip_prepare_socket the creation of a client's
+ ast_tcptls_session_args was done on the stack and stored as a
+ pointer in the newly created tcptls_session. Depending on the
+ events that followed, there was a slight possibility that pointer
+ could have been accessed after the stack returned. Given the new
+ changes, it is always accessed after the stack returns which is
+ why I found it. Notable code changes 1.I broke tcptls.c's
+ ast_tcptls_client_start() function into two functions. One for
+ creating and allocating the new tcptls_session, and a separate
+ one for starting and handling the new connection. This allowed me
+ to create the tcptls_session, launch the helper thread, and then
+ establish the connection within the helper thread. 2.Writes to a
+ tcptls_session are now done within the helper thread. This is
+ done by using an alert pipe to wake up the thread if new data
+ needs to be sent. The thread's sip_threadinfo object contains the
+ alert pipe as well as the packet queue. 3.Since the threadinfo
+ object contains the alert pipe, it must now be accessed outside
+ of the helper thread for every write (queuing of a packet). For
+ easy lookup, I moved the threadinfo objects from a linked list to
+ an ao2_container. (closes issue #13136) Reported by: pabelanger
+ Tested by: dvossel, whys (closes issue #15894) Reported by:
+ dvossel Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/380/ ........
+
+2009-10-22 21:54 +0000 [r225487] Leif Madsen <lmadsen@digium.com>
+
+ * doc/valgrind.txt, contrib/valgrind.supp (added): Merged revisions
+ 225485 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r225485 | lmadsen | 2009-10-22 16:52:30 -0500 (Thu, 22 Oct 2009)
+ | 19 lines Merged revisions 225484 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009)
+ | 11 lines Clean valgrind output by suppressing false errors.
+ Update valgrind.txt documentation and add valgrind.supp file in
+ order to allow those who are creating valgrind output to have
+ less false errors in the logfile. (closes issue #16007) Reported
+ by: atis Patches: valgrind.txt.diff uploaded by atis (license
+ 242) asterisk2.supp uploaded by atis (license 242) Tested by:
+ atis, amorsen ........ ................
+
+2009-10-22 17:14 +0000 [r225362] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /, apps/app_meetme.c, include/asterisk/channel.h:
+ Merged revisions 225360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r225360 | tilghman | 2009-10-22 12:11:23 -0500 (Thu, 22 Oct 2009)
+ | 11 lines Merged revisions 225105 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009)
+ | 4 lines Fix documentation for ast_softhangup() and correct the
+ misuse thereof. (closes issue #16103) Reported by: majorbloodnok
+ ........ ................
+
+2009-10-21 22:02 +0000 [r225062-225309] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 225307 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r225307 | dvossel | 2009-10-21 16:58:46 -0500
+ (Wed, 21 Oct 2009) | 20 lines Merged revisions 225243 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009)
+ | 13 lines IAX2: VNAK loop caused by signaling frames with no
+ destination call number It is possible for the PBX thread to
+ queue up signaling frames before a destination call number is
+ received. This can result in signaling frames being sent out with
+ no destination call number. Since recent versions of Asterisk
+ require accurate destination callnumbers for all Full Frames,
+ this can cause a VNAK loop to occur. To resolve this no signaling
+ frames are sent until a destination callnumber is received, and
+ destination call numbers are now only required for iax_pvt
+ matching when the frame is an ACK. Review:
+ https://reviewboard.asterisk.org/r/413/ ........ ................
+
+ * channels/chan_iax2.c, configs/iax.conf.sample, /,
+ channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+ 225033 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r225033 | dvossel | 2009-10-21 09:39:10 -0500 (Wed, 21 Oct 2009)
+ | 27 lines Merged revisions 225032 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009)
+ | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller
+ id removes '(', ' ', ')', non-trailing '.', and '-' from the
+ string. This means values such as 555.5555 and test-test result
+ in 555555 and testtest. There are instances, such as Skype
+ integration, where a specific value is passed via caller id that
+ must be preserved unmodified. This patch makes the shrinking of
+ caller id optional in chan_sip and chan_iax in order to support
+ such cases. By default this option is on to preserve previous
+ expected behavior. (closes issue #15940) Reported by: dimas
+ Patches: v2-15940.patch uploaded by dimas (license 88)
+ 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671)
+ Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/408/ ........ ................
+
+2009-10-21 03:17 +0000 [r224935] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/frame.h, include/asterisk/translate.h,
+ main/dsp.c, main/frame.c, /, main/translate.c,
+ include/asterisk/dsp.h, codecs/codec_dahdi.c: Merged revisions
+ 224932 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r224932 | russell | 2009-10-20 22:09:04 -0500 (Tue, 20 Oct 2009)
+ | 12 lines Merged revisions 224931 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009)
+ | 5 lines Isolate frames returned from a DSP instance or codec
+ translator. The reasoning for these changes are the same as what
+ I wrote in the commit message for rev 222878. ........
+ ................
+
+2009-10-20 22:11 +0000 [r224858] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_speex.c, /, main/audiohook.c: Merged revisions 224856
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r224856 | tilghman | 2009-10-20 17:09:07 -0500
+ (Tue, 20 Oct 2009) | 12 lines Merged revisions 224855 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009)
+ | 5 lines Pay attention to the return value of the manipulate
+ function. While this looks like an optimization, it prevents a
+ crash from occurring when used with certain audiohook callbacks
+ (diagnosed with SVN trunk, backported to 1.4 to keep the source
+ consistent across versions). ........ ................
+
+2009-10-20 17:49 +0000 [r224776] Joshua Colp <jcolp@digium.com>
+
+ * /, main/features.c: Merged revisions 224774 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r224774 | file | 2009-10-20 14:47:34 -0300 (Tue, 20 Oct 2009) |
+ 12 lines Merged revisions 224773 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5
+ lines Add support for relaying early media in the features
+ attended transfer option. (closes issue #14828) Reported by:
+ licedey ........ ................
+
+2009-10-19 23:56 +0000 [r224673] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/rtp.c: Merged revisions 224671 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r224671 | kpfleming | 2009-10-19 18:47:39 -0500 (Mon, 19 Oct
+ 2009) | 14 lines Merged revisions 224670 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 Oct
+ 2009) | 7 lines Correct timestamp calculations when RTP sample
+ rates over 8kHz are used. While testing some endpoints that
+ support 16kHz and 32kHz sample rates, some log messages were
+ generated due to calc_rxstamp() computing timestamps in a way
+ that produced odd results, so this patch sanitizes the result of
+ the computations. ........ ................
+
+2009-10-19 19:51 +0000 [r224570] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 224567 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r224567 | file | 2009-10-19 16:49:09 -0300 (Mon, 19 Oct 2009) |
+ 12 lines Merged revisions 224565 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5
+ lines Do not attempt early media bridging (ie: direct RTP setup)
+ if options are enabled that should prevent it. (closes issue
+ #14763) Reported by: cupotka ........ ................
+
+2009-10-17 02:02 +0000 [r224333-224336] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: fix typo, sorry
+
+ * channels/chan_dahdi.c, /: Merged revisions 224331 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r224331 | jpeeler | 2009-10-16 20:36:08 -0500
+ (Fri, 16 Oct 2009) | 20 lines Merged revisions 224330 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009)
+ | 13 lines Fix stale caller id data from being reported in AMI
+ NewChannel event The problem here is that chan_dahdi is designed
+ in such a way to set certain values in the dahdi_pvt only once.
+ One of those such values is the configured caller id data in
+ chan_dahdi.conf. For PRI, the configured caller id data could be
+ overwritten during a call. Instead of saving the data and
+ restoring, it was decided that for all non-analog channels it was
+ simply best to not set the configured caller id in the first
+ place and also clear it at the end of the call. (closes issue
+ #15883) Reported by: jsmith ........ ................
+
+2009-10-16 20:53 +0000 [r224263] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 224261 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r224261 | rmudgett | 2009-10-16 15:40:57 -0500
+ (Fri, 16 Oct 2009) | 25 lines Merged revisions 224260 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009)
+ | 18 lines Never released PRI channels when using Busy() or
+ Congestion() dialplan apps. When the Busy() or Congestion()
+ application is used towards ISDN (an ISDN progress is sent), the
+ responding ISDN Disconnect or Release may contain the ISDN cause
+ user busy or one of the congestion causes. In chan_dahdi.c these
+ causes will only set the needbusy or needcongestion flags and not
+ activate the softhangup procedure. Unfortunately only the latter
+ can interrupt the endless wait loop of Busy()/Congestion().
+ Result: PRI channels staying in state busy for the rest of
+ asterisk life or until the other end times out and forces the
+ call to clear. (in issue 0014292) Reported by: tomaso Patches:
+ disc_rel_userbusy.patch uploaded by tomaso (license 564) (This
+ patch is unrelated to the issue.) ........ ................
+
+2009-10-15 15:58 +0000 [r224180] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 224178 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r224178 |
+ jpeeler | 2009-10-15 10:57:14 -0500 (Thu, 15 Oct 2009) | 11 lines
+ Readd removed ability to allow listening to one side of the call
+ in app_chanspy (Option o) (closes issue #15675) Reported by:
+ john8675309 Patches: issue15675patchtrunk.txt uploaded by dbrooks
+ (license 790) Tested by: jgutierrez on users list:
+ http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html
+ ........
+
+2009-10-12 23:55 +0000 [r223834] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 223832 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r223832 | jpeeler | 2009-10-12 18:48:09 -0500 (Mon, 12 Oct 2009)
+ | 15 lines Merged revisions 223804 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009)
+ | 8 lines Ensure ringing continues for branched calls after
+ progress is received While waiting for an answer, don't send
+ progress for branched calls for which ringing was sent. (closes
+ issue #15028) Reported by: fnordian ........ ................
+
+2009-10-12 21:03 +0000 [r223758] David Vossel <dvossel@digium.com>
+
+ * configs/iax.conf.sample, /: Merged revisions 223756 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r223756 | dvossel | 2009-10-12 15:58:27 -0500 (Mon, 12 Oct 2009)
+ | 5 lines Clarifies trunkmaxsize, trunkfreq, and trunkmtu iax2
+ options SWP-151 ........
+
+2009-10-12 14:32 +0000 [r223654] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_sip.c, apps/app_fax.c: Merged revisions 223652
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r223652 | kpfleming | 2009-10-12 09:25:29 -0500 (Mon, 12
+ Oct 2009) | 13 lines Remove automatic switching from T.38 to
+ voice mode in chan_sip. chan_sip has some code to automatically
+ switch from T.38 mode to voice mode when a voice frame is written
+ to the channel while it is in T.38 mode; this was intended to
+ handle the situation when a FAX transmission has ended and the
+ channel is not yet hung up, but is causing problems at the
+ beginning of FAX sessions as well when there are still voice
+ frames 'in flight' at the time the T.38 negotiation completes.
+ This patch removes the automatic switchover, and changes app_fax
+ to explicitly switch off T.38 mode when the FAX transmission
+ process ends. (closes issue #16025) Reported by: jamicque
+ ........
+
+2009-10-11 17:31 +0000 [r223489] Russell Bryant <russell@digium.com>
+
+ * main/autoservice.c, /: Merged revisions 223487 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r223487 | russell | 2009-10-11 12:25:42 -0500 (Sun, 11 Oct 2009)
+ | 17 lines Merged revisions 223485-223486 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223485 | russell | 2009-10-11 12:22:52 -0500 (Sun, 11 Oct 2009)
+ | 6 lines Don't use data outside of its scope. The purpose of
+ this code was to have a hangup frame put on the list of deferred
+ frames. However, the code that read the hangup frame was outside
+ of the scope of where the hangup frame was declared. ........
+ r223486 | russell | 2009-10-11 12:25:06 -0500 (Sun, 11 Oct 2009)
+ | 2 lines Remove some unnecessary code. ........ ................
+
+2009-10-09 23:11 +0000 [r223405] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, channels/chan_h323.c: Fix interpretation
+ of PRIREDIRECTIONREASON set by chan_sip. This commit is the
+ simplest way to solve a problem that has already been solved in
+ trunk with the "COLP/CONP and Redirecting party information into
+ Asterisk" commit. In trunk the redirection reason is translated
+ into a generic redirect reason. I would have had to do the same
+ fix except chan_sip never reads PRIREDIRECTREASON. So both
+ chan_dahdi and chan_h323 have been modified to interpret the one
+ different redirect reason of "no-answer" properly and set the
+ ISDN reason code 2 of "no reply". (closes issue #15033) Reported
+ by: steinwej
+
+2009-10-09 21:00 +0000 [r223332] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 223330 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r223330 |
+ kpfleming | 2009-10-09 15:58:44 -0500 (Fri, 09 Oct 2009) | 10
+ lines Initiate T.38 switchover when acting as called party,
+ regardless of FAX direction. SendFAX() and ReceiveFAX() can be
+ given options to indicate whether they should act as the calling
+ or called party; this mode should be used to decide whether to
+ initiate a switchover to T.38, not the direction that the FAX
+ transfer will take place. (closes issue #16039) Reported by:
+ jamicque ........
+
+2009-10-09 18:36 +0000 [r223277] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, /: Merged revisions 223273 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r223273 | mnicholson | 2009-10-09 13:34:08 -0500 (Fri, 09 Oct
+ 2009) | 14 lines Merged revisions 223225 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct
+ 2009) | 8 lines Signal timeouts by returning AST_CONTROL_RINGING
+ when originating calls. (closes issue #15104) Reported by:
+ nblasgen Patches: manager-timeout1.diff uploaded by mnicholson
+ (license 96) Tested by: nblasgen, mnicholson ........
+ ................
+
+2009-10-09 18:25 +0000 [r223241] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 223215 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r223215 | mmichelson | 2009-10-09 13:17:34 -0500 (Fri, 09 Oct
+ 2009) | 9 lines Recorded merge of revisions 223213 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri,
+ 09 Oct 2009) | 3 lines Fix potential memory leak in app_dial.c
+ ........ ................
+
+2009-10-09 17:56 +0000 [r223209] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 223206 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r223206 | dvossel | 2009-10-09 12:53:37 -0500 (Fri, 09 Oct 2009)
+ | 16 lines Merged revisions 223205 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009)
+ | 10 lines fixes sip registration using authuser in user.conf
+ (closes issue #14954) Reported by: tornblad Tested by:
+ mmichelson, tornblad, dvossel ........ ................
+
+2009-10-09 17:27 +0000 [r223171] Matthew Nicholson <mnicholson@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 223136 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r223136 | mnicholson | 2009-10-09 12:14:38 -0500 (Fri, 09 Oct
+ 2009) | 8 lines Don't close the sqlite database when reloading.
+ Only close the database when unloading. (closes issue #15953)
+ Reported by: frawd Patches: sqlite3_rev220097.diff uploaded by
+ frawd (license 610) Tested by: frawd ........
+
+2009-10-09 17:10 +0000 [r223090-223134] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 223132 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r223132 |
+ dvossel | 2009-10-09 11:54:02 -0500 (Fri, 09 Oct 2009) | 9 lines
+ 'auth=' did not parse md5 secret correctly (closes issue #15949)
+ Reported by: ebroad Patches: authparsefix.patch uploaded by
+ ebroad (license 878) 15949_trunk.diff uploaded by dvossel
+ (license 671) Tested by: ebroad ........
+
+ * /, channels/chan_sip.c: Merged revisions 223088 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r223088 |
+ dvossel | 2009-10-09 10:49:30 -0500 (Fri, 09 Oct 2009) | 14 lines
+ p->peerauth is always empty in transmit_register() When using
+ callbackextension or specifing the peer name in a registration
+ string, the peer's specific auth settings set by the "auth="
+ strings within the peer definition are not used by the
+ registration. Thanks to ebroad for reporting the issue and
+ providing the patch. (closes issue #15955) Reported by: ebroad
+ Patches: regauthfix.patch uploaded by ebroad (license 878)
+ ........
+
+2009-10-08 19:57 +0000 [r222882] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/frame.h, include/asterisk/file.h, main/frame.c,
+ /, main/file.c: Merged revisions 222880 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r222880 | russell | 2009-10-08 14:52:03 -0500 (Thu, 08 Oct 2009)
+ | 51 lines Merged revisions 222878 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009)
+ | 44 lines Make filestream frame handling safer by isolating
+ frames before returning them. This patch is related to a number
+ of issues on the bug tracker that show crashes related to freeing
+ frames that came from a filestream. A number of fixes have been
+ made over time while trying to figure out these problems, but
+ there re still people seeing the crash. (Note that some of these
+ bug reports include information about other problems. I am
+ specifically addressing the filestream frame crash here.) I'm
+ still not clear on what the exact problem is. However, what is
+ _very_ clear is that we have seen quite a few problems over time
+ related to unexpected behavior when we try to use embedded frames
+ as an optimization. In some cases, this optimization doesn't
+ really provide much due to improvements made in other areas. In
+ this case, the patch modifies filestream handling such that the
+ embedded frame will not be returned. ast_frisolate() is used to
+ ensure that we end up with a completely mallocd frame. In
+ reality, though, we will not actually have to malloc every time.
+ For filestreams, the frame will almost always be allocated and
+ freed in the same thread. That means that the thread local frame
+ cache will be used. So, going this route doesn't hurt. With this
+ patch in place, some people have reported success in not seeing
+ the crash anymore. (SWP-150) (AST-208) (ABE-1834) (issue #15609)
+ Reported by: aragon Patches: filestream_frisolate-1.4.diff2.txt
+ uploaded by russell (license 2) Tested by: aragon, russell
+ (closes issue #15817) Reported by: zerohalo Tested by: zerohalo
+ (closes issue #15845) Reported by: marhbere Review:
+ https://reviewboard.asterisk.org/r/386/ ........ ................
+
+2009-10-08 19:42 +0000 [r222875] David Vossel <dvossel@digium.com>
+
+ * main/netsock.c, /, include/asterisk/netsock.h: Merged revisions
+ 222873 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r222873 |
+ dvossel | 2009-10-08 14:35:30 -0500 (Thu, 08 Oct 2009) | 6 lines
+ fixes an ast_netsock_list memory leak. ABE-1998 Review:
+ https://reviewboard.asterisk.org/r/395/ ........
+
+2009-10-08 16:49 +0000 [r222694-222801] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn_config.c, /: Merged revisions 222799 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r222799 | rmudgett | 2009-10-08 11:44:33 -0500
+ (Thu, 08 Oct 2009) | 19 lines Merged revisions 222797 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r222797 | rmudgett | 2009-10-08 11:33:06 -0500 (Thu, 08 Oct 2009)
+ | 12 lines Fix memory leak if chan_misdn config parameter is
+ repeated. Memory leak when the same config option is set more
+ than once in an misdn.conf section. Why must this be considered?
+ Templates! Defining a template with default port options and
+ later adding to or overriding some of them. Patches:
+ memleak-misdn.patch JIRA ABE-1998 ........ ................
+
+ * channels/chan_misdn.c, /: Merged revisions 222692 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r222692 | rmudgett | 2009-10-07 16:56:36 -0500
+ (Wed, 07 Oct 2009) | 21 lines Merged revisions 222691 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r222691 | rmudgett | 2009-10-07 16:51:24 -0500 (Wed, 07 Oct 2009)
+ | 14 lines chan_misdn.c:process_ast_dsp() memory leak misdn.conf:
+ astdtmf must be set to "yes". With "no", buffer loss does not
+ occur. The translated frame "f2" when passing through
+ ast_dsp_process() is not freed whenever it is not used further in
+ process_ast_dsp(). Then in the end it is never ever freed.
+ Patches: translate.patch JIRA ABE-1993 ........ ................
+
+2009-10-07 17:46 +0000 [r222545] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 222543 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r222543 | dvossel | 2009-10-07 12:44:52 -0500 (Wed, 07 Oct 2009)
+ | 14 lines Merged revisions 222542 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r222542 | dvossel | 2009-10-07 12:41:21 -0500 (Wed, 07 Oct 2009)
+ | 8 lines crash on transfer handle_invite_replaces() attempts to
+ uplock a pvt's owner channel without first verifing that it
+ exists. (issue #16027) ........ ................
+
+2009-10-06 23:58 +0000 [r222353-222465] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 222463 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r222463 | jpeeler | 2009-10-06 18:56:01 -0500
+ (Tue, 06 Oct 2009) | 14 lines Merged revisions 222462 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r222462 | jpeeler | 2009-10-06 18:51:19 -0500 (Tue, 06 Oct 2009)
+ | 8 lines Add missing unlock(s) in dahdi_read (two cases in
+ trunk) (closes issue #15683) Reported by: alecdavis ........
+ ................
+
+ * 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) Modified:
+ branches/1.4/channels/chan_dahdi.c
+
+ * channels/chan_dahdi.c, /: Merged revisions 222351 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r222351 | jpeeler | 2009-10-06 15:35:19 -0500 (Tue, 06 Oct 2009)
+ | 9 lines Fix 222298 (crash during destruction of second channel
+ when variable set with setvar). I mistakenly reasoned that setvar
+ would be used on all channels. Since it can be set per channel,
+ give each dahdi channel a copy of the variable. (related to
+ #15899) ........
+
+2009-10-06 19:34 +0000 [r222310] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_pgsql.c, /, cdr/cdr_pgsql.c: Recorded merge of
+ revisions 222309 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r222309 |
+ tilghman | 2009-10-06 14:31:39 -0500 (Tue, 06 Oct 2009) | 10
+ lines Change schema query to involve the use of an optional
+ schema parameter. This change is done in such a way as to allow
+ the driver to continue to function with older databases which
+ don't have these features. (closes issue #16000) Reported by:
+ jamicque Patches: 20091002__issue16000.diff.txt uploaded by
+ tilghman (license 14) 20091002__issue16000__1.6.1.diff.txt
+ uploaded by tilghman (license 14) Tested by: jamicque ........
+
+2009-10-06 19:26 +0000 [r222303] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 222298 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r222298 | jpeeler | 2009-10-06 14:24:59 -0500 (Tue, 06 Oct 2009)
+ | 9 lines Fix crash during destruction of second channel when
+ variable set with setvar. The setvar line in chan_dahdi.conf is
+ shared among all the channels, so make sure to only free the
+ resources only when the last channel is destroyed. (closes issue
+ #15899) Reported by: tzafrir ........
+
+2009-10-06 19:20 +0000 [r222282] Tilghman Lesher <tlesher@digium.com>
+
+ * res/ael/pval.c, /: Merged revisions 222273 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r222273 |
+ tilghman | 2009-10-06 14:17:11 -0500 (Tue, 06 Oct 2009) | 5 lines
+ When we call a gosub routine, the variables should be scoped to
+ avoid contaminating the caller. This affected the ~~EXTEN~~ hack,
+ where a subroutine might have changed the value before it was
+ used in the caller. Patch by myself, tested by ebroad on
+ #asterisk ........
+
+2009-11-04 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.1.9
+
+ * AST-2009-008 and AST-2009-009
+
+2009-10-26 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.1.8
+
+ * AST-2009-007
+
+2009-10-06 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.1.7-rc2
+
+2009-10-06 01:36 +0000 [r222112-222186] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_queue.c, channels/chan_iax2.c, main/astobj2.c,
+ res/res_odbc.c, /, channels/chan_sip.c, funcs/func_dialgroup.c,
+ include/asterisk/astobj2.h, res/res_phoneprov.c,
+ channels/chan_console.c, res/res_musiconhold.c: Merged revisions
+ 222176 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct
+ 2009) | 27 lines Recorded merge of revisions 222152 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05
+ Oct 2009) | 20 lines Fix ao2_iterator API to hold references to
+ containers being iterated. See Mantis issue for details of what
+ prompted this change. Additional notes: This patch changes the
+ ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum
+ instead of a macro, with a name that fits our naming policy;
+ also, it is now necessary to call ao2_iterator_destroy() on any
+ iterator that has been created. Currently this only releases the
+ reference to the container being iterated, but in the future this
+ could also release other resources used by the iterator, if the
+ iterator implementation changes to use additional resources.
+ (closes issue #15987) Reported by: kpfleming Review:
+ https://reviewboard.asterisk.org/r/383/ ........ ................
+
+ * main/udptl.c, /, channels/chan_sip.c, configs/udptl.conf.sample,
+ UPGRADE.txt, configs/sip.conf.sample: Merged revisions 222110 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r222110 | kpfleming | 2009-10-05 14:45:00 -0500 (Mon, 05
+ Oct 2009) | 25 lines Allow non-compliant T.38 endpoints to be
+ supportable via configuration option. Many T.38 endpoints
+ incorrectly send the maximum IFP frame size they can accept as
+ the T38FaxMaxDatagram value in their SDP, when in fact this value
+ is supposed to be the maximum UDPTL payload size (datagram size)
+ they can accept. If the value they supply is small enough (a
+ commonly supplied value is '72'), T.38 UDPTL transmissions will
+ likely fail completely because the UDPTL packets will not have
+ enough room for a primary IFP frame and the redundancy used for
+ error correction. If this occurs, the Asterisk UDPTL stack will
+ emit log messages warning that data loss may occur, and that the
+ value may need to be overridden. This patch extends the
+ 't38pt_udptl' configuration option in sip.conf to allow the
+ administrator to override the value supplied by the remote
+ endpoint and supply a value that allows T.38 FAX transmissions to
+ be successful with that endpoint. In addition, in any SIP call
+ where the override takes effect, a debug message will be printed
+ to that effect. This patch also removes the T38FaxMaxDatagram
+ configuration option from udptl.conf.sample, since it has not
+ actually had any effect for a number of releases. In addition,
+ this patch cleans up the T.38 documentation in sip.conf.sample
+ (which incorrectly documented that T.38 support was passthrough
+ only). (issue #15586) Reported by: globalnetinc ........
+
+2009-10-02 17:36 +0000 [r222035] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 222030 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r222030 | dvossel | 2009-10-02 12:34:07 -0500
+ (Fri, 02 Oct 2009) | 9 lines Merged revisions 222026 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r222026 | dvossel | 2009-10-02 12:32:13 -0500 (Fri, 02
+ Oct 2009) | 3 lines Removes unnecessary unlock, clarifies a
+ memcpy. ........ ................
+
+2009-10-02 17:01 +0000 [r221969-221973] Tilghman Lesher <tlesher@digium.com>
+
+ * main/astobj2.c, /, funcs/func_lock.c: Merged revisions 221971 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r221971 | tilghman | 2009-10-02 11:59:57 -0500
+ (Fri, 02 Oct 2009) | 9 lines Merged revisions 221970 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02
+ Oct 2009) | 2 lines Ensure the result of the hash function is
+ positive. Negative array offsets suck. ........ ................
+
+ * funcs/func_lock.c: Hash needs to return a positive integer
+
+2009-10-02 13:04 +0000 [r221964] Sean Bright <sean@malleable.com>
+
+ * funcs/func_strings.c: Revert XML docs that ended up in the 1.6.0
+ and 1.6.1 branches during a merge.
+
+2009-10-02 03:06 +0000 [r221922] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/logger.c: Merged revisions 221920 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221920 |
+ tilghman | 2009-10-01 22:04:34 -0500 (Thu, 01 Oct 2009) | 4 lines
+ Initialize a variable that we check immediately upon startup.
+ (closes issue #15973) Reported by: atis ........
+
+2009-10-02 01:26 +0000 [r221871] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /:
+ Merged revisions 221844 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221844 | rmudgett | 2009-10-01 20:09:31 -0500 (Thu, 01 Oct 2009)
+ | 33 lines Merged revisions 221769 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009)
+ | 26 lines Occasionally losing use of B channels in chan_misdn. I
+ have not been able to reproduce the problem of losing channels.
+ However, I have seen in the code a reentrancy problem that might
+ give these symptoms. The reentrancy patch does several things: 1)
+ Guards B channel and B channel structure allocation. 2) Makes the
+ B channel structure find routines more precise in locating
+ records. 3) Never leave a B channel allocated if we received
+ cause 44. The last item may cause temporary outgoing call
+ problems, but they should clear when the line becomes idle.
+ (closes issue #15490) Reported by: slutec18 Patches:
+ issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett
+ (license 664) Tested by: rmudgett, slutec18 (closes issue #15458)
+ Reported by: FabienToune Patches:
+ issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett
+ (license 664) Tested by: FabienToune, rmudgett, slutec18 ........
+ ................
+
+2009-10-02 00:06 +0000 [r221743-221779] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, main/rtp.c, /, main/say.c: Merged revisions
+ 221777 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221777 | tilghman | 2009-10-01 18:59:15 -0500 (Thu, 01 Oct 2009)
+ | 9 lines Merged revisions 221776 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009)
+ | 2 lines Fix a bunch of off-by-one errors ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 221705 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221705 |
+ tilghman | 2009-10-01 15:09:46 -0500 (Thu, 01 Oct 2009) | 2 lines
+ Revision 220906 (a merge from 1.4) was not merged correctly,
+ causing a problem with non-dynamic peers. ........
+
+2009-10-01 19:52 +0000 [r221702] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 221697 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221697 |
+ dvossel | 2009-10-01 14:33:33 -0500 (Thu, 01 Oct 2009) | 9 lines
+ outbound tls connections were not defaulting to port 5061 (closes
+ issue #15854) Reported by: dvossel Patches:
+ sip_port_config_trunk.diff uploaded by dvossel (license 671)
+ Tested by: dvossel ........
+
+2009-10-01 17:01 +0000 [r221661] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 221554,221589 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r221554 | oej | 2009-10-01 02:00:04 -0500 (Thu,
+ 01 Oct 2009) | 3 lines Simplify code for porturi, use TRUE/FALSE
+ constructs when it's just TRUE or FALSE. ................ r221589
+ | mnicholson | 2009-10-01 10:26:20 -0500 (Thu, 01 Oct 2009) | 9
+ lines Merged revisions 221588 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct
+ 2009) | 2 lines Use unsigned ints for portinuri flags. ........
+ ................
+
+2009-10-01 16:19 +0000 [r221602] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, /, configs/udptl.conf.sample, UPGRADE.txt: Merged
+ revisions 221592 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221592 |
+ kpfleming | 2009-10-01 11:16:09 -0500 (Thu, 01 Oct 2009) | 12
+ lines Remove ability to control T.38 FAX error correction from
+ udptl.conf. chan_sip has had the ability to control T.38 FAX
+ error correction mode on a per-peer (or global) basis for a
+ couple of releases now, which is where it should have been all
+ along. This patch removes the ability to configure it in
+ udptl.conf, but issues a warning if the user tries to do, telling
+ them to look at sip.conf.sample for how to configure it now. For
+ any SIP peers that are T.38 enabled in sip.conf, there is already
+ a default for FEC error correction even if the user does not
+ specify any mode, so this change will not turn off error
+ correction by default, it will have the same default value that
+ has been in the udptl.conf sample file. ........
+
+2009-09-30 23:10 +0000 [r221478-221487] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 221484 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221484 |
+ mnicholson | 2009-09-30 18:04:03 -0500 (Wed, 30 Sep 2009) | 2
+ lines Cleaned up merge from r221432 ........
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+ 221432 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep
+ 2009) | 17 lines Merged revisions 221360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep
+ 2009) | 10 lines Fix SRV lookup and Request-URI generation in
+ chan_sip. This patch adds a new field "portinuri" to the sip
+ dialog struct and the sip peer struct. That field is used during
+ RURI generation to determine if the port should be included in
+ the RURI. It is also used in some places to determine if an SRV
+ lookup should occur. (closes issue #14418) Reported by: klaus3000
+ Tested by: klaus3000, mnicholson Review:
+ https://reviewboard.asterisk.org/r/369/ ........ ................
+
+2009-09-30 21:41 +0000 [r221370-221470] Matthias Nick <mnick@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 221436 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221436 |
+ mnick | 2009-09-30 16:15:01 -0500 (Wed, 30 Sep 2009) | 2 lines
+ Prevents from division by zero ........
+
+ * configs/cdr_custom.conf.sample, /, funcs/func_strings.c: Merged
+ revisions 221368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221368 | mnick | 2009-09-30 14:42:36 -0500 (Wed, 30 Sep 2009) |
+ 23 lines Merged revisions 221153,221157,221303 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) |
+ 2 lines check bounds - prevents for buffer overflow ........
+ r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) |
+ 8 lines added a new dialplan function 'CSV_QUOTE' and changed the
+ cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr
+ Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by:
+ mnick ........ r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed,
+ 30 Sep 2009) | 2 lines changed the prototype definition of
+ csv_quote ........ ................
+
+2009-09-30 18:58 +0000 [r221302] Terry Wilson <twilson@digium.com>
+
+ * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h,
+ configs/sip.conf.sample: Merged revisions 221266 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r221266 | twilson | 2009-09-30 12:52:30 -0500
+ (Wed, 30 Sep 2009) | 32 lines Merged revisions 221086 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009)
+ | 25 lines Change the SSRC by default when our media stream
+ changes Be default, change SSRC when doing an audio stream
+ changes Asterisk doesn't honor marker bit when reinvited to
+ already-bridged RTP streams,resulting in far-end stack discarding
+ packets with "old" timestamps that areactually part of a new
+ stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is
+ a reinvite, unless the 'constantssrc' is set to true in sip.conf.
+ The original issue reported to Digium support detailed the
+ following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based
+ Application Server Call comes in fromITSP, Asterisk dials the app
+ server which sends a re-invite back toAsterisk--not to negotiate
+ to send media directly to the ITSP, but to indicatethat it's
+ changing the stream it's sending to Asterisk. The app
+ servergenerates a new SSRC, sequence numbers, timestamps, and
+ sets the marker bit on the new stream. Asterisk passes through
+ the teimstamp of the new stream, butdoes not reset the SSRC,
+ sequence numbers, or set the marker bit. When the timestamp on
+ the new stream is older than the timestamp on the originalstream,
+ the ITSP (which doesn't know there has been any change) discards
+ the newframes because it thinks they are too old. This patch
+ addresses this by changing the SSRC on a stream update unless
+ constantssrc=true is set in sip.conf. Review:
+ https://reviewboard.asterisk.org/r/374/ ........ ................
+
+2009-09-30 16:57 +0000 [r221203] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 221201 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221201 | tilghman | 2009-09-30 11:56:42 -0500 (Wed, 30 Sep 2009)
+ | 14 lines Merged revisions 221200 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009)
+ | 7 lines Avoid a potential NULL dereference. (closes issue
+ #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt
+ uploaded by tilghman (license 14) Tested by: kobaz ........
+ ................
+
+2009-09-30 14:55 +0000 [r221088] Sean Bright <sean@malleable.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 221085 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r221085 | seanbright | 2009-09-30 10:47:58 -0400 (Wed, 30 Sep
+ 2009) | 9 lines Clarify documentation for VoiceMailMain()'s a()
+ option. We require box numbers, not names as the documentation
+ implies. (issue #14740) Reported by: pj Patches:
+ __20090729-app_voicemail-documentation.patch uploaded by lmadsen
+ (license 10) Tested by: seanbright, lmadsen ........
+
+2009-09-30 04:41 +0000 [r220998-221046] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_lock.c: Recorded merge of revisions 221044 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r221044 | tilghman | 2009-09-29 23:32:36 -0500 (Tue, 29
+ Sep 2009) | 8 lines Allow locks to be inherited through a
+ masquerade without causing starvation. (closes issue #14859)
+ Reported by: atis Patches: 20090821__issue14859.diff.txt uploaded
+ by tilghman (license 14) 20090925__issue14859__1.6.1.diff.txt
+ uploaded by tilghman (license 14) Tested by: atis, tilghman
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 220906 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r220906 | tilghman | 2009-09-29 14:57:37 -0500 (Tue, 29 Sep 2009)
+ | 16 lines Merged revisions 220873 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009)
+ | 9 lines Reduce CPU usage related to building a peer merely for
+ devicestates. This fixes a 100% CPU problem in the SIP driver,
+ found by profiling the driver while the problem was occurring.
+ (closes issue #14309) Reported by: pkempgen Patches:
+ 20090924__issue14309.diff.txt uploaded by tilghman (license 14)
+ Tested by: pkempgen, vrban ........ ................
+
+2009-09-29 20:25 +0000 [r220938] 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 17:05 +0000 [r220835] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 220833 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r220833 | jpeeler | 2009-09-29 11:58:29 -0500 (Tue, 29 Sep 2009)
+ | 12 lines Make deletion of temporary greetings work properly
+ with IMAP_STORAGE When imapgreetings was set to yes, the message
+ was being deleted but wasn't actually being expunged. When
+ imapgreetings was set to no, the file based message was not being
+ deleted at all. All good now! (closes issue #14949) Reported by:
+ noahisaac Patches: vm_tempgreeting_removal.patch uploaded by
+ noahisaac (license 748), modified by me ........
+
+2009-09-28 19:13 +0000 [r220724] Sean Bright <sean@malleable.com>
+
+ * /, Makefile.rules: Merged revisions 220721 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r220721 | seanbright | 2009-09-28 15:11:20 -0400 (Mon, 28 Sep
+ 2009) | 10 lines Merged revisions 220717 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220717 | seanbright | 2009-09-28 15:09:25 -0400 (Mon, 28 Sep
+ 2009) | 3 lines When selecting DONT_OPTIMIZE in menuselect,
+ explicitly pass -O0 to the compiler so we override any default
+ optimization levels for a particular install. ........
+ ................
+
+2009-09-26 15:12 +0000 [r220588] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/aes.h: Merged revisions 220586 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r220586 | tilghman | 2009-09-26 10:10:28 -0500 (Sat, 26 Sep 2009)
+ | 2 lines Allow AES to compile, when OpenSSL is not present.
+ ........
+
+2009-09-24 20:38 +0000 [r220371] David Vossel <dvossel@digium.com>
+
+ * main/tcptls.c, /: Merged revisions 220365 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r220365 |
+ dvossel | 2009-09-24 15:37:20 -0500 (Thu, 24 Sep 2009) | 8 lines
+ fixes tcptls_session memory leak caused by ref count error
+ (closes issue #15939) Reported by: dvossel Review:
+ https://reviewboard.asterisk.org/r/375/ ........
+
+2009-09-24 19:42 +0000 [r220291] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_playback.c, main/pbx.c, /, apps/app_disa.c: Merged
+ revisions 220289 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r220289 | tilghman | 2009-09-24 14:41:02 -0500 (Thu, 24 Sep 2009)
+ | 13 lines Merged revisions 220288 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009)
+ | 6 lines Implicitly sending a progress signal breaks some
+ applications. Call Progress() in your dialplan if you explicitly
+ want progress to be sent. (Reverts change 216430, closes issue
+ #15957) Reported by: Pavel Troller on the Asterisk-Dev mailing
+ list
+ http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
+ ........ ................
+
+2009-09-24 18:22 +0000 [r220102-220220] Sean Bright <sean@malleable.com>
+
+ * Makefile, /: Merged revisions 220217 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r220217 | seanbright | 2009-09-24 14:19:41 -0400 (Thu, 24 Sep
+ 2009) | 9 lines Merged revisions 220213 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220213 | seanbright | 2009-09-24 14:18:18 -0400 (Thu, 24 Sep
+ 2009) | 1 line Resolve parallel build warnings. Reported by Klaus
+ Darilion on the asterisk-dev mailing list. ........
+ ................
+
+ * Makefile, build_tools/mkpkgconfig, /: Merged revisions 220100 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r220100 | seanbright | 2009-09-24 10:44:08 -0400
+ (Thu, 24 Sep 2009) | 9 lines Merged revisions 220099 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r220099 | seanbright | 2009-09-24 10:41:57 -0400 (Thu,
+ 24 Sep 2009) | 2 lines Remove the remaining bashisms in the
+ Makefile/mkpkgconfig ........ ................
+
+2009-09-24 08:40 +0000 [r220030] Michiel van Baak <michiel@vanbaak.info>
+
+ * build_tools/mkpkgconfig, /: Merged revisions 220028 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r220028 | mvanbaak | 2009-09-24 10:36:18 +0200
+ (Thu, 24 Sep 2009) | 14 lines Merged revisions 220027 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220027 | mvanbaak | 2009-09-24 10:33:50 +0200 (Thu, 24 Sep 2009)
+ | 7 lines mkpkgconfig does not need bash so make it use /bin/sh
+ This fixes building on all systems that don't have bash at
+ /bin/bash Reported by _ys on #asterisk-dev Tested by _ys on
+ #asterisk-dev ........ ................
+
+2009-09-24 07:44 +0000 [r219988] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_directory.c, /: Merged revisions 219987 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r219987 | tilghman | 2009-09-24 02:39:44 -0500 (Thu, 24 Sep 2009)
+ | 8 lines Fix two possible crashes, one only in 1.6.1 and one in
+ 1.6.1 forward. (closes issue #15739) Reported by: DLNoah, jeffg
+ Patches: 20090914__issue15739.diff.txt uploaded by tilghman
+ (license 14) 20090922__issue15739.diff.txt uploaded by tilghman
+ (license 14) Tested by: DLNoah, jeffg ........
+
+2009-09-22 21:47 +0000 [r219820] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 219818 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r219818 | tilghman | 2009-09-22 16:43:22 -0500
+ (Tue, 22 Sep 2009) | 17 lines Merged revisions 219816 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22 Sep 2009)
+ | 10 lines When IMAP variables were changed during a reload,
+ Voicemail did not use the new values. This change introduces a
+ configuration version variable, which ensures that connections
+ with the old values are not reused but are allowed to expire
+ normally. (closes issue #15934) Reported by: viniciusfontes
+ Patches: 20090922__issue15934.diff.txt uploaded by tilghman
+ (license 14) Tested by: viniciusfontes ........ ................
+
+2009-09-21 17:02 +0000 [r219723] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 219721 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r219721 | dvossel | 2009-09-21 11:59:05 -0500
+ (Mon, 21 Sep 2009) | 9 lines Merged revisions 219720 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r219720 | dvossel | 2009-09-21 11:55:53 -0500 (Mon, 21
+ Sep 2009) | 3 lines Reverting merge 219520. This change was not
+ necessary. ........ ................
+
+2009-09-20 18:21 +0000 [r219667] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/file.c: Merged revisions 219654 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r219654 | tilghman | 2009-09-20 12:55:49 -0500 (Sun, 20 Sep 2009)
+ | 15 lines Merged revisions 219653 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009)
+ | 8 lines Really stop the stream, when ast_closestream() is
+ called. (closes issue #15129) Reported by: bmh Patches:
+ 20090918__issue15129.diff.txt uploaded by tilghman (license 14)
+ Review: https://reviewboard.asterisk.org/r/372/ ........
+ ................
+
+2009-09-19 03:10 +0000 [r219589] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 219587 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r219587 | russell | 2009-09-18 21:59:52 -0500
+ (Fri, 18 Sep 2009) | 13 lines Merged revisions 219586 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219586 | russell | 2009-09-18 21:51:13 -0500 (Fri, 18 Sep 2009)
+ | 6 lines Make sure the iax_pvt exists before dereferencing it.
+ This fixes the latest crash posted on issue 15609. (issue #15609)
+ ........ ................
+
+2009-09-18 23:22 +0000 [r219453-219522] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 219520 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r219520 | dvossel | 2009-09-18 18:20:58 -0500
+ (Fri, 18 Sep 2009) | 15 lines Merged revisions 219519 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18 Sep 2009)
+ | 9 lines iax2 frame double free The iax frame's retrans sched id
+ was written over right before iax2_frame_free was called. In
+ iax2_frame_free that retrans id is used to delete the sched item.
+ By writing over the retrans field before the sched item could be
+ deleted, it was possible for a retransmit to occur on a freed
+ frame. ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 219451 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r219451 | dvossel | 2009-09-18 11:20:41 -0500 (Fri, 18 Sep 2009)
+ | 20 lines Merged revisions 219450 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009)
+ | 14 lines via-header branches not updated correctly on INVITE
+ INVITE requests must always contain a new unique branch id. When
+ a new branch id is created for an INVITE, the dialog's
+ invite_branch variable must be updated so CANCEL requests use the
+ correct branch id. (closes issue #15262) Reported by: maniax
+ Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety
+ (license 608) invite_new_branch_trunk.diff uploaded by dvossel
+ (license 671) Tested by: maniax, dvossel ........
+ ................
+
+2009-09-18 13:57 +0000 [r219414] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 219412 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r219412 | tilghman | 2009-09-18 08:54:51 -0500 (Fri, 18 Sep 2009)
+ | 6 lines Missing value setting line for maxsecs/maxmessage
+ (closes issue #15696) Reported by: fhackenberger Patches:
+ maxsecs.patch uploaded by fhackenberger (license 592) ........
+
+2009-09-17 22:36 +0000 [r219367] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 219324 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r219324 | mmichelson | 2009-09-17 17:22:01 -0500 (Thu, 17 Sep
+ 2009) | 12 lines Merged revisions 219320 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219320 | mmichelson | 2009-09-17 17:20:50 -0500 (Thu, 17 Sep
+ 2009) | 6 lines Send a 100 Trying response when we detect a
+ spiral. This was problematic during spiral tests at SIPit...
+ along with some other things as well. ........ ................
+
+2009-09-17 22:04 +0000 [r219306] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 219304 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r219304 | dvossel | 2009-09-17 16:59:21 -0500 (Thu, 17 Sep 2009)
+ | 27 lines Merged revisions 219303 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009)
+ | 21 lines INVITE w/Replaces deadlock fix This patch cleans up
+ the locking logic in chan_sip.c's handle_invite_replaces()
+ function as well as making use of ast_do_masquerade() rather than
+ forcing the masquerade on an ast_read(). The code had several
+ redundant unlocks that would result in 'freed more times than
+ we've locked!' errors. I cleaned these up as well as moving all
+ the unlock logic to the end of the function. This patch should
+ also resolve the issue people were having with the replacecall
+ channel never being unlocked with one legged calls. (closes issue
+ #15151) Reported by: irroot Patches: invite_w_replaces_1.4.diff
+ uploaded by dvossel (license 671) Tested by: irroot, dvossel
+ Review: https://reviewboard.asterisk.org/r/371/ ........
+ ................
+
+2009-09-17 19:58 +0000 [r219266] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 219264 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r219264 |
+ file | 2009-09-17 14:57:39 -0500 (Thu, 17 Sep 2009) | 2 lines
+ Ensure no spaces exist before "refresher=" when doing the
+ comparison. ........
+
+2009-09-17 Leif Madsen <lmadsen@digium.com>
+
+ * Released Asterisk 1.6.1.7-rc1
+
+2009-09-17 15:44 +0000 [r219199] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, /, include/asterisk/cdr.h,
+ include/asterisk/channel.h: Merged revisions 219139 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r219139 | mnicholson | 2009-09-17 10:18:01 -0500
+ (Thu, 17 Sep 2009) | 17 lines Merged revisions 219136 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep
+ 2009) | 10 lines Prevent a potential race condition and crash
+ when hanging up a channel by removing the channel from the
+ channel list before begining channel tear down. This fix may
+ potentially cause problems with CDR backends that access the
+ channel a CDR is associated with via the channel list. This fix
+ makes the channel unavabile at the time when the CDR backend is
+ invoked. This has been documented in include/asterisk/cdr.h.
+ (closes issue #15316) Reported by: vmarrone Tested by: mnicholson
+ Review: https://reviewboard.asterisk.org/r/362/ ........
+ ................
+
+2009-09-16 23:52 +0000 [r219062] Tilghman Lesher <tlesher@digium.com>
+
+ * main/config.c, configs/extensions.conf.sample, /: Merged
+ revisions 219061 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r219061 | tilghman | 2009-09-16 18:42:12 -0500 (Wed, 16 Sep 2009)
+ | 15 lines Merged revisions 219023 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009)
+ | 8 lines Properly deal with quotes in the arguments of '#exec'
+ includes. (closes issue #15583) Reported by: pkempgen Patches:
+ 20090726__issue15583.diff.txt uploaded by tilghman (license 14)
+ 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license
+ 169) Tested by: pkempgen ........ ................
+
+2009-09-16 19:27 +0000 [r218936] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218933 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218933 |
+ mmichelson | 2009-09-16 14:25:36 -0500 (Wed, 16 Sep 2009) | 12
+ lines Reverse order of args to fread. This way, we don't always
+ write a null byte into byte 1 of the buffer (closes issue #15905)
+ Reported by: ebroad Patches: freadfix.patch uploaded by ebroad
+ (license 878) Tested by: ebroad ........
+
+2009-09-16 19:24 +0000 [r218932] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218918 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218918 |
+ file | 2009-09-16 13:31:47 -0500 (Wed, 16 Sep 2009) | 5 lines On
+ TCP and TLS connections do not attempt to stop retransmission of
+ the packet internally. This was preventing responses from being
+ properly processed because the packet was not being found causing
+ handle_response to return prematurely. ........
+
+2009-09-16 18:23 +0000 [r218890] David Brooks <dbrooks@digium.com>
+
+ * main/pbx.c, /: Merged revisions 218868 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218868 | dbrooks | 2009-09-16 13:06:42 -0500 (Wed, 16 Sep 2009)
+ | 20 lines Merged revisions 218867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009)
+ | 13 lines Fixes CID pattern matching behavior to mirror that of
+ extension pattern matching. Pattern matching for extensions uses
+ a type of scoring system, giving values for specificity to each
+ character in the pattern. Unfortunately, this is done character
+ by character, in order. This does lead to some less specific
+ patterns being first in line for matching, but it will usually
+ get the job done. This patch merely brings CID matching to the
+ same level as extension matching. This patch does not attempt to
+ tackle the problem shared by extension matching. (closes issue
+ #14708) Reported by: klaus3000 ........ ................
+
+2009-09-16 13:37 +0000 [r218801] Russell Bryant <russell@digium.com>
+
+ * contrib/firmware/iax/iaxy.bin (removed), /, UPGRADE.txt: Merged
+ revisions 218799 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218799 | russell | 2009-09-16 08:34:41 -0500 (Wed, 16 Sep 2009)
+ | 16 lines Merged revisions 218798 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009)
+ | 9 lines Remove the IAXy firmware from Asterisk. The firmware
+ can now be found on downloads.digium.com, where the rest of our
+ binary downloads live. This was the last part of our Asterisk
+ tarballs that was considered non-free by Debian. :-) (closes
+ issue #15838) Reported by: paravoid ........ ................
+
+2009-09-15 22:46 +0000 [r218727-218734] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 218731 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r218731 | tilghman | 2009-09-15 17:33:10 -0500
+ (Tue, 15 Sep 2009) | 13 lines Merged revisions 218730 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15 Sep 2009)
+ | 6 lines If the user enters the same password as before, don't
+ signal an error when the change does nothing. (closes issue
+ #15492) Reported by: cbbs70a Patches:
+ 20090713__issue15492.diff.txt uploaded by tilghman (license 14)
+ ........ ................
+
+ * /, channels/chan_gtalk.c: Merged revisions 139281,175058,175089
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ (closes issue #13985) ................ r139281 | phsultan |
+ 2008-08-21 04:55:31 -0500 (Thu, 21 Aug 2008) | 5 lines Fix two
+ memory leaks in chan_gtalk, thanks Eliel! (closes issue #13310)
+ Reported by: eliel Patches: chan_gtalk.c.patch uploaded by eliel
+ (license 64) ................ r175058 | phsultan | 2009-02-12
+ 04:31:36 -0600 (Thu, 12 Feb 2009) | 20 lines Merged revisions
+ 175029 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009)
+ | 12 lines 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 ........
+ ................ r175089 | phsultan | 2009-02-12 08:25:03 -0600
+ (Thu, 12 Feb 2009) | 6 lines Issue a warning message if our
+ candidate's IP is the loopback address. (closes issue #13985)
+ Reported by: jcovert Tested by: phsultan ................
+
+2009-09-15 19:27 +0000 [r218689] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218687 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218687 |
+ dvossel | 2009-09-15 14:22:37 -0500 (Tue, 15 Sep 2009) | 2 lines
+ upward bound checking for port string to int conversion ........
+
+2009-09-15 16:18 +0000 [r218592] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218586 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218586 | mnicholson | 2009-09-15 11:15:02 -0500 (Tue, 15 Sep
+ 2009) | 15 lines Merged revisions 218578 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218578 | mnicholson | 2009-09-15 11:03:54 -0500 (Tue, 15 Sep
+ 2009) | 8 lines Send request contact header field with response
+ to registrer queries instead of the address of record. (closes
+ issue #14438) Reported by: ravindrad Patches: regquerypatch
+ uploaded by ravindrad (license 684) Tested by: ravindrad ........
+ ................
+
+2009-09-15 16:05 +0000 [r218581] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_followme.c, /: Merged revisions 218579 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218579 | tilghman | 2009-09-15 11:04:41 -0500 (Tue, 15 Sep 2009)
+ | 16 lines Merged revisions 218577 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009)
+ | 9 lines Ensure FollowMe sets language in channels it creates.
+ Also, not in the original bug report, but related fields are
+ accountcode and musicclass, and the inheritance of datastores.
+ (closes issue #15372) Reported by: Romik Patches:
+ 20090828__issue15372.diff.txt uploaded by tilghman (license 14)
+ Tested by: cervajs ........ ................
+
+2009-09-15 15:42 +0000 [r218574] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218566 |
+ mmichelson | 2009-09-15 10:40:14 -0500 (Tue, 15 Sep 2009) | 4
+ lines Use a better method of ensuring null-termination of the
+ buffer while reading the SDP when using TCP. ........
+
+2009-09-15 15:41 +0000 [r218569] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Merged revisions 218430 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218430 | jpeeler | 2009-09-14 17:38:25 -0500 (Mon, 14 Sep 2009)
+ | 18 lines Merged revisions 218401 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009)
+ | 11 lines Fix handling of DAHDI_EVENT_REMOVED event to prevent
+ crash in do_monitor. After talking to rmudgett about some of his
+ recent iflist locking changes, it was determined that the only
+ place that would destroy a channel without being explicitly to do
+ so was in handle_init_event. The loop to walk the interface list
+ has been modified to wait to destroy the channel until the
+ dahdi_pvt of the channel to be destroyed is no longer needed.
+ (closes issue #15378) Reported by: samy ........ ................
+
+2009-09-15 15:12 +0000 [r218506] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218499,218504 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r218499 | mmichelson | 2009-09-15 09:59:50 -0500 (Tue,
+ 15 Sep 2009) | 3 lines Fix off-by-one error when reading SDP sent
+ over TCP. ........ r218504 | mmichelson | 2009-09-15 10:05:53
+ -0500 (Tue, 15 Sep 2009) | 3 lines Ensure that SDP read from TCP
+ socket is null-terminated. ........
+
+2009-09-15 15:04 +0000 [r218502] Kevin P. Fleming <kpfleming@digium.com>
+
+ * sounds/Makefile, /: Merged revisions 218500 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218500 | kpfleming | 2009-09-15 11:02:21 -0400 (Tue, 15 Sep
+ 2009) | 9 lines Merged revisions 218497 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218497 | kpfleming | 2009-09-15 10:55:58 -0400 (Tue, 15 Sep
+ 2009) | 1 line Use proper hostname for downloading sound files.
+ ........ ................
+
+2009-09-14 19:49 +0000 [r218363] Tilghman Lesher <tlesher@digium.com>
+
+ * sounds/Makefile, apps/app_voicemail.c, /,
+ configs/voicemail.conf.sample: Merged revisions 218361 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r218361 | tilghman | 2009-09-14 14:29:48 -0500
+ (Mon, 14 Sep 2009) | 11 lines Recorded merge of revisions 218331
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009)
+ | 4 lines Don't say "Please try again" if we don't give the user
+ another chance to try again. (issue #15055, SWP-129) Reported by:
+ jthurman ........ ................
+
+2009-09-14 18:17 +0000 [r218297] Joshua Colp <jcolp@digium.com>
+
+ * /, main/features.c: Merged revisions 218295 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218295 |
+ file | 2009-09-14 13:16:39 -0500 (Mon, 14 Sep 2009) | 2 lines Do
+ not attempt to add a parking extension if an error occurred while
+ reading the configuration. ........
+
+2009-09-14 15:17 +0000 [r218227] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, apps/app_directed_pickup.c: Merged revisions 218224 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r218224 | mnicholson | 2009-09-14 09:57:23 -0500
+ (Mon, 14 Sep 2009) | 14 lines Merged revisions 218223 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep
+ 2009) | 8 lines Ensure we don't pickup ourselves when doing
+ pickup by exten. (closes issue #15100) Reported by: lmsteffan
+ Patches: (modified) pickup.patch uploaded by lmsteffan (license
+ 779) ........ ................
+
+2009-09-13 21:48 +0000 [r218218] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_phone.c, /: gcc 4.4: Remove a nop memset size 0
+ that annoys gcc This memset doesn't write beyond the end of the
+ buffer. (tmpbuf has size of 4). Merged revisions 218184 via
+ svnmerge from http://svn.digium.com/svn/asterisk/trunk
+
+2009-09-12 13:15 +0000 [r218112] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/rtp.c: Use the ip for the new 'rtp set debug ip <foo>'.
+ Since 1.6.X still has the deprecated 'rtp debug ip <foo>' this
+ patch is different from the fix that went into trunk (closes
+ issue 0015711) Reported by: davidw Patches:
+ 2009082800-rtpdebug.diff.txt uploaded by mvanbaak (license 7)
+ Tested by: davidw
+
+2009-09-11 05:59 +0000 [r217924-218054] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 218050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218050 |
+ tilghman | 2009-09-11 00:58:11 -0500 (Fri, 11 Sep 2009) | 3 lines
+ Check the origination priority for more matches, not the current
+ priority. Found by Pavel Troller on the -dev list. ........
+
+ * apps/app_queue.c, /: Merged revisions 217990 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r217990 | tilghman | 2009-09-10 18:54:51 -0500 (Thu, 10 Sep 2009)
+ | 10 lines Merged revisions 217989 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009)
+ | 3 lines Don't ring another channel, if there's not enough time
+ for a queue member to answer. (Fixes AST-228) ........
+ ................
+
+ * channels/chan_iax2.c, contrib/scripts/iax-friends.sql, /,
+ channels/chan_sip.c: Merged revisions 217916 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217916 |
+ tilghman | 2009-09-10 18:12:16 -0500 (Thu, 10 Sep 2009) | 2 lines
+ Make calltoken support work with realtime users and peers.
+ ........
+
+2009-09-10 21:23 +0000 [r217826] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 217807 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r217807 | dvossel | 2009-09-10 16:07:47 -0500
+ (Thu, 10 Sep 2009) | 28 lines Merged revisions 217806 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10 Sep 2009)
+ | 22 lines IAX2 encryption regression The IAX2 Call Token
+ security patch inadvertently broke the use of encryption due to
+ the reorganization of code in the socket_process() function. When
+ encryption is used, an incoming full frame must first be
+ decrypted before the information elements can be parsed. The
+ security release mistakenly moved IE parsing before decryption in
+ order to process the new Call Token IE. To resolve this,
+ decryption of full frames is once again done before looking into
+ the frame. This involves searching for an existing callno,
+ checking the pvt to see if encryption is turned on, and
+ decrypting the packet before the internal fields of the full
+ frame are accessed. (closes issue #15834) Reported by: karesmakro
+ Patches: iax2_encryption_fix_1.4.diff uploaded by dvossel
+ (license 671) Tested by: dvossel, karesmakro Review:
+ https://reviewboard.asterisk.org/r/355/ ........ ................
+
+2009-09-10 19:55 +0000 [r217738] mnick <mnick@localhost>:
+
+ * /, res/res_musiconhold.c: Merged revisions 217730 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r217730 | mnick | 2009-09-10 14:39:41 -0500 (Thu, 10 Sep 2009) |
+ 17 lines Sets the correct musicclass after an announcement
+ (closes issue #15279) Reported by: mbeckwell Patches: patch.txt
+ uploaded by mnick (license ) Tested by: mnick (closes issue
+ #15832) Reported by: mbeckwell Patches: patch.txt uploaded by
+ mnick (license 874) Tested by: mnick ........
+
+2009-09-10 18:18 +0000 [r217642] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_odbc.c, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 217638 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217638 |
+ tilghman | 2009-09-10 13:17:14 -0500 (Thu, 10 Sep 2009) | 4 lines
+ Verify support for wide ODBC character types before using them.
+ (closes issue #15870) Reported by: nic_bellamy ........
+
+2009-09-10 12:11 +0000 [r217595] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 217593 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217593 |
+ oej | 2009-09-10 14:06:55 +0200 (Tor, 10 Sep 2009) | 8 lines
+ Include ActionID in all events that are responsed to AMI Action
+ SIPShowRegistry (closes issue #15868) Reported by: nic_bellamy
+ Patches: manager_SIPshowregistry_actionid.patch uploaded by nic
+ bellamy (license 299) ........
+
+2009-09-09 20:30 +0000 [r217518] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * /, res/res_phoneprov.c: gcc 4.4 fix: union instead of cast gcc
+ 4.4 has more strict rules for aliasing. It doesn't like a struct
+ sockaddr_in pointer pointing to a struct sockaddr. So we make it
+ a union. Merged revisions 217445 via svnmerge from
+ http://svn.digium.com/svn/asterisk/trunk
+
+2009-09-09 11:02 +0000 [r217370] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 217368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217368 |
+ oej | 2009-09-09 12:39:43 +0200 (Ons, 09 Sep 2009) | 2 lines Not
+ having any TLS session to write to is a serious XMIT_ERROR.
+ ........
+
+2009-09-08 22:20 +0000 [r217295] Sean Bright <sean@malleable.com>
+
+ * /, apps/app_meetme.c: Merged revisions 217286 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217286 |
+ seanbright | 2009-09-08 18:17:08 -0400 (Tue, 08 Sep 2009) | 4
+ lines Fix compilation of app_meetme. Reported by ebroad in
+ #asterisk-bugs ........
+
+2009-09-08 20:32 +0000 [r217213] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 217199 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r217199 | tilghman | 2009-09-08 15:28:41 -0500 (Tue, 08 Sep 2009)
+ | 14 lines Merged revisions 217156 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009)
+ | 7 lines When MOH is playing on the channel, announcements sent
+ through the conference are not heard. (closes issue #14588)
+ Reported by: voipas Patches: 20090716__issue14588__2.diff.txt
+ uploaded by tilghman (license 14) Tested by: lmadsen, twisted,
+ tilghman ........ ................
+
+2009-09-08 16:39 +0000 [r217076] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 217074 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217074 |
+ kpfleming | 2009-09-08 11:37:28 -0500 (Tue, 08 Sep 2009) | 9
+ lines Ensure that the default autoconf CFLAGS are not used. A
+ recent change to the configure script that allows the user to
+ specify CFLAGS and/or LDFLAGS to the script had the unfortunate
+ side effect of letting autoconf's default CFLAGS (-g -O2) feed in
+ to the rest of the build system, thereby overriding the
+ DONT_OPTIMIZE setting in menuselect. That problem is now
+ corrected. ........
+
+2009-09-08 15:36 +0000 [r217035] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_limit.c: Merged revisions 217033 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217033 |
+ tilghman | 2009-09-08 10:30:18 -0500 (Tue, 08 Sep 2009) | 4 lines
+ Remove what appears to be an unnecessary define. (closes issue
+ #15851) Reported by: tzafrir ........
+
+2009-09-08 14:27 +0000 [r216995] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 216993 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216993 |
+ dvossel | 2009-09-08 09:26:30 -0500 (Tue, 08 Sep 2009) | 14 lines
+ caller id number empty parse_uri was not being given the correct
+ scheme's, as a result, uri parsing did not parse the username
+ correctly. One of the side effects of this is an empty caller id.
+ (closes issue #15839) Reported by: ebroad Patches:
+ blank_cidv2.patch uploaded by ebroad (license 878)
+ parse_uri_fix.diff uploaded by dvossel (license 671) Tested by:
+ ebroad, dvossel ........
+
+2009-09-07 16:41 +0000 [r216646-216844] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 216842 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216842 |
+ oej | 2009-09-07 18:35:12 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines
+ Make sure we reset global_exclude_static at channel reload
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 216695 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216695 |
+ oej | 2009-09-07 15:06:19 +0200 (MÃ¥n, 07 Sep 2009) | 8 lines If
+ there is no session timer setting in the INVITE, set it to
+ default value (not unset minimum = -1) Patch by oej closes issue
+ #15621 Reported by: fnordian Tested by: atis ........
+
+ * configs/sip.conf.sample: Make code and documentation agree with
+ each other
+
+ * CHANGES, channels/chan_sip.c: Turning off premature media by
+ default
+
+ * apps/app_playback.c, main/pbx.c, /, channels/chan_sip.c,
+ apps/app_disa.c, configs/sip.conf.sample: Merged revisions 216438
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre,
+ 04 Sep 2009) | 35 lines Merged revisions 216430 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27
+ lines Make apps send PROGRESS control frame for early media and
+ fix too early media issue in SIP The issue at hand is that some
+ legacy (dying) PBX systems send empty media frames on PRI links
+ *before* any call progress. The SIP channel receives these frames
+ and by default signals 183 Session progress and starts sending
+ media. This will cause phones to play silence and ignore the
+ later 180 ringing message. A bad user experience. The fix is
+ twofold: - We discovered that asterisk apps that support early
+ media ("noanswer") did not send any PROGRESS frame to indicate
+ early media. Fixed. - We introduce a setting in chan_sip so that
+ users can disable any relay of media frames before the outbound
+ channel actually indicates any sort of call progress. In 1.4,
+ 1.6.0 and 1.6.1, this will be disabled for backward
+ compatibility. In later versions of Asterisk, this will be
+ enabled. We don't assume that it will change your Asterisk phone
+ experience - only for the better. We encourage third-party
+ application developers to make sure that if they have
+ applications that wants to send early media, add a PROGRESS
+ control frame transmission to make sure that all channel drivers
+ actually will start sending early media. This has not been the
+ default in Asterisk previous to this patch, so if you got
+ inspiration from our code, you need to update accordingly. Sorry
+ for the trouble and thanks for your support. This code has been
+ running for a few months in a large scale installation (over 250
+ servers with PRI and/or BRI links to old PBX systems). That's no
+ proof that this is an excellent patch, but, well, it's tested :-)
+ ........ ................
+
+2009-09-04 19:51 +0000 [r216599] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 216594 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216594 |
+ dvossel | 2009-09-04 14:32:07 -0500 (Fri, 04 Sep 2009) | 7 lines
+ sip peer matching by address only with TCP/TLS This patch removes
+ the contact header matching logic and adds logic to match all
+ tcp/tls connections by ip only Review:
+ https://reviewboard.asterisk.org/r/354/ ........
+
+2009-09-04 19:32 +0000 [r216596] Sean Bright <sean@malleable.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 216593 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r216593 | seanbright | 2009-09-04 15:29:02 -0400 (Fri, 04 Sep
+ 2009) | 1 line Use ast_free() instead of free(). ........
+
+2009-09-04 17:53 +0000 [r216549-216552] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 216551 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r216551 | tilghman | 2009-09-04 12:50:21 -0500 (Fri, 04 Sep 2009)
+ | 2 lines Fix trunk breakage. ........
+
+ * main/pbx.c, /, UPGRADE-1.6.txt: Merged revisions 216547 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r216547 | tilghman | 2009-09-04 12:31:44 -0500 (Fri, 04
+ Sep 2009) | 3 lines Enable turning off the application delimiter
+ warning with the 'dontwarn' option. Suggested on the -dev list,
+ and implemented in an alternate way by me. ........
+
+2009-09-04 15:09 +0000 [r216440-216508] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, main/utils.c: Merged revisions 216506 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r216506 | mvanbaak | 2009-09-04 17:05:05 +0200 (Fri, 04 Sep 2009)
+ | 9 lines Merged revisions 216435 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009)
+ | 2 lines make asterisk compile under devmode with DEBUG_THREADS
+ enabled on OpenBSD ........ ................
+
+ * /, include/asterisk/lock.h: Merged revisions 216437 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r216437 | mvanbaak | 2009-09-04 16:00:38 +0200 (Fri, 04 Sep 2009)
+ | 2 lines make sure canlog is set so we can compile with
+ DEBUG_THREADS enabled on OpenBSD ........
+
+2009-09-04 13:56 +0000 [r216266-216434] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 216368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216368 |
+ russell | 2009-09-04 08:14:25 -0500 (Fri, 04 Sep 2009) | 12 lines
+ Do not treat every SIP peer as if they were configured with
+ insecure=port. There was a problem in the function responsible
+ for doing peer matching by IP address and port number such that
+ during the second pass for checking for a peer configured with
+ insecure=port, it would end up treating every peer as if it had
+ been configured that way. These changes fix the logic in the peer
+ IP and port comparison callback to handle insecure=port checking
+ properly. This problem was introduced when SIP peers were
+ converted to astobj2. Many thanks to dvossel for noticing this
+ while working on another peer matching issue. ........
+
+ * doc/IAX2-security.txt (added), /: Merged revisions 216264 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r216264 | russell | 2009-09-04 05:48:44 -0500
+ (Fri, 04 Sep 2009) | 16 lines Merged revisions 216263 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r216263 | russell | 2009-09-04 05:48:00 -0500
+ (Fri, 04 Sep 2009) | 9 lines Merged revisions 216262 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04
+ Sep 2009) | 2 lines Add a plain text version of the IAX2 security
+ document. ........ ................ ................
+
+2009-09-04 06:13 +0000 [r216224] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/astobj2.c, /: Merged revisions 216222 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216222 |
+ mvanbaak | 2009-09-04 08:08:33 +0200 (Fri, 04 Sep 2009) | 3 lines
+ make sure 'start' is always initialized. Makes asterisk compile
+ with --enable-dev-mode ........
+
+2009-09-03 19:42 +0000 [r216013-216098] Russell Bryant <russell@digium.com>
+
+ * UPGRADE.txt: tweak
+
+ * /, UPGRADE.txt: Merged revisions 216092 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r216092 | russell | 2009-09-03 14:38:35 -0500 (Thu, 03 Sep 2009)
+ | 16 lines Merged revisions 216085 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r216085 | russell | 2009-09-03 14:36:46 -0500
+ (Thu, 03 Sep 2009) | 9 lines Merged revisions 216080 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03
+ Sep 2009) | 2 lines Add a note about IAX2 to UPGRADE.txt.
+ ........ ................ ................
+
+ * /, doc/IAX2-security.pdf (added): Merged revisions 216009 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r216009 | russell | 2009-09-03 13:45:54 -0500
+ (Thu, 03 Sep 2009) | 16 lines Merged revisions 216008 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r216008 | russell | 2009-09-03 13:44:58 -0500
+ (Thu, 03 Sep 2009) | 9 lines Merged revisions 216005 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03
+ Sep 2009) | 2 lines Add IAX2 security document related to
+ AST-2009-006. ........ ................ ................
+
+2009-09-03 18:41 +0000 [r216004] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, 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: Merged revisions 215955 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r215955 | dvossel | 2009-09-03 11:31:54 -0500 (Thu, 03 Sep 2009)
+ | 6 lines Merge code associated with AST-2009-006 (closes issue
+ #12912) Reported by: rathaus Tested by: tilghman, russell,
+ dvossel, dbrooks ........
+
+2009-09-03 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.1.6 released
+
+ * AST-2009-006
+
+2009-08-28 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.1.5 released
+
+2009-08-11 Tilghman Lesher <tlesher@digium.com>
+
+ * Asterisk 1.6.1.5-rc1 released
+
+2009-08-10 19:51 +0000 [r211569-211586] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/CODING-GUIDELINES, /: Merged revisions 211584 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r211584 | tilghman | 2009-08-10 14:49:41 -0500
+ (Mon, 10 Aug 2009) | 9 lines Merged revisions 211583 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r211583 | tilghman | 2009-08-10 14:48:48 -0500 (Mon, 10
+ Aug 2009) | 1 line Conversion specifiers, not format specifiers
+ ........ ................
+
+ * channels/chan_iax2.c, res/ael/pval.c, main/cdr.c, main/channel.c,
+ main/manager.c, apps/app_setcallerid.c, apps/app_rpt.c,
+ main/asterisk.c, res/res_config_pgsql.c, apps/app_dahdibarge.c,
+ funcs/func_rand.c, funcs/func_timeout.c, apps/app_record.c,
+ codecs/codec_speex.c, apps/app_morsecode.c, main/acl.c,
+ funcs/func_cut.c, cdr/cdr_pgsql.c, apps/app_followme.c,
+ main/enum.c, res/res_config_sqlite.c, main/config.c,
+ agi/eagi-sphinx-test.c, channels/misdn_config.c,
+ channels/chan_dahdi.c, funcs/func_channel.c, apps/app_macro.c,
+ apps/app_sms.c, pbx/pbx_config.c, apps/app_verbose.c, main/dsp.c,
+ apps/app_voicemail.c, apps/app_adsiprog.c, funcs/func_speex.c,
+ channels/chan_sip.c, res/res_limit.c, channels/chan_agent.c,
+ agi/eagi-test.c, funcs/func_math.c, main/utils.c,
+ channels/iax2-provision.c, apps/app_talkdetect.c,
+ main/indications.c, channels/chan_oss.c, main/cli.c,
+ res/res_config_curl.c, pbx/pbx_loopback.c, res/res_smdi.c,
+ apps/app_osplookup.c, channels/chan_misdn.c,
+ channels/chan_skinny.c, pbx/pbx_dundi.c, utils/extconf.c,
+ apps/app_mixmonitor.c, channels/chan_mgcp.c, main/timing.c,
+ main/pbx.c, doc/CODING-GUIDELINES, utils/muted.c,
+ apps/app_readfile.c, /, apps/app_meetme.c, apps/app_privacy.c,
+ apps/app_waituntil.c, cdr/cdr_adaptive_odbc.c,
+ pbx/dundi-parser.c, res/res_http_post.c, res/res_musiconhold.c,
+ apps/app_queue.c, main/netsock.c, utils/frame.c,
+ channels/chan_usbradio.c, funcs/func_enum.c,
+ channels/chan_phone.c, apps/app_waitforring.c, pbx/pbx_spool.c,
+ funcs/func_odbc.c, apps/app_minivm.c, main/features.c,
+ res/res_agi.c, main/http.c, res/snmp/agent.c,
+ res/res_config_ldap.c, apps/app_chanspy.c, apps/app_stack.c,
+ res/res_odbc.c, funcs/func_dialplan.c, main/dnsmgr.c,
+ main/frame.c, apps/app_waitforsilence.c, funcs/func_strings.c,
+ apps/app_disa.c, apps/app_alarmreceiver.c: AST-2009-005
+
+2009-08-10 14:12 +0000 [r211349] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 211347 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r211347 |
+ file | 2009-08-10 11:07:44 -0300 (Mon, 10 Aug 2009) | 5 lines Fix
+ retrieval of the port used for the video stream when adding SDP
+ to a SIP message. (closes issue #15121) Reported by: jsmith
+ ........
+
+2009-08-09 15:43 +0000 [r211234-211277] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/astfd.c: Merged revisions 211275 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r211275 | tilghman | 2009-08-09 10:42:02 -0500 (Sun, 09 Aug 2009)
+ | 9 lines Merged revisions 211274 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211274 | tilghman | 2009-08-09 10:41:01 -0500 (Sun, 09 Aug 2009)
+ | 2 lines Small oops. Clear the flags which have been checked.
+ ........ ................
+
+ * apps/app_stack.c, /: Merged revisions 211232 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r211232 |
+ tilghman | 2009-08-09 02:11:22 -0500 (Sun, 09 Aug 2009) | 4 lines
+ Check for NULL frame, before dereferencing pointer. (closes issue
+ #15617) Reported by: rain ........
+
+2009-08-07 20:17 +0000 [r211115] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 211113 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r211113 | russell | 2009-08-07 15:12:21 -0500 (Fri, 07 Aug 2009)
+ | 11 lines Recorded merge of revisions 211112 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211112 | russell | 2009-08-07 15:11:31 -0500 (Fri, 07 Aug 2009)
+ | 4 lines Resolve a deadlock involving app_chanspy and
+ masquerades. (ABE-1936) ........ ................
+
+2009-08-07 18:19 +0000 [r211047] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 211040 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r211040 | tilghman | 2009-08-07 13:17:41 -0500 (Fri, 07 Aug 2009)
+ | 21 lines Merged revisions 211038 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009)
+ | 14 lines QUEUE_MEMBER_LIST _really_ wants the interface name,
+ not the membername. This is a partial revert of revision 82590,
+ which was an attempted cleanup, but in reality, it broke
+ QUEUE_MEMBER_LIST, which has always been intended as a method by
+ which component interfaces could be queried from the queue.
+ Membername isn't useful here, because that field cannot be used
+ to obtain further information about the member. See the
+ documentation on QUEUE_MEMBER_LIST, RemoveQueueMember,
+ QUEUE_MEMBER_PENALTY, and the various AMI commands which take a
+ member argument for further justification. (closes issue #15664)
+ Reported by: rain Patches: app_queue-queue_member_list.diff
+ uploaded by rain (license 327) ........ ................
+
+2009-08-07 13:09 +0000 [r210994] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, /: Merged revisions 210992 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r210992 |
+ kpfleming | 2009-08-07 08:08:00 -0500 (Fri, 07 Aug 2009) | 13
+ lines Workaround broken T.38 endpoints that offer tiny
+ MaxDatagram sizes. Some T.38 endpoints treat T38FaxMaxDatagram as
+ the maximum IFP size that should be sent to them, rather than the
+ maximum packet payload size. If such an endpoint also requests
+ UDPRedundancy as the error correction mode, we'll end up
+ calculating a tiny maximum IFP size, so small as to be unusable.
+ This patch sets a lower bound on what we'll consider the remote's
+ maximum IFP size to be, assuming that endpoints that do this
+ really can accept larger packets than they've offered to accept.
+ (closes issue #15649) Reported by: dazza76 ........
+
+2009-08-06 21:47 +0000 [r210910-210916] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 210914 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r210914 | tilghman | 2009-08-06 16:46:01 -0500 (Thu, 06 Aug 2009)
+ | 14 lines Merged revisions 210913 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210913 | tilghman | 2009-08-06 16:45:01 -0500 (Thu, 06 Aug 2009)
+ | 7 lines Because channel information can be accessed outside of
+ the channel thread, we must lock the channel prior to modifying
+ it. (closes issue #15397) Reported by: caspy Patches:
+ 20090714__issue15397.diff.txt uploaded by tilghman (license 14)
+ Tested by: caspy ........ ................
+
+ * apps/app_stack.c, include/asterisk/app.h, /, main/app.c: Merged
+ revisions 210908 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r210908 |
+ tilghman | 2009-08-06 16:29:26 -0500 (Thu, 06 Aug 2009) | 9 lines
+ Allow Gosub to recognize quote delimiters without consuming them.
+ (closes issue #15557) Reported by: rain Patches:
+ 20090723__issue15557.diff.txt uploaded by tilghman (license 14)
+ Tested by: rain Review: https://reviewboard.asterisk.org/r/316/
+ ........
+
+2009-08-06 17:48 +0000 [r210819] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 210817 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r210817 |
+ file | 2009-08-06 14:47:04 -0300 (Thu, 06 Aug 2009) | 11 lines
+ Accept additional T.38 reinvites after an initial one has been
+ handled. Discussion of this subject has yielded that it is not
+ actually acceptable to change T.38 parameters after the initial
+ reinvite but declining is harsh and can cause the fax to fail
+ when it may be possible to allow it to continue. This patch
+ changes things so that additional T.38 reinvites are accepted but
+ parameter changes ignored. This gives the fax a fighting chance.
+ (closes issue #15610) Reported by: huangtx2009 ........
+
+2009-08-05 20:28 +0000 [r210681] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 210640 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r210640 | rmudgett | 2009-08-05 14:40:03 -0500
+ (Wed, 05 Aug 2009) | 21 lines Merged revisions 210575 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009)
+ | 14 lines Dialplan starts execution before the channel setup is
+ complete. * Issue 15655: For the case where dialing is complete
+ for an incoming call, dahdi_new() was asked to start the PBX and
+ then the code set more channel variables. If the dialplan hungup
+ before these channel variables got set, asterisk would likely
+ crash. * Fixed potential for overlap incoming call to erroneously
+ set channel variables as global dialplan variables if the
+ ast_channel structure failed to get allocated. * Added missing
+ set of CALLINGSUBADDR in the dialing is complete case. (closes
+ issue #15655) Reported by: alecdavis ........ ................
+
+2009-08-05 18:57 +0000 [r210567] Leif Madsen <lmadsen@digium.com>
+
+ * doc/tex/imapstorage.tex, /: Merged revisions 210564 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r210564 | lmadsen | 2009-08-05 13:49:58 -0500
+ (Wed, 05 Aug 2009) | 19 lines Merged revisions 210563 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009)
+ | 11 lines Update imapstorage.txt documentation. Updated the
+ imapstorage.txt documentation to reflect that issues with
+ c-client versions older than 2007 seem to cause crashing issues
+ that are not seen with more recent versions. Documentation has
+ been updated to reflect this. (closes issue #14496) Reported by:
+ vbcrlfuser Patches: __20090727-imap-documentation-patch.txt
+ uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson,
+ dbrooks ........ ................
+
+2009-08-04 14:54 +0000 [r210240] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, /: Merged revisions 210238 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r210238 | kpfleming | 2009-08-04 09:53:00 -0500 (Tue, 04 Aug
+ 2009) | 16 lines Merged revisions 210237 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210237 | kpfleming | 2009-08-04 09:51:39 -0500 (Tue, 04 Aug
+ 2009) | 10 lines Eliminate spurious compiler warnings from system
+ headers on *BSD platforms. Ensure that system headers located in
+ /usr/local/include are actually treated as system headers by the
+ compiler, and not as local headers which are subject to warnings
+ from the -Wundef compiler option and others. (closes issue
+ #15606) Reported by: mvanbaak ........ ................
+
+2009-08-01 11:32 +0000 [r209836-209900] Russell Bryant <russell@digium.com>
+
+ * main/db1-ast/mpool/mpool.c, /: Merged revisions 209887 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r209887 | russell | 2009-08-01 06:29:25 -0500
+ (Sat, 01 Aug 2009) | 12 lines Merged revisions 209879 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209879 | russell | 2009-08-01 06:27:25 -0500 (Sat, 01 Aug 2009)
+ | 5 lines Resolve a valgrind warning about a read from
+ uninitialized memory. (issue #15396) Reported by: aragon ........
+ ................
+
+ * apps/app_milliwatt.c, /: Merged revisions 209839 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r209839 | russell | 2009-08-01 06:02:07 -0500
+ (Sat, 01 Aug 2009) | 20 lines Merged revisions 209838 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01 Aug 2009)
+ | 13 lines Modify how Playtones() is used in Milliwatt() to
+ resolve gain issue. When Milliwatt() was changed internally to
+ use Playtones() so that the proper tone was used, it introduced a
+ drop in gain in the output signal. So, use the playtones API
+ directly and specify a volume argument such that the output
+ matches the gain of the original Milliwatt() code. (closes issue
+ #15386) Reported by: rue_mohr Patches: issue_15386.rev2.diff
+ uploaded by russell (license 2) Tested by: rue_mohr ........
+ ................
+
+ * /, main/event.c: Merged revisions 209835 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209835 |
+ russell | 2009-08-01 05:43:40 -0500 (Sat, 01 Aug 2009) | 6 lines
+ Fix ast_event_queue_and_cache() to actually do the cache() part.
+ (closes issue #15624) Reported by: ffossard Tested by: russell
+ ........
+
+2009-08-01 01:25 +0000 [r209781] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/misdn/isdn_lib.c, utils/frame.c, /, main/Makefile,
+ channels/misdn/ie.c: Merged revisions 209760-209761 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r209760 | kpfleming | 2009-07-31 20:03:07 -0500
+ (Fri, 31 Jul 2009) | 13 lines Merged revisions 209759 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul
+ 2009) | 7 lines Minor changes inspired by testing with latest
+ GCC. The latest GCC (what will become 4.5.x) has a few new
+ warnings, that in these cases found some either downright buggy
+ code, or at least seriously poorly designed code that could be
+ improved. ........ ................ r209761 | kpfleming |
+ 2009-07-31 20:04:06 -0500 (Fri, 31 Jul 2009) | 1 line Revert
+ accidental Makefile change. ................
+
+2009-07-31 21:58 +0000 [r209714] Russell Bryant <russell@digium.com>
+
+ * /, main/event.c: Merged revisions 209711 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209711 |
+ russell | 2009-07-31 16:53:31 -0500 (Fri, 31 Jul 2009) | 2 lines
+ Fix some places where ast_event_type was used instead of
+ ast_event_ie_type. ........
+
+2009-07-30 18:46 +0000 [r209593] David Brooks <dbrooks@digium.com>
+
+ * include/asterisk/abstract_jb.h, channels/chan_dahdi.c,
+ contrib/init.d/rc.debian.asterisk, /, apps/app_sms.c,
+ codecs/lpc10/pitsyn.c, channels/chan_console.c: Merged revisions
+ 209554 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 |
+ dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines
+ Fixes numerous spelling errors. Patch submitted by alecdavis.
+ (closes issue #15595) Reported by: alecdavis ........
+
+2009-07-30 14:40 +0000 [r209517] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 209516 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209516 |
+ mmichelson | 2009-07-30 09:38:21 -0500 (Thu, 30 Jul 2009) | 8
+ lines Fix a crash that can result if text codecs are allowed but
+ textsupport is disabled. (closes issue #15596) Reported by:
+ fabled Patches: sip-red.patch uploaded by fabled (license 448)
+ ........
+
+2009-07-28 00:19 +0000 [r209327] Tilghman Lesher <tlesher@digium.com>
+
+ * /, sounds/sounds.xml: Merged revisions 209317 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r209317 | tilghman | 2009-07-27 19:14:12 -0500 (Mon, 27 Jul 2009)
+ | 9 lines Merged revisions 209315 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009)
+ | 2 lines Publish French extra sounds ........ ................
+
+2009-07-27 21:44 +0000 [r209262-209281] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 209279 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209279 |
+ kpfleming | 2009-07-27 16:43:36 -0500 (Mon, 27 Jul 2009) | 7
+ lines Cleanup T.38 negotiation changes. Convert LOG_NOTICE
+ messages about T.38 negotiation in debug level 1 messages, clean
+ up some looping logic, and correct an improper use of ast_free()
+ for freeing an ast_frame. ........
+
+ * /, apps/app_fax.c: Merged revisions 209256 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209256 |
+ kpfleming | 2009-07-27 16:21:43 -0500 (Mon, 27 Jul 2009) | 10
+ lines Make T.38 switchover in ReceiveFAX synchronous. In receive
+ mode, if the channel that ReceiveFAX is running on supports T.38,
+ we should *always* attempt to switch T.38, rather than listening
+ for an incoming CNG tone and only triggering on that. The channel
+ may be using a low-bitrate codec that distorts the CNG tone, the
+ sending FAX endpoint may not send CNG at all, or there could be a
+ variety of other reasons that we don't detect it, but in all
+ those cases if T.38 is available we certainly want to use it.
+ ........
+
+2009-07-27 20:57 +0000 [r209237] Mark Michelson <mmichelson@digium.com>
+
+ * main/rtp.c, /: Merged revisions 209235 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209235 |
+ mmichelson | 2009-07-27 15:54:54 -0500 (Mon, 27 Jul 2009) | 5
+ lines Gracefully handle malformed RTP text packets. AST-2009-004
+ ........
+
+2009-07-27 20:28 +0000 [r209233] David Brooks <dbrooks@digium.com>
+
+ * res/res_jabber.c, main/loader.c, channels/chan_dahdi.c,
+ channels/chan_vpb.cc, res/res_smdi.c, /,
+ include/asterisk/module.h, main/features.c, res/res_agi.c: Merged
+ revisions 209098 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 |
+ dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines
+ Fixing typos. Replaces "recieved" with "received" and "initilize"
+ with "initialize" (closes issue #15571) Reported by: alecdavis
+ ........
+
+2009-07-27 20:17 +0000 [r209134-209199] Mark Michelson <mmichelson@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 209197 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r209197 | mmichelson | 2009-07-27 15:11:42 -0500 (Mon, 27 Jul
+ 2009) | 9 lines Honor channel's music class when using realtime
+ music on hold. (closes issue #15051) Reported by: alexh Patches:
+ 15051.patch uploaded by mmichelson (license 60) Tested by: alexh
+ ........
+
+ * main/udptl.c, /, configs/udptl.conf.sample: Merged revisions
+ 209132 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r209132 | mmichelson | 2009-07-27 12:50:04 -0500 (Mon, 27 Jul
+ 2009) | 24 lines Merged revisions 209131 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul
+ 2009) | 18 lines Allow for UDPTL to use only even-numbered ports
+ if desired. There are some VoIP providers out there that will not
+ accept SDP offers with odd numbered UDPTL ports. While it is my
+ personal opinion that these VoIP providers are misinterpreting
+ RFC 2327, it really is not a big deal to play along with their
+ silly little games. Of course, since restricting UDPTL ports to
+ only even numbers reduces the range of available ports by half,
+ so the option to use only even port numbers is off by default. A
+ user can enable the behavior by setting use_even_ports=yes in
+ udptl.conf. (closes issue #15182) Reported by: CGMChris Patches:
+ 15182.patch uploaded by mmichelson (license 60) Tested by:
+ CGMChris ........ ................
+
+2009-07-27 15:40 +0000 [r209058] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, /: Merged revisions 209056 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209056 |
+ kpfleming | 2009-07-27 10:38:59 -0500 (Mon, 27 Jul 2009) | 10
+ lines Restore explicit export of ASTCFLAGS/ASTLDFLAGS and
+ underscore-variants to sub-makes. During the recent Makefile
+ improvements I made, it seemed the 'make' was automatically
+ carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes, so
+ I removed the explict export of them. However, there are some
+ circumstances where make does this, and some where it does not,
+ so I've brought them back to ensure they are always exported. I
+ also removed an extraneous double setting of _ASTLDFLAGS on *BSD
+ platforms. ........
+
+2009-07-27 01:22 +0000 [r208926] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_iax2.c, /, main/translate.c: Merged revisions
+ 208924 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208924 | jpeeler | 2009-07-26 20:20:37 -0500 (Sun, 26 Jul 2009)
+ | 9 lines Merged revisions 208923 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009)
+ | 2 lines Fix logic errors from 208746 ........ ................
+
+2009-07-26 14:04 +0000 [r208888] Michiel van Baak <michiel@vanbaak.info>
+
+ * contrib/scripts/install_prereq, /: Merged revisions 208886 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r208886 | mvanbaak | 2009-07-26 16:00:52 +0200 (Sun, 26
+ Jul 2009) | 2 lines add OpenBSD to the install_prereq script
+ ........
+
+2009-07-25 06:25 +0000 [r208754] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_iax2.c, /, channels/chan_skinny.c,
+ main/translate.c: Merged revisions 208749 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208749 | jpeeler | 2009-07-25 01:23:18 -0500 (Sat, 25 Jul 2009)
+ | 13 lines Merged revisions 208746 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009)
+ | 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly
+ trivial changes, but I did not know of any other way to fix the
+ "dereferencing type-punned pointer will break strict-aliasing
+ rules" error without creating a tmp variable in chan_skinny.
+ ........ ................
+
+2009-07-24 18:52 +0000 [r208595] Russell Bryant <russell@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 208593 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208593 | russell | 2009-07-24 13:42:32 -0500 (Fri, 24 Jul 2009)
+ | 14 lines Merged revisions 208592 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009)
+ | 7 lines Do not log an ERROR if autoservice_stop() returns -1.
+ This does not indicate an error. A return of -1 just means that
+ the channel has been hung up. (reported in #asterisk-dev)
+ ........ ................
+
+2009-07-24 18:32 +0000 [r208590] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 208588 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208588 | mmichelson | 2009-07-24 13:31:04 -0500 (Fri, 24 Jul
+ 2009) | 16 lines Merged revisions 208587 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul
+ 2009) | 10 lines Only send a BYE when hanging up a channel that
+ is up. For cases where Asterisk sends an INVITE and receives a
+ non 2XX final response, Asterisk would follow the INVITE
+ transaction by immediately sending a BYE, which was unnecessary.
+ (closes issue #14575) Reported by: chris-mac ........
+ ................
+
+2009-07-24 15:05 +0000 [r208550] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h:
+ Merged revisions 208548 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208548 |
+ kpfleming | 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) | 8
+ lines Resolve a T.38 negotiation issue left over from the
+ udptl-updates merge. The udptl-updates branch that was merged
+ yesterday failed to properly send back T.38 SDP responses with
+ the correct error correction mode, if the incoming SDP from the
+ other end caused us to change error correction modes. This patch
+ corrects that situation. ........
+
+2009-07-24 14:38 +0000 [r208544] Michiel van Baak <michiel@vanbaak.info>
+
+ * contrib/scripts/install_prereq, /: Merged revisions 208542 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r208542 | mvanbaak | 2009-07-24 16:35:49 +0200 (Fri, 24
+ Jul 2009) | 13 lines use aptitude for debian based systems The
+ function to check wether we need to install packages was using
+ dpkg-query which was gives wrong output on Debian 5 Also, the
+ apt-get has been replaced with aptitude because aptitude is now
+ the preferred way to handle packages on Debian (closes issue
+ #15570) Reported by: mvanbaak Patches:
+ 2009072400_installprereq-aptitude.diff uploaded by mvanbaak
+ (license 7) ........
+
+2009-07-23 22:32 +0000 [r208484-208503] Kevin P. Fleming <kpfleming@digium.com>
+
+ * UPGRADE.txt: Use correct formatting for T.38 change note in
+ UPGRADE.txt
+
+ * include/asterisk/frame.h, main/rtp.c, main/channel.c,
+ main/udptl.c, main/frame.c, /, channels/chan_sip.c,
+ apps/app_fax.c, UPGRADE.txt, include/asterisk/udptl.h: Merged
+ revisions 208464 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 |
+ kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46
+ lines Rework of T.38 negotiation and UDPTL API to address
+ interoperability problems Over the past couple of months, a
+ number of issues with Asterisk negotiating (and successfully
+ completing) T.38 sessions with various endpoints have been found.
+ This patch attempts to address many of them, primarily focused
+ around ensuring that the endpoints' MaxDatagram size is honored,
+ and in addition by ensuring that T.38 session parameter
+ negotiation is performed correctly according to the ITU T.38
+ Recommendation. The major changes here are: 1) T.38 applications
+ in Asterisk (app_fax) only generate/receive IFP packets, they do
+ not ever work with UDPTL packets. As a result of this, they
+ cannot be allowed to generate packets that would overflow the
+ other endpoints' MaxDatagram size after the UDPTL stack adds any
+ error correction information. With this patch, the application is
+ told the maximum *IFP* size it can generate, based on a
+ calculation using the far end MaxDatagram size and the active
+ error correction mode on the T.38 session. The same is true for
+ sending *our* MaxDatagram size to the remote endpoint; it is
+ computed from the value that the application says it can accept
+ (for a single IFP packet) combined with the active error
+ correction mode. 2) All treatment of T.38 session parameters as
+ 'capabilities' in chan_sip has been removed; these parameters are
+ not at all like audio/video stream capabilities. There are strict
+ rules to follow for computing an answer to a T.38 offer, and
+ chan_sip now follows those rules, using the desired parameters
+ from the application (or channel) that wants to accept the T.38
+ negotiation. 3) chan_sip now stores and forwards
+ ast_control_t38_parameters structures for tracking 'our' and
+ 'their' T.38 session parameters; this greatly simplifies
+ negotiation, especially for pass-through calls. 4) Since T.38
+ negotiation without specifying parameters or receiving the final
+ negotiated parameters is not very worthwhile, the AST_CONTROL_T38
+ control frame has been removed. A note has been added to
+ UPGRADE.txt about this removal, since any out-of-tree
+ applications that use it will no longer function properly until
+ they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review:
+ https://reviewboard.asterisk.org/r/310/ ........
+
+2009-07-23 20:45 +0000 [r208459] David Brooks <dbrooks@digium.com>
+
+ * apps/app_rpt.c, res/res_smdi.c, pbx/pbx_dundi.c: Just replacing
+ typos "recieved" with "received". (closes issue #15360) Reported
+ by: okrief
+
+2009-07-23 19:35 +0000 [r208390] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 208388 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208388 | mmichelson | 2009-07-23 14:34:49 -0500 (Thu, 23 Jul
+ 2009) | 24 lines Merged revisions 208386 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208386 | mmichelson | 2009-07-23 14:24:21 -0500 (Thu, 23 Jul
+ 2009) | 17 lines Fix a problem where a 491 response could be sent
+ out of dialog. This generalizes the fix for issue 13849. The
+ initial fix corrected the problem that Asterisk would reply with
+ a 491 if a reinvite were received from an endpoint and we had not
+ yet received an ACK from that endpoint for the initial INVITE it
+ had sent us. This expansion also allows Asterisk to appropriately
+ handle an INVITE with authorization credentials if Asterisk had
+ not received an ACK from the previous transaction in which
+ Asterisk had responded to an unauthorized INVITE with a 407.
+ (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch
+ uploaded by mmichelson (license 60) Tested by: klaus3000 ........
+ ................
+
+2009-07-23 19:24 +0000 [r208385] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 208383 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r208383 | jpeeler | 2009-07-23 14:21:50 -0500
+ (Thu, 23 Jul 2009) | 12 lines Merged revisions 208380 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208380 | jpeeler | 2009-07-23 14:19:53 -0500 (Thu, 23 Jul 2009)
+ | 6 lines Only set the priindication setting when not performing
+ a reload (closes issue #14696) Reported by: fdecher ........
+ ................
+
+2009-07-23 16:30 +0000 [r208265-208318] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 208314 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208314 | mmichelson | 2009-07-23 11:29:37 -0500 (Thu, 23 Jul
+ 2009) | 9 lines Merged revisions 208312 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208312 | mmichelson | 2009-07-23 11:29:18 -0500 (Thu, 23 Jul
+ 2009) | 3 lines Remove inaccurate XXX comment. ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 208263 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208263 | mmichelson | 2009-07-23 10:46:34 -0500 (Thu, 23 Jul
+ 2009) | 15 lines Merged revisions 208262 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208262 | mmichelson | 2009-07-23 10:43:07 -0500 (Thu, 23 Jul
+ 2009) | 8 lines Properly handle 183 responses which do not
+ contain an SDP. (closes issue #15442) Reported by: ffloimair
+ Patches: 15442.patch uploaded by mmichelson (license 60) Tested
+ by: tkarl, ffloimair ........ ................
+
+2009-07-22 21:45 +0000 [r208115] Jason Parker <jparker@digium.com>
+
+ * /, apps/app_festival.c: Merged revisions 208113 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208113 |
+ qwell | 2009-07-22 16:43:57 -0500 (Wed, 22 Jul 2009) | 9 lines
+ Restore an int declaration on PPC platforms. This x is one crafty
+ little bugger... It was used for 2 different things (one of which
+ was only done on PPC) in 1.4. One of the uses were removed in
+ trunk, and with it went the declaration. (closes issue #14038)
+ Reported by: ffloimair ........
+
+2009-07-21 22:48 +0000 [r207948] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_strings.c: Merged revisions 207946 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r207946 | tilghman | 2009-07-21 17:45:32 -0500
+ (Tue, 21 Jul 2009) | 15 lines Merged revisions 207945 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21 Jul 2009)
+ | 8 lines Force an error if a blank is passed to QUOTE (because
+ the documentation states the argument is not optional). This
+ change makes URIENCODE and QUOTE behave similarly, since the
+ documentation states that the argument is not optional, for both.
+ (closes issue #15439) Reported by: pkempgen Patches:
+ 20090706__issue15439.diff.txt uploaded by tilghman (license 14)
+ ........ ................
+
+2009-07-21 20:29 +0000 [r207784-207861] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 207854 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r207854 | jpeeler | 2009-07-21 15:26:02 -0500
+ (Tue, 21 Jul 2009) | 16 lines Merged revisions 207827 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009)
+ | 9 lines Wait for wink before dialing when using E&M wink
+ signaling There was already code for other signaling types in
+ dahdi_handle_event to handle dialing if a dial operation dial
+ string was present. Simply add SIG_EMWINK to the list. (closes
+ issue #14434) Reported by: araasch ........ ................
+
+ * channels/chan_dahdi.c: Revert r207637, this approach could
+ potentially block for an unacceptable amount of time.
+
+2009-07-21 14:31 +0000 [r207726] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c, /: Merged revisions 207723 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r207723 | mmichelson | 2009-07-21 09:29:40 -0500 (Tue, 21 Jul
+ 2009) | 11 lines Merged revisions 207714 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207714 | mmichelson | 2009-07-21 09:26:00 -0500 (Tue, 21 Jul
+ 2009) | 5 lines Document default timeout for AMI originations.
+ AST-224 ........ ................
+
+2009-07-21 13:48 +0000 [r207684] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/Makefile, doc/video_console.txt, Makefile, agi/Makefile,
+ codecs/Makefile, utils/Makefile, funcs/Makefile,
+ codecs/lpc10/Makefile, main/db1-ast/Makefile, /, main/Makefile,
+ codecs/gsm/Makefile, Makefile.moddir_rules, Makefile.rules,
+ pbx/Makefile, res/Makefile: Merged revisions 207680 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r207680 | kpfleming | 2009-07-21 08:28:04 -0500
+ (Tue, 21 Jul 2009) | 18 lines Merged revisions 207647 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul
+ 2009) | 12 lines Ensure that user-provided CFLAGS and LDFLAGS are
+ honored. This commit changes the build system so that
+ user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to
+ the compiler/linker *after* all flags provided by the build
+ system itself, so that the user can effectively override the
+ build system's flags if desired. In addition, ASTCFLAGS and
+ ASTLDFLAGS can now be provided *either* in the environment before
+ running 'make', or as variable assignments on the 'make' command
+ line. As a result, the use of COPTS and LDOPTS is no longer
+ necessary, so they are no longer documented, but are still
+ supported so as not to break existing build systems that supply
+ them when building Asterisk. ........ ................
+
+2009-07-21 04:45 +0000 [r207637] 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 20:02 +0000 [r207426] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 207424 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r207424 | mmichelson | 2009-07-20 14:48:12 -0500 (Mon, 20 Jul
+ 2009) | 39 lines Merged revisions 207423 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul
+ 2009) | 33 lines Answer video SDP offers properly when
+ videosupport is not enabled. Copied from Review board: In issue
+ 12434, the reporter describes a situation in which audio and
+ video is offered on the call, but because videosupport is
+ disabled in sip.conf, Asterisk gives no response at all to the
+ video offer. According to RFC 3264, all media offers should have
+ a corresponding answer. For offers we do not intend to actually
+ reply to with meaningful values, we should still reply with the
+ port for the media stream set to 0. In this patch, we take note
+ of what types of media have been offered and save the information
+ on the sip_pvt. The SDP in the response will take into account
+ whether media was offered. If we are not otherwise going to
+ answer a media offer, we will insert an appropriate m= line with
+ the port set to 0. It is important to note that this patch is
+ pretty much a bandage being applied to a broken bone. The patch
+ *only* helps for situations where video is offered but
+ videosupport is disabled and when udptl_pt is disabled but T.38
+ is offered. Asterisk is not guaranteed to respond to every media
+ offer. Notable cases are when multiple streams of the same type
+ are offered. The 2 media stream limit is still present with this
+ patch, too. In trunk and the 1.6.X branches, things will be a bit
+ different since Asterisk also supports text in SDPs as well.
+ (closes issue #12434) Reported by: mnnojd Review:
+ https://reviewboard.asterisk.org/r/311 Review:
+ https://reviewboard.asterisk.org/r/313 ........ ................
+
+2009-07-20 16:40 +0000 [r207363] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 207361 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r207361 | russell | 2009-07-20 11:36:15 -0500 (Mon, 20 Jul 2009)
+ | 16 lines Merged revisions 207360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009)
+ | 9 lines Only do the chan->fdno check in ast_read() in a
+ developer build. I changed this check to only happen in a
+ dev-mode build. I also added a comment explaining what is going
+ on. I also made it so that detection of this situation does not
+ affect ast_read() operation. (closes issue #14723) Reported by:
+ seadweller ........ ................
+
+2009-07-18 04:17 +0000 [r207321] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Recorded merge of revisions 207317 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r207317 | tilghman | 2009-07-17 23:16:44 -0500 (Fri, 17
+ Jul 2009) | 3 lines Flag field in wrong position. Reported by
+ "Hoggins!" on asterisk-dev list. ........
+
+2009-07-18 02:09 +0000 [r207287] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/misdn_config.c,
+ channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h,
+ doc/tex/misdn.tex, channels/chan_misdn.c, main/callerid.c,
+ configs/misdn.conf.sample: Merged revisions 145293,158010 from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 to make
+ merging easier. These changes are already on trunk.
+ ................ r145293 | rmudgett | 2008-09-30 18:55:24 -0500
+ (Tue, 30 Sep 2008) | 54 lines channels/chan_misdn.c
+ channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk
+ to make merging easier later. ........ r145200 | rmudgett |
+ 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines *
+ Miscellaneous formatting changes to make v1.4 and trunk more
+ merge compatible in the mISDN area. channels/chan_misdn.c *
+ Eliminated redundant code in cb_events() EVENT_SETUP ........
+ r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008)
+ | 9 lines improved helptext of misdn_set_opt. ........ r142181 |
+ rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line
+ Cleaned up comment ........ r138738 | rmudgett | 2008-08-18
+ 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines
+ channels/chan_misdn.c * Made bearer2str() use
+ allowed_bearers_array[] * Made use the causes.h defines instead
+ of hardcoded numbers. * Made use Asterisk presentation indicator
+ values if either of the mISDN presentation or screen options are
+ negative. * Updated the misdn_set_opt application option
+ descriptions. * Renamed the awkward Caller ID presentation
+ misdn_set_opt application option value not_screened to
+ restricted. Deprecated the not_screened option value.
+ channels/misdn/isdn_lib.c * Made use the causes.h defines instead
+ of hardcoded numbers. * Fixed some spelling errors and typos. *
+ Added all defined facility code strings to fac2str().
+ channels/misdn/isdn_lib.h * Added doxygen comments to struct
+ misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen
+ comments to struct misdn_stack. channels/misdn_config.c
+ configs/misdn.conf.sample * Updated the mISDN presentation and
+ screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex)
+ * Updated the misdn_set_opt application option descriptions. *
+ Fixed some spelling errors and typos. ................ r158010 |
+ rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines
+ Merged revision 157977 from
+ https://origsvn.digium.com/svn/asterisk/team/group/issue8824
+ ........ Fixes JIRA ABE-1726 The dial extension could be empty if
+ you are using MISDN_KEYPAD to control ISDN provider features.
+ ................
+
+2009-07-17 22:30 +0000 [r207227-207256] Tilghman Lesher <tlesher@digium.com>
+
+ * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 207255 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r207255 | tilghman | 2009-07-17 17:29:50 -0500 (Fri, 17
+ Jul 2009) | 2 lines Add flag here, too (as requested by jsmith)
+ ........
+
+ * /, doc/tex/odbcstorage.tex, UPGRADE.txt: Recorded merge of
+ revisions 207224 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r207224 |
+ tilghman | 2009-07-17 17:04:43 -0500 (Fri, 17 Jul 2009) | 2 lines
+ Document the "flag" field in the voicemessages table. ........
+
+2009-07-17 19:39 +0000 [r207101-207158] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 207156 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r207156 | jpeeler | 2009-07-17 14:37:38 -0500
+ (Fri, 17 Jul 2009) | 14 lines Merged revisions 207155 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207155 | jpeeler | 2009-07-17 14:36:19 -0500 (Fri, 17 Jul 2009)
+ | 7 lines Fix format specifier to print out an unsigned long
+ long. Yep, it's even ifdefed out code. But it made it to the RR
+ list... (closes issue #14726) Reported by: lmadsen ........
+ ................
+
+ * configs/chan_dahdi.conf.sample, /: Merged revisions 207095 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r207095 | jpeeler | 2009-07-17 14:16:35 -0500 (Fri, 17
+ Jul 2009) | 2 lines Update some missing allowed options for
+ overlapdial ........
+
+2009-07-17 17:53 +0000 [r206870-207031] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 207029 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r207029 |
+ dvossel | 2009-07-17 12:51:44 -0500 (Fri, 17 Jul 2009) | 6 lines
+ sip option flags handled incorrectly (closes issue #15376)
+ Reported by: Takehiko Ooshima Tested by: dvossel,
+ Takehiko_Ooshima ........
+
+ * /, channels/chan_sip.c: Merged revisions 206939 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r206939 | dvossel | 2009-07-17 11:13:22 -0500 (Fri, 17 Jul 2009)
+ | 20 lines Merged revisions 206938 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009)
+ | 14 lines SIP incorrect From: header information when callpres
+ is prohib Some ITSP make use of the "Anonymous" display name to
+ detect a requirement to withhold caller id across the PSTN. This
+ does not work if the display name is "Unknown". (closes issue
+ #14465) Reported by: Nick_Lewis Patches:
+ chan_sip.c-callerpres.patch uploaded by Nick (license 657)
+ chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license
+ 671) Tested by: Nick_Lewis, dvossel ........ ................
+
+ * /, funcs/func_timeout.c: Merged revisions 206877 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r206877 | dvossel | 2009-07-16 16:45:14 -0500 (Thu, 16 Jul 2009)
+ | 6 lines TIMEOUT(absolute) returned negative value. (closes
+ issue #15513) Reported by: ys ........
+
+ * configs/iax.conf.sample, /: Merged revisions 206873 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r206873 | dvossel | 2009-07-16 16:33:51 -0500
+ (Thu, 16 Jul 2009) | 12 lines Merged revisions 206872 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16 Jul 2009)
+ | 6 lines error in iax.conf related IP-based access control
+ (closes issue #15518) Reported by: pkempgen ........
+ ................
+
+ * /, main/callerid.c: Merged revisions 206868 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r206868 | dvossel | 2009-07-16 16:25:22 -0500 (Thu, 16 Jul 2009)
+ | 14 lines Merged revisions 206867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206867 | dvossel | 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009)
+ | 8 lines avoid segfault caused by user error If the CALLERPRES()
+ dialplan function is set to nothing, a segfault occurs. This is
+ user error to begin with, but I'd rather see a cli warning
+ message than have Asterisk crash on me. ........ ................
+
+2009-07-16 16:53 +0000 [r206810] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_realtime.c, /: Merged revisions 206808 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r206808 | tilghman | 2009-07-16 11:51:05 -0500
+ (Thu, 16 Jul 2009) | 13 lines Merged revisions 206807 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206807 | tilghman | 2009-07-16 11:27:35 -0500 (Thu, 16 Jul 2009)
+ | 6 lines Fix a memory leak. (closes issue #15517) Reported by:
+ adomjan Patches: func_realtime.c-ast_variable_destroy.diff
+ uploaded by adomjan (license 487) ........ ................
+
+2009-07-15 22:06 +0000 [r206774] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 206768 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r206768 |
+ dvossel | 2009-07-15 17:04:13 -0500 (Wed, 15 Jul 2009) | 8 lines
+ Session timer were not activated if Supported header field in
+ INVITE had both "timer" and other options. (closes issue #15403)
+ Reported by: makoto Patches: sip-session-timer.patch uploaded by
+ makoto (license ........
+
+2009-07-15 21:40 +0000 [r206764] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /:
+ Merged revisions 206707 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r206707 | rmudgett | 2009-07-15 16:14:41 -0500 (Wed, 15 Jul 2009)
+ | 33 lines Merged revisions 206706 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r206706 | rmudgett | 2009-07-15 15:44:55 -0500
+ (Wed, 15 Jul 2009) | 26 lines Merged revision 206700 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
+ .......... Fixed chan_misdn crash because mISDNuser library is
+ not thread safe. With Asterisk the mISDNuser library is driven by
+ two threads concurrently: 1.
+ channels/misdn/isdn_lib.c::manager_event_handler() 2.
+ channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher() Calls
+ into the library are done concurrently and recursively from
+ isdn_lib.c. Both threads can fiddle with the master/child
+ layer3_proc_t lists. One thread may traverse the list when the
+ other interrupts it and then removes the list element which the
+ first thread was currently handling. This is exactly what caused
+ the crash. About 60 calls were needed to a Gigaset CX475 before
+ it occurred once. This patch adds locking when calling into the
+ mISDNuser library. This also fixes some cb_log calls with wrong
+ port parameter. JIRA ABE-1913 Patches: misdn-locking.patch
+ (Modified with mostly cosmetic changes) ..........
+ ................ ................
+
+2009-07-15 20:21 +0000 [r206704] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 206702 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r206702 |
+ dvossel | 2009-07-15 15:20:01 -0500 (Wed, 15 Jul 2009) | 10 lines
+ callerid(num) is wrong when username is missing A domain only sip
+ uri <sip:123.123.123.123> would return 123.123.123.123 as callid
+ num. Now, if the username is missing from a uri, the callerid num
+ field is left empty. (closes issue #15476) Reported by: viraptor
+ ........
+
+2009-07-15 16:03 +0000 [r206638] Sean Bright <sean@malleable.com>
+
+ * /, codecs/codec_dahdi.c: Merged revisions 206636 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r206636 | seanbright | 2009-07-15 12:00:24 -0400
+ (Wed, 15 Jul 2009) | 9 lines Merged revisions 206635 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed,
+ 15 Jul 2009) | 1 line Only print debug info in codec_dahdi if we
+ are asking for it. ........ ................
+
+2009-07-14 20:25 +0000 [r206596] Tilghman Lesher <tlesher@digium.com>
+
+ * /, contrib/scripts/meetme.sql: Recorded merge of revisions 206567
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r206567 | tilghman | 2009-07-14 15:14:45 -0500 (Tue, 14
+ Jul 2009) | 6 lines Document all meetme realtime fields, and in
+ the process, make some field lengths more consistent. (closes
+ issue #15493) Reported by: lasko Patches: meetme.diff uploaded by
+ lasko (license 833) ........
+
+2009-07-14 18:32 +0000 [r206558] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
+ channels/chan_misdn.c, /: Merged revisions 206489 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r206489 | rmudgett | 2009-07-14 12:01:48 -0500
+ (Tue, 14 Jul 2009) | 35 lines Merged revisions 206487 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14 Jul 2009)
+ | 28 lines Fixes several call transfer issues with chan_misdn. *
+ issue #14355 - Crash if attempt to transfer a call to an
+ application. Masquerade the other pair of the four asterisk
+ channels involved in the two calls. The held call already must be
+ a bridged call (not an applicaton) or it would have been
+ rejected. * issue #14692 - Held calls are not automatically
+ cleared after transfer. Allow the core to initate disconnect of
+ held calls to the ISDN port. This also fixes a similar case where
+ the party on hold hangs up before being transferred or taken off
+ hold. * JIRA ABE-1903 - Orphaned held calls left in
+ music-on-hold. Do not simply block passing the hangup event on
+ held calls to asterisk core. * Fixed to allow held calls to be
+ transferred to ringing calls. Previously, held calls could only
+ be transferred to connected calls. * Eliminated unused call
+ states to simplify hangup code. * Eliminated most uses of
+ "holded" because it is not a word. (closes issue #14355) (closes
+ issue #14692) Reported by: sodom Patches:
+ misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664)
+ Tested by: rmudgett ........ ................
+
+2009-07-14 14:56 +0000 [r206388] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 206386 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r206386 | russell | 2009-07-14 09:51:44 -0500
+ (Tue, 14 Jul 2009) | 20 lines Merged revisions 206385 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r206385 | russell | 2009-07-14 09:48:00 -0500
+ (Tue, 14 Jul 2009) | 13 lines Merged revisions 206384 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009)
+ | 6 lines Ensure apathetic replies are sent out on the proper
+ socket. chan_iax2 supports multiple address bindings. The
+ send_apathetic_reply() function did not attempt to send its
+ response on the same socket that the incoming message came in on.
+ ........ ................ ................
+
+2009-07-14 01:35 +0000 [r206372] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
+ revisions 206341 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r206341 | rmudgett | 2009-07-13 19:48:59 -0500 (Mon, 13 Jul 2009)
+ | 11 lines Merged revisions 206284 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009)
+ | 4 lines Fix some memory leaks in chan_misdn. JIRA ABE-1911
+ ........ ................
+
+2009-07-13 23:33 +0000 [r206282] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 206280 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r206280 |
+ dvossel | 2009-07-13 18:26:51 -0500 (Mon, 13 Jul 2009) | 9 lines
+ dns lookup of peername rather than peer's host in
+ transmit_register() (closes issue #15052) Reported by: fsantulli
+ Patches: chan_sip_bug_15052_[20090626204511].patch uploaded by
+ fsantulli (license 818) Tested by: fsantulli ........
+
+2009-07-13 16:24 +0000 [r206186] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 206185 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r206185 | tilghman | 2009-07-13 11:23:07 -0500 (Mon, 13 Jul 2009)
+ | 2 lines Remove reference to non-existent help file ........
+
+2009-07-10 21:52 +0000 [r205987] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 205985 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205985 |
+ dvossel | 2009-07-10 16:42:10 -0500 (Fri, 10 Jul 2009) | 16 lines
+ SIP register not using peer's outbound proxy If callbackextension
+ is defined for a peer it successfully causes a registration to
+ occur, but the registration ignores the outboundproxy settings
+ for the peer. This patch allows the peer to be passed to
+ obproxy_get() in transmit_register(). (closes issue #14344)
+ Reported by: Nick_Lewis Patches:
+ callbackextension_peer_trunk.diff uploaded by dvossel (license
+ 671) Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/294/ ........
+
+2009-07-10 18:45 +0000 [r205941] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, /: Merged revisions 205939 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205939 |
+ kpfleming | 2009-07-10 13:44:09 -0500 (Fri, 10 Jul 2009) | 1 line
+ Update comments about the level of T.38 support in Asterisk.
+ ........
+
+2009-07-10 17:50 +0000 [r205881] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 205878 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205878 | mmichelson | 2009-07-10 12:39:57 -0500 (Fri, 10 Jul
+ 2009) | 30 lines Merged revisions 205877 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r205877 | mmichelson | 2009-07-10 12:39:13 -0500
+ (Fri, 10 Jul 2009) | 23 lines Merged revisions 205776 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500
+ (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul
+ 2009) | 10 lines Ensure that outbound NOTIFY requests are
+ properly routed through stateful proxies. With this change, we
+ make note of Record-Route headers present in any SUBSCRIBE
+ request that we receive so that our outbound NOTIFY requests will
+ have the proper Route headers in them. (closes issue #14725)
+ Reported by: ibc ........ ................ ................
+ ................
+
+2009-07-10 16:48 +0000 [r205842] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 205840 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205840 | dvossel | 2009-07-10 11:42:04 -0500 (Fri, 10 Jul 2009)
+ | 37 lines Merged revisions 205804 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009)
+ | 31 lines SIP registration auth loop caused by stale nonce If an
+ endpoint sends two registration requests in a very short period
+ of time with the same nonce, both receive 401 responses from
+ Asterisk, each with a different nonce (the second 401 containing
+ the current nonce and the first one being stale). If the endpoint
+ responds to the first 401, it does not match the current nonce so
+ Asterisk sends a third 401 with a newly generated nonce (which
+ updates the current nonce)... Now if the endpoint responds to the
+ second 401, it does not match the current nonce either and
+ Asterisk sends a fourth 401 with a newly generated nonce... This
+ loop goes on and on. There appears to be a simple fix for this.
+ If the nonce from the request does not match our nonce, but is a
+ good response to a previous nonce, instead of sending a 401 with
+ a newly generated nonce, use the current one instead. This breaks
+ the loop as the nonce is not updated until a response is
+ received. Additional logic has been added to make sure no nonce
+ can be responded to twice though. (closes issue #15102) Reported
+ by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license
+ 809) nonce_sip.diff uploaded by dvossel (license 671) Tested by:
+ Jamuel Review: https://reviewboard.asterisk.org/r/289/ ........
+ ................
+
+2009-07-10 15:57 +0000 [r205778] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 205776 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul
+ 2009) | 16 lines Merged revisions 205775 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul
+ 2009) | 10 lines Ensure that outbound NOTIFY requests are
+ properly routed through stateful proxies. With this change, we
+ make note of Record-Route headers present in any SUBSCRIBE
+ request that we receive so that our outbound NOTIFY requests will
+ have the proper Route headers in them. (closes issue #14725)
+ Reported by: ibc ........ ................
+
+2009-07-10 15:36 +0000 [r205772] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 205770 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205770 |
+ kpfleming | 2009-07-10 10:28:11 -0500 (Fri, 10 Jul 2009) | 12
+ lines Fix some remaining T.38 negotiation problems in app_fax.
+ Revision 205696 did not quite fix all the issues with the T.38
+ negotiation changes and app_fax; this patch corrects them, along
+ with a couple of other minor issues. (closes issue #15480)
+ Reported by: dimas Patches: test2-15480.patch uploaded by dimas
+ (license 88) ........
+
+2009-07-09 23:51 +0000 [r205730] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Merged revisions 205728 via svn merge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009)
+ | 21 lines 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 21:27 +0000 [r205698] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/frame.h, /, channels/chan_sip.c, apps/app_fax.c:
+ Merged revisions 205696 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205696 |
+ kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16
+ lines Repair ability of SendFAX/ReceiveFAX to respond to T.38
+ switchover. Recent changes in T.38 negotiation in Asterisk caused
+ these applications to not respond when the other endpoint
+ initiated a switchover to T.38; this resulted in the T.38
+ switchover failing, and the FAX attempt to be made using an audio
+ connection, instead of T.38 (which would usually cause the FAX to
+ fail completely). This patch corrects this problem, and the
+ applications will now correctly respond to the T.38 switchover
+ request. In addition, the response will include the appopriate
+ T.38 session parameters based on what the other end offered and
+ what our end is capable of. (closes issue #14849) Reported by:
+ afosorio ........
+
+2009-07-09 16:20 +0000 [r205596-205605] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/time.h, /: Merged revisions 205600 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r205600 | dvossel | 2009-07-09 11:19:09 -0500
+ (Thu, 09 Jul 2009) | 9 lines Merged revisions 205599 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09
+ Jul 2009) | 2 lines Changing ast_samp2tv to not use floating
+ point. ........ ................
+
+ * channels/chan_iax2.c, include/asterisk/frame.h, main/rtp.c, /:
+ Merged revisions 205479 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205479 | dvossel | 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009)
+ | 16 lines Merged revisions 205471 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009)
+ | 10 lines Fixes 8khz assumptions Many calculations assume 8khz
+ is the codec rate. This is not always the case. This patch only
+ addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there
+ are other areas that make this assumption as well. Review:
+ https://reviewboard.asterisk.org/r/306/ ........ ................
+
+2009-07-09 08:33 +0000 [r205534] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, main/ssl.c: Merged revisions 205532 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205532 |
+ mvanbaak | 2009-07-09 10:31:24 +0200 (Thu, 09 Jul 2009) | 5 lines
+ pthread_self returns a pthread_t which is not an unsigned int on
+ all pthread implementations. Casting it to an unsigned int fixes
+ compiler warnings. Tested on OpenBSD and Linux both 32 and 64 bit
+ ........
+
+2009-07-08 22:16 +0000 [r205414] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/devicestate.h, main/pbx.c, /,
+ main/devicestate.c, include/asterisk/pbx.h: Merged revisions
+ 205412 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205412 | dvossel | 2009-07-08 17:15:06 -0500 (Wed, 08 Jul 2009)
+ | 12 lines Merged revisions 205409 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009)
+ | 6 lines moving ast_devstate_to_extenstate to pbx.c from
+ devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This
+ change fixes a compile time error with chan_vpb as well. ........
+ ................
+
+2009-07-08 19:27 +0000 [r205352] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 205350 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205350 | mmichelson | 2009-07-08 14:26:55 -0500 (Wed, 08 Jul
+ 2009) | 20 lines Merged revisions 205349 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul
+ 2009) | 14 lines Prevent phantom calls to queue members. If a
+ caller were to hang up while a periodic announcement or position
+ were being said, the return value for those functions would
+ incorrectly indicate that the caller was still in the queue. With
+ these changes, the problem does not occur. (closes issue #14631)
+ Reported by: latinsud Patches: queue_announce_ghost_call2.diff
+ uploaded by latinsud (license 745) (with small modification from
+ me) ........ ................
+
+2009-07-08 18:21 +0000 [r205299] Jason Parker <jparker@digium.com>
+
+ * config.guess, config.sub, /: Merged revisions 205291 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r205291 | qwell | 2009-07-08 13:19:46 -0500
+ (Wed, 08 Jul 2009) | 9 lines Merged revisions 205288 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul
+ 2009) | 1 line Update config.guess and config.sub from the
+ savannah.gnu.org git repo. ........ ................
+
+2009-07-08 18:07 +0000 [r205279] David Brooks <dbrooks@digium.com>
+
+ * /, main/features.c: Merged revisions 205254 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205254 |
+ dbrooks | 2009-07-08 12:26:26 -0500 (Wed, 08 Jul 2009) | 8 lines
+ Fixes Park() argument handling Park() was not respecting the
+ arguments passed to it. Any extension/context/priority given to
+ it was being ignored. This patch remedies this. (closes issue
+ #15380) Reported by: DLNoah ........
+
+2009-07-08 16:59 +0000 [r205222] Tilghman Lesher <tlesher@digium.com>
+
+ * main/say.c: oops, fixing build
+
+2009-07-08 16:56 +0000 [r205218] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/time.h, /: Merged revisions 205216 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r205216 | dvossel | 2009-07-08 11:54:24 -0500
+ (Wed, 08 Jul 2009) | 17 lines Merged revisions 205215 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009)
+ | 10 lines ast_samp2tv needs floating point for 16khz audio In
+ ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000. The
+ .5 is currently stripped off because we don't calculate using
+ floating points. This causes madness with 16khz audio. (issue
+ ABE-1899) Review: https://reviewboard.asterisk.org/r/305/
+ ........ ................
+
+2009-07-08 16:29 +0000 [r205203] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/say.c: Merged revisions 205196 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205196 | tilghman | 2009-07-08 11:27:50 -0500 (Wed, 08 Jul 2009)
+ | 9 lines Merged revisions 205188 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205188 | tilghman | 2009-07-08 11:26:15 -0500 (Wed, 08 Jul 2009)
+ | 2 lines Add redirection warnings for the invalid language codes
+ previously removed. ........ ................
+
+2009-07-08 15:57 +0000 [r205147-205153] Russell Bryant <russell@digium.com>
+
+ * /, main/ssl.c: Merged revisions 205151 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205151 |
+ russell | 2009-07-08 10:56:28 -0500 (Wed, 08 Jul 2009) | 2 lines
+ Use tabs instead of spaces for indentation. ........
+
+ * res/res_jabber.c, main/asterisk.c, /, main/Makefile,
+ res/res_crypto.c, main/ssl.c (added),
+ include/asterisk/_private.h: Merged revisions 205120 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r205120 | russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009)
+ | 16 lines Move OpenSSL initialization to a single place, make
+ library usage thread-safe. While doing some reading about
+ OpenSSL, I noticed a couple of things that needed to be improved
+ with our usage of OpenSSL. 1) We had initialization of the
+ library done in multiple modules. This has now been moved to a
+ core function that gets executed during Asterisk startup. We
+ already link OpenSSL into the core for TCP/TLS functionality, so
+ this was the most logical place to do it. 2) OpenSSL is not
+ thread-safe by default. However, making it thread safe is very
+ easy. We just have to provide a couple of callbacks. One callback
+ returns a thread ID. The other handles locking. For more
+ information, start with the "Is OpenSSL thread-safe?" question on
+ the FAQ page of openssl.org. ........
+
+2009-07-06 14:24 +0000 [r204976] Ryan Brindley <rbrindley@digium.com>
+
+ * main/config.c, /: Merged revisions 202753 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202753 |
+ rbrindley | 2009-06-23 16:25:17 -0500 (Tue, 23 Jun 2009) | 9
+ lines If we delete the info, lets also delete the lines (closes
+ issue 0014509) Reported by: timeshell Patches:
+ 20090504__bug14509.diff.txt uploaded by tilghman (license 14)
+ Tested by: awk, timeshell ........
+
+2009-07-06 13:40 +0000 [r204950] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/channel.c, /: Merged revisions 204948 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r204948 |
+ kpfleming | 2009-07-06 08:38:29 -0500 (Mon, 06 Jul 2009) | 7
+ lines Improve handling of AST_CONTROL_T38 and
+ AST_CONTROL_T38_PARAMETERS for non-T.38-capable channels. This
+ change allows applications that request T.38 negotiation on a
+ channel that does not support it to get the proper indication
+ that it is not supported, rather than thinking that negotiation
+ was started when it was not. ........
+
+2009-07-02 22:05 +0000 [r204837] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 204835 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r204835 | rmudgett | 2009-07-02 17:01:28 -0500
+ (Thu, 02 Jul 2009) | 17 lines Merged revisions 204834 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009)
+ | 10 lines Removed confusing warning message "Got Busy in
+ Connected State" If an incoming mISDN call is answered with the
+ Answer application and a subsequent Dial gets a busy endpoint
+ then it is valid for that already connected channel to get the
+ busy indication. Asterisk will play the busy tones until the
+ dialplan plays something else or hangs up the call. (closes issue
+ #11974) Reported by: fvdb ........ ................
+
+2009-07-02 16:28 +0000 [r204736] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/devicestate.h, main/pbx.c, /,
+ main/devicestate.c: Merged revisions 204710 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r204710 | dvossel | 2009-07-02 11:03:44 -0500 (Thu, 02 Jul 2009)
+ | 21 lines Merged revisions 204681 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009)
+ | 14 lines Improved mapping of extension states from combined
+ device states. This fixes a few issues with incorrect extension
+ states and adds a cli command, core show device2extenstate, to
+ display all possible state mappings. (closes issue #15413)
+ Reported by: legart Patches: exten_helper.diff uploaded by
+ dvossel (license 671) Tested by: dvossel, legart, amilcar Review:
+ https://reviewboard.asterisk.org/r/301/ ........ ................
+
+2009-06-30 21:30 +0000 [r204612] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/say.c, UPGRADE.txt: Merged revisions 204563 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r204563 | tilghman | 2009-06-30 15:41:04 -0500
+ (Tue, 30 Jun 2009) | 13 lines Merged revisions 204556 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009)
+ | 6 lines More incorrect language codes, plus ensuring that
+ regionalizations use the specified language, and not English for
+ grammar. (closes issue #15022) Reported by: greenfieldtech
+ Patches: 20090519__issue15022.diff.txt uploaded by tilghman
+ (license 14) ........ ................
+
+2009-06-30 18:52 +0000 [r204477] Jason Parker <jparker@digium.com>
+
+ * /, main/say.c: Merged revisions 204475 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r204475 | qwell | 2009-06-30 13:48:35 -0500 (Tue, 30 Jun 2009) |
+ 9 lines Merged revisions 204474 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204474 | qwell | 2009-06-30 13:47:06 -0500 (Tue, 30 Jun 2009) |
+ 1 line Fix ast_say_counted_noun to correctly handle Polish. Fix a
+ comment typo in passing. ........ ................
+
+2009-06-30 18:44 +0000 [r204472] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /, main/say.c, UPGRADE.txt: Recorded merge
+ of revisions 204470 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r204470 | tilghman | 2009-06-30 13:36:24 -0500 (Tue, 30 Jun 2009)
+ | 18 lines Recorded merge of revisions 204469 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009)
+ | 11 lines "tw" is the language specification for Twi (from
+ Ghana) not Taiwanese. (closes issue #15346) Reported by: volivier
+ Patches: 20090617__issue15346__1.4.diff.txt uploaded by tilghman
+ (license 14) 20090617__issue15346__trunk.diff.txt uploaded by
+ tilghman (license 14) 20090617__issue15346__1.6.0.diff.txt
+ uploaded by tilghman (license 14)
+ 20090617__issue15346__1.6.1.diff.txt uploaded by tilghman
+ (license 14) 20090617__issue15346__1.6.2.diff.txt uploaded by
+ tilghman (license 14) Tested by: volivier ........
+ ................
+
+2009-06-29 22:53 +0000 [r204249-204303] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 204301 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r204301 | mmichelson | 2009-06-29 17:50:35 -0500 (Mon, 29 Jun
+ 2009) | 15 lines Merged revisions 204300 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun
+ 2009) | 9 lines Add error message so that it is clear why a SIP
+ peer was not processed when a DNS lookup fails on a host or
+ outboundproxy. (closes issue #13432) Reported by: p_lindheimer
+ Patches: outboundproxy.patch uploaded by p (license 558) ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 204247 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r204247 | mmichelson | 2009-06-29 16:48:54 -0500 (Mon, 29 Jun
+ 2009) | 32 lines Merged revisions 204243,204246 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun
+ 2009) | 22 lines Fix a problem where chan_sip would ignore "old"
+ but valid responses. chan_sip has had a problem for quite a long
+ time that would manifest when Asterisk would send multiple SIP
+ responses on the same dialog before receiving a response. The
+ problem occurred because chan_sip only kept track of the highest
+ outgoing sequence number used on the dialog. If Asterisk sent two
+ requests out, and a response arrived for the first request sent,
+ then Asterisk would ignore the response. The result was that
+ Asterisk would continue retransmitting the requests and ignoring
+ the responses until the maximum number of retransmissions had
+ been reached. The fix here is to rearrange the code a bit so that
+ instead of simply comparing the sequence number of the response
+ to our latest outgoing sequence number, we walk our list of
+ outstanding packets and determine if there is a match. If there
+ is, we continue. If not, then we ignore the response. In doing
+ this, I found a few completely useless variables that I have now
+ removed. (closes issue #11231) Reported by: flefoll Review:
+ https://reviewboard.asterisk.org/r/298 ........ r204246 |
+ mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3
+ lines Fix build oops. ........ ................
+
+2009-06-27 01:18 +0000 [r203918] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 203909 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r203909 | rmudgett | 2009-06-26 20:07:52 -0500
+ (Fri, 26 Jun 2009) | 23 lines Merged revisions 203908 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009)
+ | 16 lines The ISDN CPE side should not exclusively pick B
+ channels normally. Before this patch, Asterisk unconditionally
+ picked B channels exclusively on the CPE side and normally
+ allowed alternative B channels on the network side. Now Asterisk
+ does the opposite. Reasons for the CPE side to normally not pick
+ B channels exclusively: * For CPE point-to-multipoint mode (i.e.
+ phone side), the CPE side does not have enough information to
+ exclusively pick B channels. (There may be other devices on the
+ line.) * Q.931 gives preference to the network side picking B
+ channels. * Some telcos require the CPE side to not pick B
+ channels exclusively. (closes issue #14383) Reported by:
+ mbrancaleoni ........ ................
+
+2009-06-26 22:13 +0000 [r203856] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 203853 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r203853 | jpeeler | 2009-06-26 17:11:31 -0500
+ (Fri, 26 Jun 2009) | 12 lines Merged revisions 203848 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203848 | jpeeler | 2009-06-26 17:09:19 -0500 (Fri, 26 Jun 2009)
+ | 5 lines Make sure to recreate the dahdi pseudo channel after
+ dahdi restart (closes issue #14477) Reported by: timking ........
+ ................
+
+2009-06-26 21:26 +0000 [r203781-203823] Russell Bryant <russell@digium.com>
+
+ * /, main/file.c: Merged revisions 203802 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r203802 | russell | 2009-06-26 16:21:48 -0500 (Fri, 26 Jun 2009)
+ | 22 lines Merged revisions 203785 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009)
+ | 15 lines Don't fast forward past the end of a message. This is
+ nice change for users of the voicemail application. If someone
+ gets a little carried away with fast forwarding through a
+ message, they can easily get to the end and accidentally exit the
+ voicemail application by hitting the fast forward key during the
+ following prompt. This adds some safety by not allowing a fast
+ forward past the end of a message. (closes issue #14554) Reported
+ by: lacoursj Patches: 21761.patch uploaded by lacoursj (license
+ 707) Tested by: lacoursj ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 203779 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r203779 |
+ russell | 2009-06-26 15:45:00 -0500 (Fri, 26 Jun 2009) | 5 lines
+ Ensure the TCP read buffer is fully initialized before handling
+ each packet. (closes issue #14452) Reported by: umberto71
+ ........
+
+2009-06-26 20:18 +0000 [r203727] David Brooks <dbrooks@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 203721 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r203721 | dbrooks | 2009-06-26 15:13:51 -0500 (Fri, 26 Jun 2009)
+ | 16 lines Fixing voicemail's error in checking max silence vs
+ min message length Max silence was represented in milliseconds,
+ yet vmminsecs (minmessage) was represented as seconds. Also, the
+ inequality was reversed. The warning, if triggered, was "Max
+ silence should be less than minmessage or you may get empty
+ messages", which should have been logged if max silence was
+ greater than minmessage, but the check was for less than. Also,
+ conforming if statement to coding guidelines. closes issue
+ #15331) Reported by: markd Review:
+ https://reviewboard.asterisk.org/r/293/ ........
+
+2009-06-26 19:56 +0000 [r203718] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: reverse whitespace change 203713 that was
+ based on looking at sig_analog (which has about a 1000 line
+ indentation change that is not worth doing here)
+
+2009-06-26 19:48 +0000 [r203714] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 203710 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r203710 | dvossel | 2009-06-26 14:47:11 -0500 (Fri, 26 Jun 2009)
+ | 7 lines moving debug message from level 0 to 1. (closes issue
+ #15404) Reported by: leobrown Patches: iax_codec_debug.patch
+ uploaded by leobrown (license 541) ........
+
+2009-06-26 19:48 +0000 [r203713] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: whitespace fix
+
+2009-06-26 19:37 +0000 [r203704] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/devicestate.h, main/pbx.c, /,
+ main/devicestate.c: Merged revisions 203702 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r203702 |
+ russell | 2009-06-26 14:31:14 -0500 (Fri, 26 Jun 2009) | 5 lines
+ Make invalid hints report Unavailable instead of Idle. (closes
+ issue #14413) Reported by: pj ........
+
+2009-06-26 19:31 +0000 [r203703] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/frame.h, main/rtp.c, main/channel.c,
+ main/frame.c, /, channels/chan_sip.c, apps/app_fax.c,
+ configs/sip.conf.sample: Merged revisions 203699 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r203699 | file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2
+ lines Improve T.38 negotiation by exchanging session parameters
+ between application and channel. ........
+
+2009-06-26 19:28 +0000 [r203700] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 203672 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r203672 | jpeeler | 2009-06-26 14:03:25 -0500 (Fri, 26 Jun 2009)
+ | 16 lines Check if polarityonanswerdelay has elapsed before
+ setting a channel as answered after a polarity reversal.
+ Previously on a polarity switch event chan_dahdi would set the
+ channel immediately as answered. This would cause problems if a
+ polarity reversal occurred when the line was picked up as the
+ dial would not have yet occurred. Now if the polarity reversal
+ occurs before delay has elapsed after coming off hook or an
+ answer, it is ignored. Also, some refactoring was done in
+ _handle_event. (closes issue #13917) Reported by: alecdavis
+ Patches: chan_dahdi.bug13917.feb09.diff2.txt uploaded by
+ alecdavis (license 585) Tested by: alecdavis ........
+
+2009-06-25 21:46 +0000 [r203446] David Vossel <dvossel@digium.com>
+
+ * main/ast_expr2.fl, main/ast_expr2.c, /: Merged revisions 203444
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r203444 | dvossel | 2009-06-25 16:45:32 -0500 (Thu, 25
+ Jun 2009) | 4 lines fixes a few redundant conditions (issue
+ #15269) ........
+
+2009-06-25 21:19 +0000 [r203393] Terry Wilson <twilson@digium.com>
+
+ * main/cli.c, /: Merged revisions 203381 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r203381 | twilson | 2009-06-25 16:15:11 -0500 (Thu, 25 Jun 2009)
+ | 11 lines Merged revisions 203380 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203380 | twilson | 2009-06-25 16:13:10 -0500 (Thu, 25 Jun 2009)
+ | 4 lines I didn't see that Mark already fixed the underlying
+ issue! Yay for removing useless code. ........ ................
+
+2009-06-25 21:07 +0000 [r203378] Russell Bryant <russell@digium.com>
+
+ * /, main/features.c: Merged revisions 203376 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r203376 | russell | 2009-06-25 16:04:55 -0500 (Thu, 25 Jun 2009)
+ | 16 lines Merged revisions 203375 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009)
+ | 9 lines Fix a case where CDR answer time could be before the
+ start time involving parking. (closes issue #13794) Reported by:
+ davidw Patches: 13794.patch uploaded by murf (license 17)
+ 13794.patch.160 uploaded by murf (license 17) Tested by: murf,
+ dbrooks ........ ................
+
+2009-06-25 19:27 +0000 [r203274] Jason Parker <jparker@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 203258 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r203258 | qwell | 2009-06-25 14:22:46 -0500 (Thu, 25 Jun 2009) |
+ 10 lines Unmute when we get a dtmfup (we muted on dtmfdown)
+ event. This would occasionally cause one-way audio when using
+ hardware DTMF detection. (closes issue #14761) Reported by:
+ tzafrir Patches: v1-14761.patch uploaded by dimas (license 88)
+ Tested by: tzafrir, dimas ........
+
+2009-06-25 16:07 +0000 [r203118] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 203116 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r203116 | russell | 2009-06-25 11:04:10 -0500 (Thu, 25 Jun 2009)
+ | 18 lines Merged revisions 203115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009)
+ | 11 lines Resolve a crash related to a T.38 reinvite race
+ condition. This change resolves a crash observed locally during
+ some T.38 testing. A call was set up using a call file, and when
+ the T.38 reinvite came in, the channel state was still
+ AST_STATE_DOWN. The reason is explained by a comment in the code
+ that previously lived in the handling of AST_STATE_RINGING. This
+ change modifies the logic to handle the same race condition for
+ any channel state that is not UP. (closes ABE-1895) ........
+ ................
+
+2009-06-24 21:22 +0000 [r203057] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 203037 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r203037 | rmudgett | 2009-06-24 16:08:55 -0500
+ (Wed, 24 Jun 2009) | 15 lines Merged revisions 203036 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24 Jun 2009)
+ | 8 lines Improved chan_dahdi.conf pritimer error checking. Valid
+ format is: pritimer=timer_name,timer_value * Fixed segfault if
+ the ',' is missing. * Completely check the range returned by
+ pri_timer2idx() to prevent possible access outside array bounds.
+ ........ ................
+
+2009-06-24 18:30 +0000 [r202969] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 202967 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202967 | mmichelson | 2009-06-24 13:29:10 -0500 (Wed, 24 Jun
+ 2009) | 9 lines Merged revisions 202966 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202966 | mmichelson | 2009-06-24 13:28:47 -0500 (Wed, 24 Jun
+ 2009) | 3 lines Use the handy UNLINK macro instead of hand-coding
+ the same thing in-line. ........ ................
+
+2009-06-24 18:10 +0000 [r202927] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 202925 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202925 |
+ file | 2009-06-24 15:08:17 -0300 (Wed, 24 Jun 2009) | 2 lines
+ Ensure the default settings are applied for T.38 when we set it
+ up for a peer. ........
+
+2009-06-23 22:11 +0000 [r202764] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 202761 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r202761 | mattf | 2009-06-23 17:08:43 -0500 (Tue, 23 Jun 2009) |
+ 1 line I could have sworn I committed this patch ages ago, but...
+ bug fix with setting NAI properly on linksets in certain
+ situations. ........
+
+2009-06-23 16:34 +0000 [r202674] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 202672 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202672 | dvossel | 2009-06-23 11:31:30 -0500 (Tue, 23 Jun 2009)
+ | 18 lines Merged revisions 202671 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202671 | dvossel | 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009)
+ | 12 lines MWI NOTIFY contains a wrong URI if Asterisk listens to
+ non-standard port and transport (closes issue #14659) Reported
+ by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt uploaded
+ by klaus3000 (license 65) mwi_port-transport_trunk.diff uploaded
+ by dvossel (license 671) Tested by: dvossel, klaus3000 Review:
+ https://reviewboard.asterisk.org/r/288/ ........ ................
+
+2009-06-22 20:18 +0000 [r202503] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 202497 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202497 | russell | 2009-06-22 15:11:04 -0500 (Mon, 22 Jun 2009)
+ | 11 lines Merged revisions 202496 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009)
+ | 4 lines Report CallerID change during a masquerade. Reported
+ by: markster ........ ................
+
+2009-06-22 16:31 +0000 [r202472] Sean Bright <sean@malleable.com>
+
+ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 202417 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r202417 | seanbright | 2009-06-22 12:09:50 -0400 (Mon, 22 Jun
+ 2009) | 4 lines Fix lock usage in cdr_sqlite3_custom to avoid
+ potential crashes during reload. Pointed out by Russell while
+ working on the CEL branch. ........
+
+2009-06-22 16:14 +0000 [r202418] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 202415 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202415 | russell | 2009-06-22 11:05:08 -0500 (Mon, 22 Jun 2009)
+ | 9 lines Merged revisions 202414 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202414 | russell | 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009)
+ | 2 lines Make Polycom subscription type override check more
+ explicit. ........ ................
+
+2009-06-22 15:41 +0000 [r202412] David Vossel <dvossel@digium.com>
+
+ * main/loader.c, /, include/asterisk/module.h: Merged revisions
+ 202410 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202410 |
+ dvossel | 2009-06-22 10:33:35 -0500 (Mon, 22 Jun 2009) | 5 lines
+ attempting to load running modules Modules placed in the priority
+ heap for loading were not properly removed from the linked list.
+ This resulted in some modules attempting to load twice. ........
+
+2009-06-22 15:10 +0000 [r202339-202345] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 202343 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202343 | mmichelson | 2009-06-22 09:58:24 -0500 (Mon, 22 Jun
+ 2009) | 36 lines Merged revisions 202341-202342 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun
+ 2009) | 26 lines Fix a situation in which Asterisk would not stop
+ retransmitting 487s. If a CANCEL were received by Asterisk, we
+ would send a 487 in response to the original INVITE and a 200 OK
+ for the CANCEL. If there were a network hiccup which caused the
+ 200 OK and the 487 to be lost, then the UA communicating with
+ Asterisk may try to retransmit its CANCEL. Asterisk's response to
+ this used to be to try sending another 487 to the canceled INVITE
+ and another 200 OK to the CANCEL. The problem here is that the
+ originally-sent 487 was sent "reliably" meaning that it will be
+ retransmitted until it is received properly. So when we receive
+ the second CANCEL it is likely that the first batch of 487s we
+ sent is still going strong and reaches the UA. The result was
+ that the second set of 487s would be retransmitted constantly
+ until the maximum number of retries had been reached. The fix for
+ this is that if we receive a second CANCEL for an INVITE, then we
+ cancel the retransmission of the first set of 487s and start a
+ second set. This causes the dialog to be terminated reasonably.
+ (closes issue #14584) Reported by: klaus3000 Patches:
+ 14584_v2.patch uploaded by mmichelson (license 60) Tested by:
+ klaus3000 ........ r202342 | mmichelson | 2009-06-22 09:44:58
+ -0500 (Mon, 22 Jun 2009) | 3 lines Remove an extra debug line
+ left from previous commit. ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 202337 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202337 | mmichelson | 2009-06-22 09:35:09 -0500 (Mon, 22 Jun
+ 2009) | 31 lines Merged revisions 202336 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun
+ 2009) | 25 lines Fix a possible infinite loop in SDP parsing
+ during glare situation. There was a while loop in
+ get_ip_and_port_from_sdp which was controlled by a call to
+ get_sdp_iterate. The loop would exit either if what we were
+ searching for was found or if the return was NULL. The problem is
+ that get_sdp_iterate never returns NULL. This means that if what
+ we were searching for was not present, the loop would run
+ infinitely. This modification of the loop fixes the problem.
+ (closes issue #15213) Reported by: schmidts (closes issue #15349)
+ Reported by: samy (closes issue #14464) Reported by: pj (closes
+ issue #15345) Reported by: aragon Patches: sip_inf_loop.patch
+ uploaded by mmichelson (license 60) Tested by: aragon ........
+ ................
+
+2009-06-21 16:15 +0000 [r202260-202264] Russell Bryant <russell@digium.com>
+
+ * cdr/cdr_manager.c, /: Merged revisions 202262 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202262 |
+ russell | 2009-06-21 11:11:48 -0500 (Sun, 21 Jun 2009) | 2 lines
+ Fix possibility of crashiness during reload in custom fields
+ handling. ........
+
+ * cdr/cdr_manager.c, /: Merged revisions 202258 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202258 |
+ russell | 2009-06-21 11:00:23 -0500 (Sun, 21 Jun 2009) | 2 lines
+ Standardize return values of load_config() so reload() doesn't
+ report an error on success. ........
+
+2009-06-20 19:14 +0000 [r202185] Sean Bright <sean@malleable.com>
+
+ * /, apps/app_fax.c: Merged revisions 202183 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202183 |
+ seanbright | 2009-06-20 15:09:47 -0400 (Sat, 20 Jun 2009) | 5
+ lines Fix version detection for API changes in spandsp. (closes
+ issue #15355) Reported by: deuffy ........
+
+2009-06-19 21:08 +0000 [r202008] 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 20:26 +0000 [r201996] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 201994 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r201994 | dvossel | 2009-06-19 15:24:37 -0500
+ (Fri, 19 Jun 2009) | 14 lines Merged revisions 201993 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201993 | dvossel | 2009-06-19 15:22:02 -0500 (Fri, 19 Jun 2009)
+ | 8 lines timestamp was being converted to host order as a short
+ rather than a long (closes issue #15361) Reported by: ffloimair
+ Patches: ts_issue.diff uploaded by dvossel (license 671) ........
+ ................
+
+2009-06-19 15:48 +0000 [r201784-201905] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_odbc.c, /: Merged revisions 201904 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r201904 | tilghman | 2009-06-19 10:47:55 -0500 (Fri, 19 Jun 2009)
+ | 4 lines Fix 2 typos and add support for wide character types.
+ Reported by Benny Amorsen via the asterisk-users mailing list.
+ http://lists.digium.com/pipermail/asterisk-users/2009-June/233622.html
+ ........
+
+ * main/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.
+
+ * /, apps/Makefile: Merged revisions 201783 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201783 |
+ tilghman | 2009-06-18 15:52:36 -0500 (Thu, 18 Jun 2009) | 6 lines
+ One of the changes in 1.6.1 was to allow app_directory to use
+ functionality within app_voicemail for directory functions. It is
+ therefore no longer necessary for app_directory to be linked
+ against the ODBC libraries (and it never was necessary for
+ app_directory to be linked against IMAP, though it was). ........
+
+2009-06-18 16:51 +0000 [r201680] David Vossel <dvossel@digium.com>
+
+ * channels/misdn/isdn_lib.c, utils/conf2ael.c, main/ast_expr2.c,
+ utils/stereorize.c, main/ast_expr2f.c, res/ael/ael_lex.c,
+ utils/ael_main.c, utils/extconf.c, channels/xpmr/xpmr.c,
+ pbx/pbx_config.c, res/res_config_ldap.c, apps/app_rpt.c,
+ main/asterisk.c, codecs/gsm/src/gsm_destroy.c, /,
+ channels/h323/ast_h323.cxx: Merged revisions 201678 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r201678 | dvossel | 2009-06-18 11:37:42 -0500 (Thu, 18 Jun 2009)
+ | 11 lines fixes some memory leaks and redundant conditions
+ (closes issue #15269) Reported by: contactmayankjain Patches:
+ patch.txt uploaded by contactmayankjain (license 740)
+ memory_leak_stuff.trunk.diff uploaded by dvossel (license 671)
+ Tested by: contactmayankjain, dvossel ........
+
+2009-06-18 15:36 +0000 [r201613] Russell Bryant <russell@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 201610 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r201610 | russell | 2009-06-18 10:27:10 -0500
+ (Thu, 18 Jun 2009) | 36 lines Merged revisions 201600 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18 Jun 2009)
+ | 29 lines Fix memory corruption and leakage related reloads of
+ non files mode MoH classes. For Music on Hold classes that are
+ not files mode, meaning that we are executing an application that
+ will feed us audio data, we use a thread to monitor the external
+ application and read audio from it. This thread also makes use of
+ the MoH class object. In the MoH class destructor, we used
+ pthread_cancel() to ask the thread to exit. Unfortunately, the
+ code did not wait to ensure that the thread actually went away.
+ What needed to be done is a pthread_join() to ensure that the
+ thread fully cleans up before we proceed. By adding this one
+ line, we resolve two significant problems: 1) Since the thread
+ was never joined, it never fully goes away. So, on every reload
+ of non-files mode MoH, an unused thread was sticking around. 2)
+ There was a race condition here where the application monitoring
+ thread could still try to access the MoH class, even though the
+ thread executing the MoH reload has already destroyed it. (issue
+ #15109) Reported by: jvandal (issue #15123) Reported by:
+ axisinternet (issue #15195) Reported by: amorsen (issue AST-208)
+ ........ ................
+
+2009-06-18 15:24 +0000 [r201601] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 201570 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201570 |
+ dvossel | 2009-06-18 10:16:05 -0500 (Thu, 18 Jun 2009) | 11 lines
+ parsing extension correctly from sip register lines If a
+ transport type was specified, but no extension, parsing of the
+ extension would return whatever was after the transport rather
+ than defaulting to 's'. (closes issue #15111) Reported by: ffs
+ Patches: chan_sip.c_register-parser.patch uploaded by ffs
+ (license 730) Tested by: ffs, dvossel ........
+
+2009-06-17 21:32 +0000 [r201532] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 201531 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r201531 | tilghman | 2009-06-17 16:31:39 -0500 (Wed, 17 Jun 2009)
+ | 7 lines Initialize additional variables, to prevent a possible
+ crash. (closes issue #15186) Reported by: ajohnson Patches:
+ 20090528__issue15186.diff.txt uploaded by tilghman (license 14)
+ Tested by: ajohnson ........
+
+2009-06-17 20:11 +0000 [r201460-201464] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 201462 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201462 |
+ mmichelson | 2009-06-17 15:10:01 -0500 (Wed, 17 Jun 2009) | 12
+ lines Fix problem with no audio due to ignoring the SDP. A recent
+ change to our SDP version comparison made audio not function on
+ some calls. This was because of a test wherein we were trying to
+ see if an unsigned value was less than 0. This is a dumb
+ comparison and arguably the compiler should have warned about it.
+ Alas, though, it slipped past. Now it's fixed by changing the
+ variable to be a signed type. Found by several developers. Tested
+ by mnicholson and dbrooks. ........
+
+ * main/channel.c, /: Merged revisions 201458 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r201458 | mmichelson | 2009-06-17 15:04:12 -0500 (Wed, 17 Jun
+ 2009) | 15 lines Merged revisions 201450 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun
+ 2009) | 9 lines Change the datastore traversal in
+ ast_do_masquerade to use a safe list traversal. It is possible
+ for datastore fixup functions to remove the datastore from the
+ list and free it. In particular, the queue_transfer_fixup in
+ app_queue does this. While I don't yet know of this causing any
+ crashes, it certainly could. Found while discussing a separate
+ issue with Brian Degenhardt. ........ ................
+
+2009-06-17 20:01 +0000 [r201448-201456] David Vossel <dvossel@digium.com>
+
+ * doc/datastores.txt, /: Merged revisions 201453 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201453 |
+ dvossel | 2009-06-17 15:00:51 -0500 (Wed, 17 Jun 2009) | 3 lines
+ ast_channel_datastore_alloc is no longer used. updating
+ datastores.txt to reflect that. ........
+
+ * apps/app_mixmonitor.c, /: Merged revisions 201445 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r201445 | dvossel | 2009-06-17 14:45:35 -0500
+ (Wed, 17 Jun 2009) | 25 lines Merged revisions 201423 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17 Jun 2009)
+ | 19 lines StopMixMonitor race condition (not giving up file
+ immediately) StopMixMonitor only indicates to the MixMonitor
+ thread to stop writing to the file. It does not guarantee that
+ the recording's file handle is available to the dialplan
+ immediately after execution. This results in a race condition. To
+ resolve this, the filestream pointer is placed in a datastore on
+ the channel. When StopMixMonitor is called, the datastore is
+ retrieved from the channel and the filestream is closed
+ immediately before returning to the dialplan. Documentation
+ indicating the use of StopMixMonitor to free files has been
+ updated as well. (closes issue #15259) Reported by: travisghansen
+ Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/283/ ........ ................
+
+2009-06-17 19:39 +0000 [r201444] David Brooks <dbrooks@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 201381 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r201381 | dbrooks | 2009-06-17 14:15:07 -0500 (Wed, 17 Jun 2009)
+ | 16 lines Merged revisions 201380 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009)
+ | 9 lines Checks for NULL sip_pvt pointer in
+ chan_sip.c->acf_channel_read() Zombie channels could be passed,
+ and chan_sip.c wasn't checking for it. Could crash Asterisk. Now
+ checking for NULL pointer. (closes issue #15330) Reported by:
+ okrief Tested by: dbrooks ........ ................
+
+2009-06-17 15:32 +0000 [r201365] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 201344 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201344 |
+ dvossel | 2009-06-17 10:20:26 -0500 (Wed, 17 Jun 2009) | 16 lines
+ SIP registry ref count error During a sip reload, the list of
+ sip_registry objects are supposed to be traversed, unlinked, and
+ destroyed, but destruction never takes place due to a ref
+ counting error. This causes a memory leak when registry items are
+ removed from sip.conf and reloaded. While the registries are
+ removed from the global list, they are not removed from the
+ scheduler. Because of this, SIP register attempts continue to be
+ sent out for the item even though it may no longer be in the
+ .conf. (closes issue #15295) Reported by: amorsen Review:
+ https://reviewboard.asterisk.org/r/282/ ........
+
+2009-06-17 12:05 +0000 [r201264] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, include/asterisk/linkedlists.h: Merged revisions 201262 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r201262 | kpfleming | 2009-06-17 07:04:17 -0500
+ (Wed, 17 Jun 2009) | 15 lines Merged revisions 201261 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun
+ 2009) | 9 lines Correct AST_LIST_APPEND_LIST behavior when list
+ to be appended is empty. When the list to be appended is empty,
+ and the list to be appended to is *not*, AST_LIST_APPEND_LIST
+ would actually cause the target list to become broken, and no
+ longer have a pointer to its last entry. This patch fixes the
+ problem. (reported by Stanislaw Pitucha on the asterisk-dev
+ mailing list) ........ ................
+
+2009-06-16 22:31 +0000 [r201225] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 201223 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201223 |
+ dvossel | 2009-06-16 17:29:30 -0500 (Tue, 16 Jun 2009) | 2 lines
+ fix issue with build_contact introduced by the "SIP trasnport
+ type issues" commit ........
+
+2009-06-16 19:42 +0000 [r200989-201096] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/frame.h, apps/app_chanspy.c,
+ apps/app_mixmonitor.c, main/channel.c, main/autoservice.c,
+ main/frame.c, /, apps/app_meetme.c, main/slinfactory.c,
+ include/asterisk/linkedlists.h, main/file.c,
+ include/asterisk/channel.h: Merged revisions 201056 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r201056 | kpfleming | 2009-06-16 13:54:30 -0500
+ (Tue, 16 Jun 2009) | 18 lines Merged revisions 200991 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun
+ 2009) | 11 lines Improve support for media paths that can
+ generate multiple frames at once. There are various media paths
+ in Asterisk (codec translators and UDPTL, primarily) that can
+ generate more than one frame to be generated when the application
+ calling them expects only a single frame. This patch addresses a
+ number of those cases, at least the primary ones to solve the
+ known problems. In addition it removes the broken TRACE_FRAMES
+ support, fixes a number of bugs in various frame-related API
+ functions, and cleans up various code paths affected by these
+ changes. https://reviewboard.asterisk.org/r/175/ ........
+ ................
+
+ * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac: Merged
+ revisions 201090 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201090 |
+ kpfleming | 2009-06-16 14:27:12 -0500 (Tue, 16 Jun 2009) | 5
+ lines Another minor fix to compiler attribute checking.
+ Defaulting to 'static' for the function scope was bad... so
+ remove it. ........
+
+ * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac: Merged
+ revisions 200985 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200985 |
+ kpfleming | 2009-06-16 11:32:36 -0500 (Tue, 16 Jun 2009) | 7
+ lines Fix problems with new compiler attribute checking in
+ configure script. The last changes to ast_gcc_attribute.m4 caused
+ some problems checking for various attributes, because the scope
+ of the symbol the attribute is applied to can be important; this
+ patch allows the scope to be specified for the check. ........
+
+2009-06-16 16:34 +0000 [r200987] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 200946 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200946 |
+ dvossel | 2009-06-16 11:03:30 -0500 (Tue, 16 Jun 2009) | 32 lines
+ SIP transport type issues What this patch addresses: 1.
+ ast_sip_ouraddrfor() by default binds to the UDP address/port
+ reguardless if the sip->pvt is of type UDP or not. Now when no
+ remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's
+ transport type, attempting to set the address and port to the
+ correct TCP/TLS bindings if necessary. 2. It is not necessary to
+ send the port number in the Contact header unless the port is
+ non-standard for the transport type. This patch fixes this and
+ removes the todo note. 3. In sip_alloc(), the default dialog
+ built always uses transport type UDP. Now sip_alloc() looks at
+ the sip_request (if present) and determines what transport type
+ to use by default. 4. When changing the transport type of a
+ sip_socket, the file descriptor must be set to -1 and in some
+ cases the tcptls_session's ref count must be decremented and set
+ to NULL. I've encountered several issues associated with this
+ process and have created a function, set_socket_transport(), to
+ handle the setting of the socket type. (closes issue #13865)
+ Reported by: st Patches: dont_add_port_if_tls.patch uploaded by
+ Kristijan (license 753) 13865.patch uploaded by mmichelson
+ (license 60) tls_port_v5.patch uploaded by vrban (license 756)
+ transport_issues.diff uploaded by dvossel (license 671) Tested
+ by: mmichelson, Kristijan, vrban, jmacz, dvossel Review:
+ https://reviewboard.asterisk.org/r/278/ ........
+
+2009-06-16 16:04 +0000 [r200947] Michiel van Baak <michiel@vanbaak.info>
+
+ * apps/app_voicemail.c, /: Merged revisions 200943 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r200943 | mvanbaak | 2009-06-16 17:51:36 +0200 (Tue, 16 Jun 2009)
+ | 9 lines add FILE_STORAGE to Voicemail Build Options Voicemail
+ can only use one storage module at the moment. Because it's
+ unclear that selecting one of the storage modules in menuselect
+ will disable filesystem storage we now have a FILE_STORAGE option
+ that conflicts with the other modules. (closes issue #15333)
+ ........
+
+2009-06-16 01:32 +0000 [r200707-200766] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in,
+ autoconf/ast_gcc_attribute.m4: Merged revisions 200764 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r200764 | kpfleming | 2009-06-15 20:28:08 -0500 (Mon, 15
+ Jun 2009) | 11 lines Ensure that configure-script testing for
+ compiler attributes actually works. The configure script tests
+ for compiler attributes didn't actually enable enough warnings or
+ provide a proper test harness to determine whether the compiler
+ supports the attribute in question or not; this caused gcc 4.1 to
+ report that it supports 'weakref', but it doesn't actually
+ support it in the way that is needed for our optional API
+ mechanism. The new configure script test will properly
+ distinguish between full support and partial support for this
+ attribute, among others. ........
+
+ * CHANGES, /: Merged revisions 200726 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200726 |
+ kpfleming | 2009-06-15 20:03:22 -0500 (Mon, 15 Jun 2009) | 6
+ lines Document the new automatic 'ignoresdpversion' behavior.
+ Asterisk will now automatically ignore incorrect incoming SDP
+ version numbers when necessary to complete a T.38 re-INVITE
+ operation. ........
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+ 165180,200689 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r165180 |
+ mnicholson | 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) | 14
+ lines This patch adds a new 'ignoresdpversion' option to
+ sip.conf. When this is enabled (either globally or for a specific
+ peer), chan_sip will treat any SDP data it receives as new data
+ and update the media stream accordingly. By default, Asterisk
+ will only modify the media stream if the SDP session version
+ received is different from the current SDP session version. This
+ option is required to interoperate with devices that have
+ non-standard SDP session version implementations (observed by toc
+ on the bug tracker with Microsoft OCS which always uses 0 as the
+ session version). http://reviewboard.digium.com/r/94/ (closes
+ issue #13958) Reported by: toc Tested by: toc ........ r200689 |
+ kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 12
+ lines Accept T.38 re-INVITE responses with invalid SDP versions.
+ This commit changes the 'incoming SDP version' check logic a bit
+ more; when 'ignoresdpversion' is *not* set for a peer, if we
+ initiate a re-INVITE to switch to T.38, we'll always accept the
+ peer's SDP response, even if they don't properly increment the
+ SDP version number as they should. If this situation occurs, a
+ warning message will be generated suggesting that the peer's
+ configuration be changed to include the 'ignoresdpversion'
+ configuration option (although ideally they'd fix their SIP
+ implementation to be RFC compliant). AST-221 ........
+
+2009-06-15 15:23 +0000 [r200516] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 200514 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r200514 | mmichelson | 2009-06-15 10:22:11 -0500 (Mon, 15 Jun
+ 2009) | 11 lines Merged revisions 200513 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun
+ 2009) | 5 lines Add INFO to our allowed methods so that endpoints
+ know they may send it to us. AST-223 ........ ................
+
+2009-06-12 19:08 +0000 [r200363] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 200361 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r200361 | mmichelson | 2009-06-12 14:07:51 -0500 (Fri, 12 Jun
+ 2009) | 16 lines Merged revisions 200360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun
+ 2009) | 10 lines Suppress a warning message and give a better
+ return code when generating inband ringing after a call is
+ answered. (closes issue #15158) Reported by: madkins Patches:
+ 15158.patch uploaded by mmichelson (license 60) Tested by:
+ madkins ........ ................
+
+2009-06-11 22:44 +0000 [r200229] Sean Bright <sean@malleable.com>
+
+ * Makefile, /: Merged revisions 199781 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r199781 |
+ seanbright | 2009-06-09 14:08:53 -0400 (Tue, 09 Jun 2009) | 2
+ lines Fix all of the parallel build warnings issued when running
+ make -j#. ........
+
+2009-06-11 21:25 +0000 [r200171] Terry Wilson <twilson@digium.com>
+
+ * main/rtp.c: Don't access rtp->rtcp->* if rtp->rtcp is null
+
+2009-06-11 21:18 +0000 [r200152] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 200146 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200146 |
+ mmichelson | 2009-06-11 16:17:14 -0500 (Thu, 11 Jun 2009) | 5
+ lines Fix a crash due to a potentially NULL p->options. Thanks to
+ mnicholson for pointing it out. ........
+
+2009-06-11 12:16 +0000 [r200041] Leif Madsen <lmadsen@digium.com>
+
+ * build_tools/make_version_h, /, build_tools/make_version_c: Merged
+ revisions 200039 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200039 |
+ lmadsen | 2009-06-11 08:15:09 -0400 (Thu, 11 Jun 2009) | 8 lines
+ Fix path for .flavor and .version (issue #14737) Reported by:
+ davidw Patches: flavor.patch uploaded by davidw (license 780)
+ Tested by: davidw ........
+
+2009-06-10 20:35 +0000 [r199996] David Brooks <dbrooks@digium.com>
+
+ * main/pbx.c, /: Fixes the argument order in definition of
+ new_find_extension(). In the definition of new_find_extension(),
+ the arguments 'callerid' and 'label' were swapped. The prototype
+ declaration and all calls to the function are ordered 'callerid'
+ then 'label', but the function itself was ordered 'label' then
+ 'callerid'. (closes issue #15303) Reported by: JimDickenson
+
+2009-06-10 20:18 +0000 [r199963] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 199958 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r199958 |
+ mmichelson | 2009-06-10 15:15:48 -0500 (Wed, 10 Jun 2009) | 6
+ lines Only try to use the invite_branch on outgoing INVITEs with
+ auth credentials. I have added a comment to the code to help ease
+ understanding of the logic here as well. ........
+
+2009-06-10 16:13 +0000 [r199859] Sean Bright <sean@malleable.com>
+
+ * include/asterisk/utils.h, /: Merged revisions 199857 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r199857 | seanbright | 2009-06-10 12:10:23 -0400
+ (Wed, 10 Jun 2009) | 9 lines Merged revisions 199856 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed,
+ 10 Jun 2009) | 2 lines __WORDSIZE is not available on all
+ platforms, so use sizeof(void *) instead. ........
+ ................
+
+2009-06-09 20:50 +0000 [r199745-199820] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 199818 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r199818 |
+ dvossel | 2009-06-09 15:47:57 -0500 (Tue, 09 Jun 2009) | 11 lines
+ CLI NOTIFY sending wrong transport type. SIP's cli NOTIFY command
+ only used UDP rather than copying the transport type from the
+ peer. (closes issue #15283) Reported by: jthurman Patches:
+ sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614)
+ Tested by: jthurman, dvossel ........
+
+ * main/loader.c, /, res/res_timing_pthread.c,
+ include/asterisk/module.h, res/res_timing_dahdi.c: Merged
+ revisions 199743 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r199743 |
+ dvossel | 2009-06-09 11:22:04 -0500 (Tue, 09 Jun 2009) | 11 lines
+ module load priority This patch adds the option to give a module
+ a load priority. The value represents the order in which a
+ module's load() function is initialized. The lower the value, the
+ higher the priority. The value is only checked if the
+ AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER
+ flag is not set, the value will never be read and the module will
+ be given the lowest possible priority on load. Since some modules
+ are reliant on a timing interface, the timing modules have been
+ given a high load priorty. (closes issue #15191) Reported by:
+ alecdavis Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/262/ ........
+
+2009-06-08 19:39 +0000 [r199633] Sean Bright <sean@malleable.com>
+
+ * include/asterisk/utils.h, /: Merged revisions 199630 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r199630 | seanbright | 2009-06-08 15:33:09 -0400
+ (Mon, 08 Jun 2009) | 32 lines Merged revisions 199626,199628 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun
+ 2009) | 21 lines Increase the size of our thread stack on 64 bit
+ processors. We were setting the stack size for each thread to
+ 240KB regardless of architecture, which meant that in some
+ scenarios we actually had less available stack space on 64 bit
+ processors (pointers use 8 bytes instead of 4). So now we
+ calculate the stack size we reserve based on the platform's
+ __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128
+ bit -> 1008KB (that's right, we're ready for 128 bit processors)
+ Patch typed by me but written by several members of
+ #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes
+ issue #14932) Reported by: jpiszcz Patches:
+ 06052009_issue14932.patch uploaded by seanbright (license 71)
+ Tested by: seanbright ........ r199628 | seanbright | 2009-06-08
+ 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the
+ stack size calculation just introduced. ........ ................
+
+2009-06-08 17:35 +0000 [r199590] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Recorded merge of revisions 199588 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r199588 | mmichelson | 2009-06-08 12:32:04 -0500 (Mon,
+ 08 Jun 2009) | 9 lines Fix a deadlock that could occur when
+ setting rtp stats on SIP calls. (closes issue #15143) Reported
+ by: cristiandimache Patches: 15143.patch uploaded by mmichelson
+ (license 60) Tested by: cristiandimache ........
+
+2009-06-05 21:32 +0000 [r199300] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/devicestate.h, /, main/devicestate.c: Merged
+ revisions 199298 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r199298 | dvossel | 2009-06-05 16:21:22 -0500 (Fri, 05 Jun 2009)
+ | 21 lines Merged revisions 199297 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009)
+ | 14 lines Fixes issue with hints giving unexpected results.
+ Hints with two or more devices that include ONHOLD gave
+ unexpected results. (closes issue #15057) Reported by:
+ p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel
+ (license 671) pbx.c.1.4.patch uploaded by p (license 558)
+ devicestate.c.trunk.patch uploaded by p (license 671) Tested by:
+ p_lindheimer, dvossel Review:
+ https://reviewboard.asterisk.org/r/254/ ........ ................
+
+2009-06-05 13:51 +0000 [r199229] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 199227 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r199227 | mmichelson | 2009-06-05 08:51:08 -0500 (Fri, 05 Jun
+ 2009) | 14 lines Correct "dahdi show channels" output when
+ specifying a group. Since a DAHDI channel may belong to multiple
+ groups, we need to use a bitwise and instead of equivalence to
+ determine whether to display the channel information. (closes
+ issue #15248) Reported by: gentian Patches: 15248.patch uploaded
+ by mmichelson (license 60) Tested by: gentian ........
+
+2009-06-04 19:16 +0000 [r199141] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 199139 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r199139 | dvossel | 2009-06-04 14:10:16 -0500
+ (Thu, 04 Jun 2009) | 9 lines Merged revisions 199138 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04
+ Jun 2009) | 3 lines Additional updates to AST-2009-001 ........
+ ................
+
+2009-06-04 14:53 +0000 [r199053] Sean Bright <sean@malleable.com>
+
+ * main/asterisk.c, main/loader.c, /, include/asterisk/_private.h:
+ Merged revisions 199051 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun
+ 2009) | 47 lines Merged revisions 199022 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun
+ 2009) | 40 lines Safely handle AMI connections/reload requests
+ that occur during startup. During asterisk startup, a lock on the
+ list of modules is obtained by the primary thread while each
+ module is initialized. Issue 13778 pointed out a problem with
+ this approach, however. Because the AMI is loaded before other
+ modules, it is possible for a module reload to be issued by a
+ connected client (via Action: Command), causing a deadlock. The
+ resolution for 13778 was to move initialization of the manager to
+ happen after the other modules had already been lodaded. While
+ this fixed this particular issue, it caused a problem for users
+ (like FreePBX) who call AMI scripts via an #exec in a
+ configuration file (See issue 15189). The solution I have come up
+ with is to defer any reload requests that come in until after the
+ server is fully booted. When a call comes in to ast_module_reload
+ (from wherever) before we are fully booted, the request is added
+ to a queue of pending requests. Once we are done booting up, we
+ then execute these deferred requests in turn. Note that I have
+ tried to make this a bit more intelligent in that it will not
+ queue up more than 1 request for the same module to be reloaded,
+ and if a general reload request comes in ('module reload') the
+ queue is flushed and we only issue a single deferred reload for
+ the entire system. As for how this will impact existing
+ installations - Before 13778, a reload issued before module
+ initialization was completed would result in a deadlock. After
+ 13778, you simply couldn't connect to the manager during startup
+ (which causes problems with #exec-that-calls-AMI configuration
+ files). I believe this is a good general purpose solution that
+ won't negatively impact existing installations. (closes issue
+ #15189) (closes issue #13778) Reported by: p_lindheimer Patches:
+ 06032009_15189_deferred_reloads.diff uploaded by seanbright
+ (license 71) Tested by: p_lindheimer, seanbright Review:
+ https://reviewboard.asterisk.org/r/272/ ........ ................
+
+2009-06-03 15:26 +0000 [r198826-198887] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /, main/features.c, include/asterisk/channel.h:
+ Merged revisions 198856 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r198856 |
+ dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines
+ Generic call forward api, ast_call_forward() The function
+ ast_call_forward() forwards a call to an extension specified in
+ an ast_channel's call_forward string. After an ast_channel is
+ called, if the channel's call_forward string is set this function
+ can be used to forward the call to a new channel and terminate
+ the original one. I have included this api call in both
+ channel.c's ast_request_and_dial() and feature.c's
+ feature_request_and_dial(). App_dial and app_queue already
+ contain call forward logic specific for their application and
+ options. (closes issue #13630) Reported by: festr Review:
+ https://reviewboard.asterisk.org/r/271/ ........
+
+ * channels/chan_iax2.c, /: Merged revisions 198824 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r198824 | dvossel | 2009-06-02 12:55:35 -0500 (Tue, 02 Jun 2009)
+ | 8 lines fixes issue with channels not going down after transfer
+ Iax2 currently does not support native bridging if the timeoutms
+ value is set. We check for that in iax2_bridge, but then set
+ timeoutms to 0 by default. If the timeoutms is not provided it is
+ set to -1. By setting timeoutms to 0 it is processed causing a
+ bridging retry loop. (closes issue #15216) Reported by: oxymoron
+ Tested by: dvossel ........
+
+2009-06-02 13:50 +0000 [r198793] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+ 198791 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r198791 |
+ file | 2009-06-02 10:48:06 -0300 (Tue, 02 Jun 2009) | 5 lines
+ Correct documentation for the register line, specifically where
+ the domain should be specified. (closes issue #14367) Reported
+ by: Nick_Lewis ........
+
+2009-06-01 18:44 +0000 [r198628] Tilghman Lesher <tlesher@digium.com>
+
+ * /, contrib/scripts/meetme.sql: Merged revisions 198626 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r198626 | tilghman | 2009-06-01 13:40:35 -0500 (Mon, 01
+ Jun 2009) | 2 lines Add information for new meetme realtime
+ fields ........
+
+2009-05-31 01:58 +0000 [r198441] Eliel C. Sardanons <eliels@gmail.com>
+
+ * /, res/res_timing_dahdi.c: Merged revisions 198437 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r198437 | eliel | 2009-05-30 21:22:15 -0400 (Sat, 30 May 2009) |
+ 11 lines Avoid a crash when res_timing_dahdi is unloaded but
+ wasn't properly loaded. if dahdi_test_timer() fails,
+ timing_funcs_handle remains NULL causing a crash when calling
+ ast_unregister_timing_interface() with a NULL pointer. (closes
+ issue #15234) Reported by: eliel Patches: timing_dahdi1.diff
+ uploaded by eliel (license 64) ........
+
+2009-05-30 20:21 +0000 [r198373-198390] Sean Bright <sean@malleable.com>
+
+ * res/res_jabber.c, /: Merged revisions 198375 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r198375 |
+ seanbright | 2009-05-30 16:11:33 -0400 (Sat, 30 May 2009) | 13
+ lines Properly terminate the receive buffer before sending to
+ iksemel. aji_io_recv takes the maximum number of bytes to read
+ (instead of the total buffer size), so we have to subtract 1 from
+ our buffer size. Without this, when we receive packets that are
+ larger than our buffer, iksemel will choke and things get wonky.
+ (closes issue #15232) Reported by: lp0 Patches:
+ 05302009_res_jabber.c.patch uploaded by seanbright (license 71)
+ Tested by: seanbright, lp0 ........
+
+ * res/res_jabber.c, /: Merged revisions 198371 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r198371 | seanbright | 2009-05-30 15:38:58 -0400 (Sat, 30 May
+ 2009) | 19 lines Merged revisions 198370 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May
+ 2009) | 12 lines Properly terminate AMI JabberSend response
+ messages. The response message (either Error or Success) needs an
+ extra trailing \r\n after the fields to inform the client that
+ the message is complete. (closes issue #14876) Reported by: srt
+ Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright
+ (license 71) asterisk_14876.patch uploaded by srt (license 378)
+ trunk-14876-2.diff uploaded by phsultan (license 73) ........
+ ................
+
+2009-05-30 03:49 +0000 [r198314] Russell Bryant <russell@digium.com>
+
+ * res/res_smdi.c, /: Merged revisions 198312 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r198312 | russell | 2009-05-29 22:43:23 -0500 (Fri, 29 May 2009)
+ | 12 lines Merged revisions 198311 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r198311 | russell | 2009-05-29 22:42:46 -0500 (Fri, 29 May 2009)
+ | 5 lines Fix a crash that occurred when MWI SMDI messages
+ expired. (closes issue #14561) Reported by: cmoss28 ........
+ ................
+
+2009-05-30 03:28 +0000 [r198295] Sean Bright <sean@malleable.com>
+
+ * apps/app_dial.c, /: Merged revisions 198285 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r198285 | seanbright | 2009-05-29 23:26:06 -0400 (Fri, 29 May
+ 2009) | 15 lines Merged revisions 198251 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May
+ 2009) | 8 lines Treat an empty FORWARD_CONTEXT the same way we
+ treat a missing one. (closes issue #15056) Reported by:
+ p_lindheimer Patches: 05292009_bug15056.diff uploaded by
+ seanbright (license 71) Tested by: p_lindheimer ........
+ ................
+
+2009-05-30 02:34 +0000 [r198249] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 198248 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r198248 |
+ file | 2009-05-29 23:31:48 -0300 (Fri, 29 May 2009) | 2 lines
+ When removing all packets from a dialog we also need to free the
+ data if present. ........
+
+2009-05-29 23:05 +0000 [r198147-198187] Russell Bryant <russell@digium.com>
+
+ * /, configs/modules.conf.sample: Merged revisions 198186 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r198186 | russell | 2009-05-29 18:04:31 -0500 (Fri, 29
+ May 2009) | 2 lines Suggesting that only a single timing module
+ be loaded is no longer necessary. ........
+
+ * /, res/res_timing_pthread.c: Merged revisions 198183 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r198183 | russell | 2009-05-29 17:33:31 -0500 (Fri, 29 May 2009)
+ | 2 lines Improve handling of trying to ACK too many timer
+ expirations. ........
+
+ * /, res/res_timing_pthread.c: Merged revisions 198146 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r198146 | russell | 2009-05-29 15:06:59 -0500 (Fri, 29 May 2009)
+ | 38 lines Resolve issues with choppy sound when using
+ res_timing_pthread. The situation that caused this problem was
+ when continuous mode was being turned on and off while a rate was
+ set for a timing interface. A very easy way to replicate this bug
+ was to do a Playback() from behind a Local channel. In this
+ scenario, a rate gets set on the channel for doing file playback.
+ At the same time, continuous mode gets turned on and off about
+ every 20 ms as frames get queued on to the PBX side channel from
+ the other side of the Local channel. Essentially, this module
+ treated continuous mode and a set rate as mutually exclusive
+ states for the timer to be in. When I dug deep enough, I observed
+ the following pattern: 1) Set timer to tick every 20 ms. 2) Wait
+ almost 20 ms ... 3) Continuous mode gets turned on for a queued
+ up frame 4) Continuous mode gets turned off 5) The timer goes
+ back to its tick per 20 ms. state but starts counting at 0 ms. 6)
+ Goto step 2. Sometimes, res_timing_pthread would make it 20 ms
+ and produce a timer tick, but not most of the time. This is what
+ produced the choppy sound (or sometimes no sound at all). Now,
+ the module treats continuous mode and a set rate as completely
+ independent timer modes. They can be enabled and disabled
+ independently of each other and things work as expected. (closes
+ issue #14412) Reported by: dome Patches: issue14412.diff.txt
+ uploaded by russell (license 2) issue14412-1.6.1.0.diff.txt
+ uploaded by russell (license 2) Tested by: DennisD, russell
+ ........
+
+2009-05-29 19:13 +0000 [r198074] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/cdr.c, main/channel.c, /, include/asterisk/cdr.h: Merged
+ revisions 198072 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r198072 | mnicholson | 2009-05-29 14:04:24 -0500 (Fri, 29 May
+ 2009) | 21 lines Merged revisions 198068 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May
+ 2009) | 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as
+ the default CDR disposition. This change also involves the
+ addition of an AST_CDR_FLAG_ORIGINATED flag that is used on
+ originated channels to distinguish: them from dialed channels.
+ (closes issue #12946) Reported by: meral Patches: null-cdr2.diff
+ uploaded by mnicholson (license 96) Tested by: mnicholson,
+ dbrooks (closes issue #15122) Reported by: sum Tested by: sum
+ ........ ................
+
+2009-05-29 18:39 +0000 [r198065] Joshua Colp <jcolp@digium.com>
+
+ * /, main/file.c: Merged revisions 198064 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r198064 |
+ file | 2009-05-29 15:39:04 -0300 (Fri, 29 May 2009) | 2 lines Fix
+ a memory leak of the write buffer when writing a file. ........
+
+2009-05-29 18:17 +0000 [r198005] Sean Bright <sean@malleable.com>
+
+ * Makefile, /: Merged revisions 198000 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r198000 | seanbright | 2009-05-29 14:15:15 -0400 (Fri, 29 May
+ 2009) | 15 lines Merged revisions 197998 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r197998 | seanbright | 2009-05-29 14:14:12 -0400 (Fri, 29 May
+ 2009) | 8 lines Fix 'make config' target for Slackware. There was
+ a missing semi-colon after the echo statement in the Makefile
+ that was causing problems for some users. Fix suggested by
+ reporter. (closes issue #15225) Reported by: pdavis ........
+ ................
+
+2009-05-29 16:19 +0000 [r197969] Russell Bryant <russell@digium.com>
+
+ * /, res/res_timing_pthread.c: Merged revisions 197960 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r197960 | russell | 2009-05-29 11:15:30 -0500 (Fri, 29 May 2009)
+ | 2 lines Trim trailing whitespace so that I can work on this bug
+ without it bothering me. :-) ........
+
+2009-05-28 23:59 +0000 [r197897] 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 18:47 +0000 [r197700] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 197697 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r197697 | file | 2009-05-28 15:45:11 -0300 (Thu, 28 May 2009) | 2
+ lines Fix a bug where the trunkmtu setting was not set to the
+ default value of 1240 on load but was on reload. ........
+
+2009-05-28 18:26 +0000 [r197696] Eliel C. Sardanons <eliels@gmail.com>
+
+ * /, channels/chan_sip.c: Merged revisions 197621 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r197621 | eliel | 2009-05-28 12:01:48 -0400 (Thu, 28 May 2009) |
+ 19 lines Merged revisions 197562 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) |
+ 13 lines Use the address we already know when reloading a peer
+ with nat=yes. If we already have an address for a peer, and we
+ are reloading the sip configuration, try to use that address to
+ contact the peer, instead of getting it from the Contact. (closes
+ issue #15194) Reported by: ibc Patches: sip.patch uploaded by
+ eliel (license 64) Tested by: manwe ........ ................
+
+2009-05-28 16:08 +0000 [r197623] 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 15:39 +0000 [r197545-197618] Mark Michelson <mmichelson@digium.com>
+
+ * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h:
+ Merged revisions 197606 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May
+ 2009) | 22 lines Recorded merge of revisions 197588 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu,
+ 28 May 2009) | 16 lines Allow for media to arrive from an
+ alternate source when responding to a reinvite with 491. When we
+ receive a SIP reinvite, it is possible that we may not be able to
+ process the reinvite immediately since we have also sent a
+ reinvite out ourselves. The problem is that whoever sent us the
+ reinvite may have also sent a reinvite out to another party, and
+ that reinvite may have succeeded. As a result, even though we are
+ not going to accept the reinvite we just received, it is
+ important for us to not have problems if we suddenly start
+ receiving RTP from a new source. The fix for this is to grab the
+ media source information from the SDP of the reinvite that we
+ receive. This information is passed to the RTP layer so that it
+ will know about the alternate source for media. Review:
+ https://reviewboard.asterisk.org/r/252 ........ ................
+
+ * apps/app_chanspy.c, /, include/asterisk/audiohook.h,
+ main/audiohook.c: Merged revisions 197543 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r197543 | mmichelson | 2009-05-28 09:58:06 -0500 (Thu, 28 May
+ 2009) | 27 lines Merged revisions 197537 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May
+ 2009) | 21 lines Add flags to chanspy audiohook so that audio
+ stays in sync. There are two flags being added to the chanspy
+ audiohook here. One is the pre-existing
+ AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that
+ the read and write slinfactories on the audiohook do not skew
+ beyond a certain tolerance. In addition, there is a new audiohook
+ flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set,
+ we do not allow for a slinfactory to build up a substantial
+ amount of audio before flushing it. For this particular issue,
+ this means that the person spying on the call will hear the
+ conversations in real time with very little delay in the audio.
+ (closes issue #13745) Reported by: geoffs Patches: 13745.patch
+ uploaded by mmichelson (license 60) Tested by: snblitz ........
+ ................
+
+2009-05-28 14:54 +0000 [r197470-197540] Joshua Colp <jcolp@digium.com>
+
+ * /, main/utils.c: Merged revisions 197538 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r197538 |
+ file | 2009-05-28 11:51:43 -0300 (Thu, 28 May 2009) | 5 lines Fix
+ a bug in stringfields where it did not actually free the pools of
+ memory. (closes issue #15074) Reported by: pj ........
+
+ * /, channels/chan_sip.c: Merged revisions 197467 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r197467 | file | 2009-05-28 10:47:45 -0300 (Thu, 28 May 2009) |
+ 15 lines Merged revisions 197466 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8
+ lines Fix a bug where the flag indicating the presence of rport
+ would get overwritten by the nat setting. The presence of rport
+ is now stored as a separate flag. Once the dialog is setup and
+ authenticated (or it passes through unauthenticated) the proper
+ nat flag is set. (closes issue #13823) Reported by: dimas
+ ........ ................
+
+2009-05-28 11:40 +0000 [r197440] Gavin Henry <ghenry@suretecsystems.com>
+
+ * contrib/scripts/asterisk.ldap-schema,
+ contrib/scripts/asterisk.ldif, doc/ldap.txt,
+ configs/res_ldap.conf.sample: issue #15155 and issue #15156 from
+ trunk
+
+2009-05-27 20:11 +0000 [r197262] Sean Bright <sean@malleable.com>
+
+ * Makefile, /: Merged revisions 197260 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r197260 |
+ seanbright | 2009-05-27 16:08:16 -0400 (Wed, 27 May 2009) | 6
+ lines Use bash explicitly when calling build_tools/mkpkgconfig
+ from the Makefile. Since we use bashisms in
+ build_tools/mkpkgconfig, we should call on bash explicitly when
+ running from the Makefile, otherwise we get errors during a 'make
+ install.' (closes issue #15209) Reported by: seandarcy ........
+
+2009-05-27 19:29 +0000 [r197245] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_cut.c: Recorded merge of revisions 197209 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r197209 | tilghman | 2009-05-27 14:20:56 -0500
+ (Wed, 27 May 2009) | 12 lines Recorded merge of revisions 197194
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r197194 | tilghman | 2009-05-27 14:09:42 -0500 (Wed, 27 May 2009)
+ | 5 lines Use a different determinator on whether to print the
+ delimiter, since leading fields may be blank. (closes issue
+ #15208) Reported by: ramonpeek Patch by me, though inspired in
+ part by a patch from ramonpeek ........ ................
+
+2009-05-27 17:21 +0000 [r197145] 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 16:12 +0000 [r197091] Sean Bright <sean@malleable.com>
+
+ * configs/smdi.conf.sample, configs/extensions.conf.sample,
+ configs/sla.conf.sample, configs/chan_dahdi.conf.sample, /,
+ configs/vpb.conf.sample: Merged revisions 197089 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r197089 | seanbright | 2009-05-27 12:07:57 -0400 (Wed, 27 May
+ 2009) | 6 lines Fix references to /etc/dahdi/system.conf and
+ /etc/asterisk/chan_dahdi.conf in the sample configuration files.
+ (closes issue #15207) Reported by: seandarcy ........
+
+2009-05-27 15:59 +0000 [r197087] David Vossel <dvossel@digium.com>
+
+ * channels/chan_sip.c: Fixes merge issue for r196453.
+
+2009-05-27 13:05 +0000 [r196990] Sean Bright <sean@malleable.com>
+
+ * /, channels/chan_alsa.c: Merged revisions 196988 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r196988 | seanbright | 2009-05-27 09:02:54 -0400 (Wed, 27 May
+ 2009) | 9 lines Display an error message when chan_alsa fails to
+ load due to a missing or inaccessible configuration file. Before
+ this change, when chan_alsa failed to load due to a missing or
+ inaccessible configuration file, no message would be displayed.
+ With this change, when chan_alsa fails to load due to a missing
+ or inaccessible configuration file, a message will be displayed.
+ (closes issue #14760) Reported by: Nick_Lewis Patches:
+ chan_alsa.c-confload.patch uploaded by Nick (license 657)
+ ........
+
+2009-05-26 22:42 +0000 [r196869-196947] Russell Bryant <russell@digium.com>
+
+ * /, autoconf/ast_check_osptk.m4 (added), configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 196946 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r196946 |
+ russell | 2009-05-26 17:40:34 -0500 (Tue, 26 May 2009) | 8 lines
+ Update configure script to check for OSP toolkit 3.5.0. (closes
+ issue #14988) Reported by: tzafrir Patches: configure.ac.diff
+ uploaded by homesick (license 91) new_ast_check_osptk.m4 uploaded
+ by homesick (license 91) ........
+
+ * /, res/res_convert.c: Merged revisions 196843 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r196843 | russell | 2009-05-26 13:20:57 -0500 (Tue, 26 May 2009)
+ | 16 lines Merged revisions 196826 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r196826 | russell | 2009-05-26 13:14:36 -0500 (Tue, 26 May 2009)
+ | 9 lines Resolve a file handle leak. The frames here should have
+ always been freed. However, out of luck, there was never any
+ memory leaked. However, after file streams became reference
+ counted, this code would leak the file stream for the file being
+ read. (closes issue #15181) Reported by: jkroon ........
+ ................
+
+2009-05-26 13:46 +0000 [r196660-196723] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 196721 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r196721 |
+ file | 2009-05-26 10:43:13 -0300 (Tue, 26 May 2009) | 7 lines Fix
+ a bug where the sip unregister CLI command did not completely
+ unregister the peer. (closes issue #15118) Reported by: alecdavis
+ Patches: chan_sip_unregister.diff2.txt uploaded by alecdavis
+ (license 585) ........
+
+ * contrib/scripts/safe_asterisk, /: Merged revisions 196658 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r196658 | file | 2009-05-26 10:06:50 -0300 (Tue,
+ 26 May 2009) | 14 lines Merged revisions 196657 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r196657 | file | 2009-05-26 10:06:09 -0300 (Tue, 26 May 2009) | 7
+ lines Remove some bash specific stuff from safe_asterisk. (closes
+ issue #10812) Reported by: paravoid Patches:
+ safe_asterisk_bashism.diff uploaded by tzafrir (license 46)
+ ........ ................
+
+2009-05-22 22:35 +0000 [r196453] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+ 196416 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r196416 |
+ dvossel | 2009-05-22 16:09:45 -0500 (Fri, 22 May 2009) | 19 lines
+ SIP set outbound transport type from Registration In sip.conf the
+ transport option allows for the configuration of what transport
+ types (udp, tcp, and tls) a peer will accept, but only the first
+ type listed was used for outbound connections. This patch changes
+ this. Now the default transport type is only used until the peer
+ registers. When registration takes place the transport type is
+ parsed out of the Contact header. If the Contact header's
+ transport type is equal to one that the peer supports, the peer's
+ default transport type for outbound connections is set to match
+ the Contact header's type. If the Contact header's transport type
+ is not present, then the peer's default transport type is set to
+ match the one the peer registered with. When a peer unregisters
+ or the registration expires, the default transport type for that
+ peer is reset. (closes issue #12282) Reported by: rjain Patches:
+ reg_patch_1.diff uploaded by dvossel (license 671) Tested by:
+ dvossel (closes issue #14727) Reported by: pj Patches:
+ reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj,
+ dvossel Review: https://reviewboard.asterisk.org/r/249/ ........
+
+2009-05-22 13:58 +0000 [r196119] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 196117 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r196117 | file | 2009-05-22 10:56:47 -0300 (Fri,
+ 22 May 2009) | 12 lines Merged revisions 196116 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r196116 | file | 2009-05-22 10:54:17 -0300 (Fri, 22 May 2009) | 5
+ lines Fix a bug where using immediate with mISDN caused a cause
+ code of 16 to get sent back instead of 1 if the 's' extension did
+ not exist. (closes issue #12286) Reported by: lmamane ........
+ ................
+
+2009-05-21 19:13 +0000 [r195998] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 195995 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r195995 | dvossel | 2009-05-21 14:11:49 -0500
+ (Thu, 21 May 2009) | 20 lines Merged revisions 195991 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21 May 2009)
+ | 14 lines Sign problem calculating timestamp for iax frame leads
+ to no audio on the receiving peer. There are rare cases in which
+ a frame's delivery timestamp is slightly less than the iax2_pvt's
+ offset. This causes the pvt's timestamp to be a small negative
+ number, but since the timestamp value is unsigned it looks like a
+ huge positive number. This patch checks for this negative case
+ and sets the ms to zero. A similar check is already done right
+ below this one in the 'else' statement. (closes issue #15032)
+ Reported by: guillecabeza Patches: chan_iax2.c.patch_timestamp
+ uploaded by guillecabeza (license 380) Tested by: guillecabeza
+ (closes issue #14216) Reported by: Andrey Sofronov ........
+ ................
+
+2009-05-21 16:19 +0000 [r195892] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/cdr.c, /, include/asterisk/cdr.h: Merged revisions 195882
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r195882 | mnicholson | 2009-05-21 10:33:55 -0500
+ (Thu, 21 May 2009) | 20 lines Merged revisions 195881 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May
+ 2009) | 13 lines This commit prevents cdr records with
+ AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated
+ in certain cases. This is accomplished by adding two functions to
+ update the answer time and disposition of calls that checks for
+ the proper lock flags. These functions are used in the
+ ast_bridge_call() function so that ForkCDR(A) calls are
+ respected. This patch also modifies the way ast_bridge_call()
+ chooses the cdr record to base the bridged_cdr on. Previously the
+ first unlocked cdr record would be chosen, now instead the first
+ cdr record is chosen and forked cdr records are moved to the
+ bridge_cdr. This allows the original cdr record and any forked
+ cdr records to be properly updated with answer and end times.
+ (closes issue #13797) Reported by: sh0t Tested by: sh0t (closes
+ issue #14744) Reported by: deepesh ........ ................
+
+2009-05-20 23:31 +0000 [r195841] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_stack.c, /: Merged revisions 195839 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r195839 |
+ tilghman | 2009-05-20 18:30:05 -0500 (Wed, 20 May 2009) | 3 lines
+ If a variable had a blank value upon the initial setting, then it
+ would do nothing. Identified by Dmitry Andrianov via private
+ email, fixed by me. ........
+
+2009-05-20 17:34 +0000 [r195638-195705] Joshua Colp <jcolp@digium.com>
+
+ * /, main/features.c: Merged revisions 195698 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r195698 | file | 2009-05-20 14:33:02 -0300 (Wed, 20 May 2009) |
+ 12 lines Merged revisions 195688 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195688 | file | 2009-05-20 14:30:25 -0300 (Wed, 20 May 2009) | 5
+ lines Fix some code that wrongly assumed a pointer would always
+ be non-NULL when dealing with CDRs after a bridge. (closes issue
+ #15079) Reported by: barryf ........ ................
+
+ * /, apps/app_meetme.c: Merged revisions 195636 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r195636 | file | 2009-05-20 14:14:42 -0300 (Wed, 20 May 2009) |
+ 12 lines Merged revisions 195635 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195635 | file | 2009-05-20 14:14:00 -0300 (Wed, 20 May 2009) | 5
+ lines Fix a bug where the MeetMe option 'D' did not actually
+ prompt for the pin. (closes issue #15050) Reported by: pmhaddad
+ ........ ................
+
+2009-05-19 20:18 +0000 [r195526] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 195521 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r195521 | tilghman | 2009-05-19 15:16:01 -0500
+ (Tue, 19 May 2009) | 14 lines Merged revisions 195520 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195520 | tilghman | 2009-05-19 15:12:20 -0500 (Tue, 19 May 2009)
+ | 7 lines Ensure thread keys are initialized before attempting to
+ access them. (closes issue #14889) Reported by: jaroth Patches:
+ app_voicemail.c.patch uploaded by msirota (license 758) Tested
+ by: msirota, BlargMaN ........ ................
+
+2009-05-19 14:47 +0000 [r195451] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 195449 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r195449 | file | 2009-05-19 11:43:54 -0300 (Tue, 19 May 2009) |
+ 14 lines Merged revisions 195448 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195448 | file | 2009-05-19 11:41:45 -0300 (Tue, 19 May 2009) | 7
+ lines Fix a bug where direct RTP setup would partially occur even
+ when disabled if the calling channel was answered. (issue #13545)
+ Reported by: davidw (issue #14244) Reported by: mbnwa ........
+ ................
+
+2009-05-18 21:31 +0000 [r195429] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/manager.c, /: Merged revisions 195369 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r195369 |
+ eliel | 2009-05-18 16:49:20 -0400 (Mon, 18 May 2009) | 8 lines
+ Fix the CLI command 'manager show command' documentation and
+ functionality. The CLI command 'manager show command' supports
+ passing multiple action names in the same line, but it was not
+ allowing that because of a incorrect check in the argumentes
+ counter. Also the documentation was updated to show that this
+ usage of the command is possible. ........
+
+2009-05-18 20:54 +0000 [r195358-195372] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c, include/asterisk/smdi.h, apps/app_voicemail.c,
+ res/res_smdi.c, /, include/asterisk/monitor.h: Recorded merge of
+ revisions 195370 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r195370 | tilghman | 2009-05-18 15:52:33 -0500 (Mon, 18 May 2009)
+ | 15 lines Recorded merge of revisions 195366 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195366 | tilghman | 2009-05-18 15:24:13 -0500 (Mon, 18 May 2009)
+ | 8 lines Add a similar dependency on SMDI for voicemail as
+ already exists for ADSI. (closes issue #14846) Reported by: pj
+ Patches: 20090413__bug14846__1.4.diff.txt uploaded by tilghman
+ (license 14) 20090507__issue14846__1.6.0.diff.txt uploaded by
+ tilghman (license 14) 20090507__issue14846__1.6.1.diff.txt
+ uploaded by tilghman (license 14) ........ ................
+
+ * main/asterisk.c, /: Merged revisions 195320 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r195320 |
+ tilghman | 2009-05-18 14:17:15 -0500 (Mon, 18 May 2009) | 9 lines
+ Move the spawn of astcanary down, until after the call to
+ daemon(3). This avoids possible conflicts with the internal
+ implementation of daemon(3). (closes issue #15093) Reported by:
+ tzafrir Patches: 20090513__issue15093__2.diff.txt uploaded by
+ tilghman (license 14) Tested by: tzafrir ........
+
+2009-05-18 19:00 +0000 [r195318] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_externalivr.c: Merged revisions 195316 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r195316 | mmichelson | 2009-05-18 13:58:26 -0500 (Mon, 18 May
+ 2009) | 18 lines Fix externalivr's setvariable command so that it
+ properly sets multiple variables. The command had a for loop that
+ was guaranteed to only execute once since the continuation
+ operation of the loop would set the input buffer NULL. I rewrote
+ the loop so that its operation was more obvious, and it would set
+ multiple variables correctly. I also reduced stack space required
+ for the function, constified the input string, and modified the
+ function so that it would not modify the input string while I was
+ at it. (closes issue #15114) Reported by: chris-mac Patches:
+ 15114.patch uploaded by mmichelson (license 60) Tested by:
+ chris-mac ........
+
+2009-05-18 15:55 +0000 [r195209] Joshua Colp <jcolp@digium.com>
+
+ * main/frame.c, /: Merged revisions 195207 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r195207 | file | 2009-05-18 12:53:26 -0300 (Mon, 18 May 2009) |
+ 14 lines Merged revisions 195206 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195206 | file | 2009-05-18 12:51:22 -0300 (Mon, 18 May 2009) | 7
+ lines Fix a typo which caused loss of audio when using G729 in
+ some scenarios with a smoother present. (closes issue #15105)
+ Reported by: bamby Patches: process-vad-correctly.diff uploaded
+ by bamby (license 430) ........ ................
+
+2009-05-18 15:13 +0000 [r195167] Eliel C. Sardanons <eliels@gmail.com>
+
+ * apps/app_dial.c, main/pbx.c, /, apps/app_macro.c: Merged
+ revisions 195162 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r195162 |
+ eliel | 2009-05-18 10:45:23 -0400 (Mon, 18 May 2009) | 9 lines
+ Warn about the use of the application WaitExten() within a
+ Macro(). Update applications documentation to warn the user about
+ the use of the WaitExten() application within a Macro().
+ Recommend the use of Read() instead. (closes issue #14444)
+ Reported by: ewieling ........
+
+2009-05-18 13:58 +0000 [r195091-195098] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c, /: Merged revisions 195096 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r195096 | file | 2009-05-18 10:56:16 -0300 (Mon, 18 May 2009) |
+ 12 lines Merged revisions 195095 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195095 | file | 2009-05-18 10:53:39 -0300 (Mon, 18 May 2009) | 5
+ lines Fix a bug where the codecs of the called party leg were not
+ properly sent back to the caller call leg when reinvited. (closes
+ issue #13569) Reported by: bkw918 ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 195089 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r195089 |
+ file | 2009-05-18 10:36:17 -0300 (Mon, 18 May 2009) | 5 lines Fix
+ a bug where specifying an empty outboundproxy would cause packets
+ to get sent to ourself. (closes issue #15106) Reported by:
+ timeshell ........
+
+2009-05-18 13:07 +0000 [r195023] Russell Bryant <russell@digium.com>
+
+ * main/manager.c, /: Merged revisions 195021 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r195021 | russell | 2009-05-18 07:59:11 -0500 (Mon, 18 May 2009)
+ | 12 lines Recorded merge of revisions 195020 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r195020 | russell | 2009-05-18 07:57:46 -0500 (Mon, 18 May 2009)
+ | 5 lines Don't try to unlock a bogus channel. (closes issue
+ #15144) Reported by: cristiandimache ........ ................
+
+2009-05-15 22:46 +0000 [r194835-194876] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 194874 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r194874 | dvossel | 2009-05-15 17:44:44 -0500
+ (Fri, 15 May 2009) | 23 lines Merged revisions 194873 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r194873 | dvossel | 2009-05-15 17:43:13 -0500 (Fri, 15 May 2009)
+ | 17 lines IAX2 REGAUTH loop IAX was not sending REGREJ to
+ terminate invalid registrations. Instead it sent another REGAUTH
+ if the authentication challenge failed. This caused a loop of
+ REGREQ and REGAUTH frames. (Related to Security fix AST-2009-001)
+ (closes issue #14867) Reported by: aragon Tested by: dvossel
+ (closes issue #14717) Reported by: mobeck Patches:
+ regauth_loop_update_patch.diff uploaded by dvossel (license 671)
+ Tested by: dvossel ........ ................
+
+ * channels/chan_iax2.c, channels/iax2-parser.c,
+ channels/iax2-parser.h, /, channels/iax2.h: Merged revisions
+ 194833 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r194833 | dvossel | 2009-05-15 15:52:12 -0500 (Fri, 15 May 2009)
+ | 24 lines Merged revisions 194557,194685 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r194557 | dvossel | 2009-05-14 17:59:43 -0500 (Thu, 14 May 2009)
+ | 10 lines IAX2 "Ghost" Channels There is a bug tracker issue
+ where people are reporting "Ghost" channels in their 'iax2 show
+ channels' output. The confusion is caused by channels being
+ listed as "(NONE)" with format "unknown". These are not channels
+ of coarse. They are usually just pending registration or poke
+ requests, but it is confusing output. To help make sense of this
+ I have added two columns to 'iax2 show channels'. One shows the
+ first message which started the transaction, and the second shows
+ the last message sent by either side of the call. This helps
+ diagnose why the entry exists and why it may not go away. (closes
+ issue #14207) Reported by: clive18 Review:
+ https://reviewboard.asterisk.org/r/246/ ........ r194685 |
+ dvossel | 2009-05-15 10:40:37 -0500 (Fri, 15 May 2009) | 6 lines
+ Update to previous IAX2 "Ghost" Channels patch. Fixed some
+ comments made on reviewboard for the previous patch. (issue
+ #14207) ........ ................
+
+2009-05-15 18:44 +0000 [r194716-194767] Russell Bryant <russell@digium.com>
+
+ * configs/logger.conf.sample, /: Merged revisions 194765 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r194765 | russell | 2009-05-15 13:43:42 -0500
+ (Fri, 15 May 2009) | 10 lines Merged revisions 194764 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r194764 | russell | 2009-05-15 13:43:18 -0500 (Fri, 15 May 2009)
+ | 2 lines Fix some spelling fail. ........ ................
+
+ * /, codecs/g722/g722_encode.c, codecs/g722/g722_decode.c: Merged
+ revisions 194722 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r194722 |
+ russell | 2009-05-15 12:59:08 -0500 (Fri, 15 May 2009) | 4 lines
+ Shuttle some bits around to address some gain issues with G.722.
+ (closes AST-209) ........
+
+ * codecs/Makefile, codecs/g722/Makefile (removed), /: Merged
+ revisions 194718 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r194718 |
+ russell | 2009-05-15 12:37:12 -0500 (Fri, 15 May 2009) | 2 lines
+ Further simplify codec_g722 build. ........
+
+ * codecs/Makefile, /: Merged revisions 194714 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r194714 |
+ russell | 2009-05-15 12:24:39 -0500 (Fri, 15 May 2009) | 2 lines
+ Actually force running make for g722. ........
+
+2009-05-14 22:30 +0000 [r194542] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /: Merged revisions 194520 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r194520 | kpfleming | 2009-05-14 17:26:02 -0500 (Thu, 14 May
+ 2009) | 9 lines Merged revisions 194509 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r194509 | kpfleming | 2009-05-14 17:23:49 -0500 (Thu, 14 May
+ 2009) | 1 line Update URL to Reviewboard ........
+ ................
+
+2009-05-14 22:23 +0000 [r194507] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 194496 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r194496 | mmichelson | 2009-05-14 17:20:51 -0500 (Thu, 14 May
+ 2009) | 30 lines Merged revisions 194484 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r194484 | mmichelson | 2009-05-14 17:17:55 -0500 (Thu, 14 May
+ 2009) | 24 lines Fix a race condition where a reinvite could
+ trigger a 482 response. The loop detection/spiral detection code
+ in chan_sip used the owner channel's state as a criterion for
+ determining if the incoming INVITE is a looped request. The
+ problem with this is that the INVITE-handling code happens in a
+ different thread than the thread that marks the owner channel as
+ being up. As a result, if a reinvite were to come in very
+ quickly, say from another Asterisk on the same LAN, it was
+ possible for the reinvite to arrive before the owner channel had
+ been set to the up state. This patch corrects the problem by
+ using the invitestate of the sip_pvt instead, since that can be
+ guaranteed to be set correctly by the time the reinvite arrives.
+ Since there is a switch statement further in the INVITE-handling
+ code, the AST_STATE_RINGING state also checks the invitestate of
+ the sip_pvt in case we should actually be treating the channel as
+ if it were up already. (closes issue #12215) Reported by: jpyle
+ Patches: 12215_confirmed.patch uploaded by mmichelson (license
+ 60) Tested by: lmadsen ........ ................
+
+2009-05-14 17:07 +0000 [r194436] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 194434 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r194434 |
+ file | 2009-05-14 14:05:33 -0300 (Thu, 14 May 2009) | 7 lines Fix
+ a bug where the 'T' option to Meetme did not work. (closes issue
+ #15031) Reported by: Stochastic (closes issue #13801) Reported
+ by: justdave ........
+
+2009-05-13 13:41 +0000 [r194212] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c, /: Merged revisions 194209 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r194209 | file | 2009-05-13 10:39:10 -0300 (Wed, 13 May 2009) |
+ 18 lines Merged revisions 194208 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r194208 | file | 2009-05-13 10:38:01 -0300 (Wed, 13 May 2009) |
+ 11 lines Fix RFC2833 issues with DTMF getting duplicated and with
+ duration wrapping over. (closes issue #14815) Reported by:
+ geoff2010 Patches: v1-14815.patch uploaded by dimas (license 88)
+ Tested by: geoff2010, file, dimas, ZX81, moliveras (closes issue
+ #14460) Reported by: moliveras Tested by: moliveras ........
+ ................
+
+2009-05-13 00:54 +0000 [r194140] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 194138 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r194138 | tilghman | 2009-05-12 19:52:49 -0500 (Tue, 12 May 2009)
+ | 14 lines Merged revisions 194137 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r194137 | tilghman | 2009-05-12 19:52:03 -0500 (Tue, 12 May 2009)
+ | 7 lines Fix logic for how to proceed with a single digit
+ extension. (closes issue #15091) Reported by: andrew Patches:
+ 20090512__issue15091.diff.txt uploaded by tilghman (license 14)
+ Tested by: andrew ........ ................
+
+2009-05-12 23:01 +0000 [r194062] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 194057 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r194057 | mnicholson | 2009-05-12 17:32:13 -0500 (Tue, 12 May
+ 2009) | 22 lines Merged revisions 194028 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r194028 | mnicholson | 2009-05-12 17:15:45 -0500 (Tue, 12 May
+ 2009) | 16 lines This change modifies app_queue to properly
+ generate CDR records in failure situations. This involves setting
+ a proper cdr disposition coresponding to the given failure
+ condition and ensuring the proper information is stored in the
+ cdr record. (closes issue #13691) Reported by: dferrer Tested by:
+ mnicholson (closes issue #13637) Reported by: atis Tested by:
+ atis ........ ................
+
+2009-05-12 20:51 +0000 [r193961] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 193954 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r193954 |
+ mmichelson | 2009-05-12 15:28:13 -0500 (Tue, 12 May 2009) | 18
+ lines Update spiral support in trunk and 1.6.X to match what is
+ in 1.4. In 1.4, a SIP spiral is treated the same way as a call
+ forward. This works much better than what is currently in trunk
+ and 1.6.X. The code in trunk and 1.6.X did not create a new call
+ to the recipient of the spiral, instead trying to continue the
+ same call. In addition to just being plain wrong, this also had
+ the side effect of only being able to spiral calls to other SIP
+ channels. With this in place, as long as call forwards are
+ honored, SIP spirals will work properly. This means that it will
+ work for outbound calls made by the Queue, Dial, and Page
+ applications. For originated calls and spool calls, however, the
+ spiral will not work properly until a generic call forward
+ mechanism is introduced into Asterisk. (relates to issue #13630)
+ ........
+
+2009-05-12 20:42 +0000 [r193822-193958] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 193956 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r193956 | tilghman | 2009-05-12 15:40:22 -0500
+ (Tue, 12 May 2009) | 13 lines Merged revisions 193955 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193955 | tilghman | 2009-05-12 15:39:21 -0500 (Tue, 12 May 2009)
+ | 6 lines Avoid initializing routines if the authentication
+ fails. Fixes a crash (RR) issue. (closes issue #14508) Reported
+ by: tiziano Patches: 20090221_2_wrongmailbox.diff.txt uploaded by
+ tiziano (license 377) ........ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 193870 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r193870 | tilghman | 2009-05-12 12:29:33 -0500 (Tue, 12 May 2009)
+ | 2 lines Convert a THREADSTORAGE object into a simple malloc'd
+ object (as suggested by Russell on -dev) ........
+
+ * apps/app_voicemail.c, /: Recorded merge of revisions 193756 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r193756 | tilghman | 2009-05-11 17:50:47 -0500
+ (Mon, 11 May 2009) | 25 lines Recorded merge of revisions 193755
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193755 | tilghman | 2009-05-11 17:48:20 -0500 (Mon, 11 May 2009)
+ | 18 lines Move 300 bytes around on the stack, to make more room
+ for an extension buffer. This allows more concurrent extensions
+ to be copied for a single voicemail, without creating a
+ possibility of upsetting existing users, where a dialplan could
+ run out of stack space where it had run fine before.
+ Alternatively, we could have allocated off the heap, but that is
+ a larger change and would have increased the chance for
+ instability introduced by this change. This is really solved
+ starting in 1.6.0.11, as the use of an ast_str buffer allows an
+ unlimited number of extensions (up to available memory). We
+ additionally create a new warning message when the buffer length
+ is exceeded, permitting administrators to see an issue after the
+ fact, whereas previously the list was silently truncated. (closes
+ issue #14739) Reported by: p_lindheimer Patches:
+ 20090417__bug14739.diff.txt uploaded by tilghman (license 14)
+ Tested by: p_lindheimer ........ ................
+
+2009-05-11 19:16 +0000 [r193616] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 193614 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r193614 | rmudgett | 2009-05-11 14:11:29 -0500
+ (Mon, 11 May 2009) | 19 lines Merged revisions 193613 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11 May 2009)
+ | 12 lines Sent wrong message to clear a call we started if the
+ other end has not responed yet. In the state MISDN_CALLING (i.e.
+ SETUP was sent but no answer has arrived yet), it is not allowed
+ to clear the call with RELEASE_COMPLETE. It must be cleared with
+ DISCONNECT. A RELEASE_COMPLETE is only allowed as an answer to a
+ SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a, 5.3.2.b) Patches:
+ chan-misdn-ccstate7.patch uploaded by customer. JIRA ABE-1862
+ ........ ................
+
+2009-05-11 18:07 +0000 [r193547] Leif Madsen <lmadsen@digium.com>
+
+ * /, funcs/func_channel.c: Recorded merge of revisions 193545 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r193545 | lmadsen | 2009-05-11 14:01:44 -0400
+ (Mon, 11 May 2009) | 14 lines Recorded merge of revisions 193544
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193544 | lmadsen | 2009-05-11 13:35:17 -0400 (Mon, 11 May 2009)
+ | 7 lines Document CHANNEL(transfercapability) in CLI
+ documentation. (issue #15073) Reported by: pkempgen Patches:
+ 20090511__issue15073.diff.txt uploaded by tilghman (license 14)
+ ........ ................
+
+2009-05-08 20:51 +0000 [r193389] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 193387 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r193387 |
+ dvossel | 2009-05-08 15:32:51 -0500 (Fri, 08 May 2009) | 7 lines
+ TCP not matching valid peer. find_peer() does not find a valid
+ peer when using pvt->recv as the sockaddr_in argument. Because of
+ the way TCP works, the port number in pvt->recv is not what we're
+ looking for at all. There is currently only one place that
+ find_peer searches for a peer using the sockaddr_in argument. If
+ the peer is not found after using pvt->recv (works for UDP since
+ the port number will be correct), a temp sockaddr_in struct is
+ made using the Contact header in the sip_request. This has the
+ correct port number in it. Review:
+ http://reviewboard.digium.com/r/236/ ........
+
+2009-05-08 15:36 +0000 [r193335] Sean Bright <sean@malleable.com>
+
+ * funcs/func_devstate.c, /: Merged revisions 193274 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r193274 | seanbright | 2009-05-08 11:18:40 -0400 (Fri, 08 May
+ 2009) | 2 lines Fix the spelling of UNAVAILABLE in func_devstate
+ CLI completion. ........
+
+2009-05-08 14:54 +0000 [r193265] David Vossel <dvossel@digium.com>
+
+ * channels/misdn_config.c, /: Merged revisions 193263 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r193263 | dvossel | 2009-05-08 09:52:19 -0500
+ (Fri, 08 May 2009) | 15 lines Merged revisions 193262 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193262 | dvossel | 2009-05-08 09:51:09 -0500 (Fri, 08 May 2009)
+ | 9 lines "misdn show config" segfaults asterisk, if no MSN lists
+ (closes issue #14976) Reported by: alecdavis Patches:
+ misdn_config.diff.txt uploaded by alecdavis (license 585) Tested
+ by: alecdavis, FabienToune ........ ................
+
+2009-05-08 14:10 +0000 [r193196] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configs/logger.conf.sample, /, main/logger.c: Merged revisions
+ 193194 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r193194 | kpfleming | 2009-05-08 09:06:15 -0500 (Fri, 08 May
+ 2009) | 13 lines Merged revisions 193193 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193193 | kpfleming | 2009-05-08 09:03:28 -0500 (Fri, 08 May
+ 2009) | 7 lines Make absolute paths for logger channels work
+ properly (Note: This is not a new feature, it was previously
+ undocumented and broken.) The Asterisk logger has a feature to
+ support absolute pathnames for logger channels, but the code
+ implementing the feature was broken. This has been fixed, and the
+ absolute path feature is now documented in the sample
+ logger.conf. ........ ................
+
+2009-05-07 23:44 +0000 [r193122] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 193120 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r193120 | tilghman | 2009-05-07 18:42:28 -0500 (Thu, 07 May 2009)
+ | 26 lines Merged revisions 193119 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193119 | tilghman | 2009-05-07 18:41:11 -0500 (Thu, 07 May 2009)
+ | 19 lines Fix Background within a Macro for FreePBX. If the
+ single digit DTMF is an extension in the specified context, then
+ go there and signal no DTMF. Otherwise, we should exit with that
+ DTMF. If we're in Macro, we'll exit and seek that DTMF as the
+ beginning of an extension in the Macro's calling context. If
+ we're not in Macro, then we'll simply seek that extension in the
+ calling context. Previously, someone complained about the
+ behavior as it related to the interior of a Gosub routine, and
+ the fix (#14011) inadvertently broke FreePBX (#14940). This
+ change should fix both of these situations, but with the possible
+ incompatibility that if a single digit extension does not exist
+ (but a longer extension COULD have matched), it would have
+ previously gone immediately to the "i" extension, but will now
+ need to wait for a timeout. (closes issue #14940) Reported by:
+ p_lindheimer Patches: 20090420__bug14940.diff.txt uploaded by
+ tilghman (license 14) Tested by: p_lindheimer ........
+ ................
+
+2009-05-07 22:42 +0000 [r193079] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 193077 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r193077 | rmudgett | 2009-05-07 17:24:04 -0500
+ (Thu, 07 May 2009) | 12 lines Merged revisions 193050 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193050 | rmudgett | 2009-05-07 17:17:06 -0500 (Thu, 07 May 2009)
+ | 5 lines Give a more helpful message when an incoming call's
+ dialed extension does not match. Added the dialed extension and
+ context to the chan_misdn messages warning that the dialed number
+ cannot be matched in the dialplan. ........ ................
+
+2009-05-07 17:52 +0000 [r192935-193007] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_odbc.c: Merged revisions 193006 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r193006 |
+ tilghman | 2009-05-07 12:51:13 -0500 (Thu, 07 May 2009) | 7 lines
+ Second result should not contain data from the first result.
+ (closes issue #15039) Reported by: jims Patches:
+ 20090506__issue15039.diff.txt uploaded by tilghman (license 14)
+ Tested by: jims ........
+
+ * channels/chan_unistim.c, /: Merged revisions 192938 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r192938 | tilghman | 2009-05-07 12:13:36 -0500 (Thu, 07 May 2009)
+ | 6 lines Send DTMF frame before playing back audio. (closes
+ issue #14858) Reported by: barryf Patches:
+ 20090507__bug14858.diff.txt uploaded by tilghman (license 14)
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 192933 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r192933 | tilghman | 2009-05-07 11:43:56 -0500 (Thu, 07 May 2009)
+ | 17 lines Merged revisions 192932 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r192932 | tilghman | 2009-05-07 11:29:08 -0500 (Thu, 07 May 2009)
+ | 10 lines Eliminate repetition of fullcontact during
+ reconstruction. If the fullcontact field appears in both the
+ sippeers and the sipregs table, then during reconstruction of the
+ field, it will otherwise be doubled. (closes issue #14754)
+ Reported by: Alexei Gradinari Patches:
+ 20090506__bug14754.diff.txt uploaded by tilghman (license 14)
+ Tested by: lmadsen ........ ................
+
+2009-05-06 22:19 +0000 [r192869] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 192861 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r192861 | jpeeler | 2009-05-06 17:17:27 -0500 (Wed, 06 May 2009)
+ | 17 lines Merged revisions 192858 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r192858 | jpeeler | 2009-05-06 17:15:19 -0500 (Wed, 06 May 2009)
+ | 10 lines Make ParkedCall application stop execution of the
+ dialplan after hang up Just changed park_exec to always return
+ non-zero. I really wasn't entirely sure at first if this was a
+ bug. Decided it was since it would be surprising when not using
+ ParkedCall in the dialplan to hang up and have dialplan execution
+ continue. (closes issue #14555) Reported by: francesco_r ........
+ ................
+
+2009-05-06 17:53 +0000 [r192812] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 190946 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r190946 | mattf | 2009-04-28 17:05:05 -0500 (Tue, 28 Apr 2009) |
+ 1 line Make sure that we do not clear the down flag on the BRI
+ during PTMP link transients. Also refix SS7 audio that the early
+ media patch broke. ........
+
+2009-05-06 17:39 +0000 [r192636-192809] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 192808 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r192808 | file | 2009-05-06 14:38:51 -0300 (Wed, 06 May 2009) |
+ 10 lines Fix a bug where a timer would be created but not
+ acknowledged. This scenario crept up if chan_iax2 was loaded with
+ no configuration file present. It would create a timer and tell
+ it to go at an interval but the thread that normally acknowledges
+ it would not be created because no configuration file was
+ present. The timer will now be closed if no configuration file is
+ present. (closes issue #15014) Reported by: madkins ........
+
+ * /, channels/chan_sip.c: Merged revisions 192634 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r192634 | file | 2009-05-06 10:34:35 -0300 (Wed, 06 May 2009) |
+ 14 lines Merged revisions 192633 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r192633 | file | 2009-05-06 10:30:51 -0300 (Wed, 06 May 2009) | 7
+ lines Update some old logic to stop both begin and end DTMF
+ frames from reaching the core if rfc2833 is not enabled. (closes
+ issue #15036) Reported by: dimas Patches: v1-15036.patch uploaded
+ by dimas (license 88) ........ ................
+
+2009-05-05 20:02 +0000 [r192527] Sean Bright <sean@malleable.com>
+
+ * /, static-http/astman.js: Merged revisions 192525 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r192525 | seanbright | 2009-05-05 15:57:49 -0400
+ (Tue, 05 May 2009) | 18 lines Merged revisions 192524 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r192524 | seanbright | 2009-05-05 15:56:11 -0400 (Tue, 05 May
+ 2009) | 11 lines Fix Javascript error when using astman.js in
+ Internet Explorer. Internet Explorer (tested with 7.0) does not
+ like trailing commas on constructs like object initializers, so
+ get rid of them to avoid some errors. (closes issue #15026)
+ Reported by: rajnishgiri Patches: bug15026.patch uploaded by
+ seanbright (license 71) Tested by: seanbright ........
+ ................
+
+2009-05-05 18:26 +0000 [r192401-192473] Joshua Colp <jcolp@digium.com>
+
+ * /, main/features.c: Merged revisions 192462 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r192462 | file | 2009-05-05 15:23:58 -0300 (Tue, 05 May 2009) |
+ 15 lines Merged revisions 192454 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r192454 | file | 2009-05-05 15:22:27 -0300 (Tue, 05 May 2009) | 8
+ lines Fix an incorrect assumption that certain values on the
+ channel will always exist when they may not. The CDR code
+ involved with bridges wrongly assumed that the currently
+ executing application and data values will always exist. It is
+ possible for this to be false when call forwarding is involved.
+ (closes issue #14984) Reported by: gincantalupo ........
+ ................
+
+ * apps/app_followme.c, /: Merged revisions 192430 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r192430 | file | 2009-05-05 14:46:51 -0300 (Tue, 05 May 2009) |
+ 12 lines Merged revisions 192429 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r192429 | file | 2009-05-05 14:43:30 -0300 (Tue, 05 May 2009) | 5
+ lines Fix a bug where the followme application would continue
+ trying numbers after the caller hung up. (closes issue #13624)
+ Reported by: sgenyuk ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 192387 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r192387 |
+ file | 2009-05-05 11:22:47 -0300 (Tue, 05 May 2009) | 10 lines
+ Fix a bug with setting t38pt_udptl at the user or peer level. If
+ an incoming call authenticated as a user or peer and t38pt_udptl
+ was not set to yes in general then no UDPTL session would be
+ present and any T38 related things would fail. This commit
+ changes it so that if after authenticating T38 is enabled but no
+ UDPTL session is present one will be created. (issue AST-215)
+ ........
+
+2009-05-05 13:37 +0000 [r192281-192359] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/astobj2.c, include/asterisk/stringfields.h, /, main/utils.c:
+ Merged revisions 192357 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r192357 |
+ kpfleming | 2009-05-05 15:18:21 +0200 (Tue, 05 May 2009) | 5
+ lines Correct some flaws in the memory accounting code for
+ stringfields and ao2 objects Under some conditions, the memory
+ allocation for stringfields and ao2 objects would not have
+ supplied valid file/function names for MALLOC_DEBUG tracking, so
+ this commit corrects that. ........
+
+ * main/astobj2.c, main/datastore.c, main/channel.c, /,
+ include/asterisk/astobj2.h, include/asterisk/datastore.h,
+ include/asterisk/channel.h: Merged revisions 192318 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r192318 | kpfleming | 2009-05-05 12:34:19 +0200 (Tue, 05 May
+ 2009) | 5 lines Properly account for memory allocated for
+ channels and datastores As in previous commits, when channels are
+ allocated (with ast_channel_alloc) or datastores are allocated
+ (with ast_datastore_alloc) properly account for the memory being
+ owned by the caller, instead of the allocator function itself.
+ ........
+
+ * include/asterisk/stringfields.h, /, main/utils.c: Merged
+ revisions 192279 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r192279 |
+ kpfleming | 2009-05-05 10:51:06 +0200 (Tue, 05 May 2009) | 5
+ lines Ensure that string pools allocated to hold stringfields are
+ properly accounted in MALLOC_DEBUG mode This commit modifies the
+ stringfield pool allocator to remember the 'owner' of the
+ stringfield manager the pool is being allocated for, and ensures
+ that pools allocated in the future when fields are populated are
+ owned by that file/function. ........
+
+2009-05-04 22:48 +0000 [r192216] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 192214 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r192214 | dvossel | 2009-05-04 17:44:51 -0500
+ (Mon, 04 May 2009) | 17 lines Merged revisions 192213 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r192213 | dvossel | 2009-05-04 17:37:31 -0500 (Mon, 04 May 2009)
+ | 11 lines global mohinterpret setting is ignored mohinterpret
+ and mohsuggest global variables were not copied over during
+ build_users and build_peers. (closes issue #14728) Reported by:
+ dimas Patches: v1-14728.patch uploaded by dimas (license 88)
+ Tested by: dimas, dvossel ........ ................
+
+2009-05-04 19:30 +0000 [r192172] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, res/res_agi.c: Recorded merge of revisions 192171
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r192171 | tilghman | 2009-05-04 14:29:13 -0500 (Mon, 04
+ May 2009) | 8 lines Restore 'asyncagi break' command to 1.6.1 and
+ higher. (closes issue #14985) Reported by: nikkk Patches:
+ 20090428__bug14985.diff.txt uploaded by tilghman (license 14)
+ 20090429__bug14985__1.6.1.diff.txt uploaded by tilghman (license
+ 14) Tested by: nikkk ........
+
+2009-05-04 19:20 +0000 [r192154] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions
+ 192059 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r192059 |
+ kpfleming | 2009-05-04 18:24:16 +0200 (Mon, 04 May 2009) | 5
+ lines Ensure that astobj2 memory allocations are properly
+ accounted for when MALLOC_DEBUG is used This commit ensures that
+ all astobj2 allocated objects are properly accounted for in
+ MALLOC_DEBUG mode by passing down the file/function/line
+ information from the module/function that actually called the
+ astobj2 allocation function. ........
+
+2009-05-04 18:44 +0000 [r192134] Tilghman Lesher <tlesher@digium.com>
+
+ * autoconf/ast_ext_tool_check.m4, /: Merged revisions 192132 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r192132 | tilghman | 2009-05-04 13:42:56 -0500 (Mon, 04
+ May 2009) | 6 lines Pass libraries in LIBS, not LDFLAGS. (closes
+ issue #14671) Reported by: Chainsaw Patches:
+ asterisk-1.6.0.6-toolcheck-libs-not-ldflags.patch uploaded by
+ Chainsaw (license 723) ........
+
+2009-05-04 17:30 +0000 [r192094] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_forkcdr.c: Resolve grammatical mistakes in the
+ application description in app_forkcdr. (closes issue #14801)
+ Reported by: festr
+
+2009-05-04 10:00 +0000 [r191957] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, configs/modules.conf.sample: Merged revisions 191955 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r191955 | kpfleming | 2009-05-04 11:57:36 +0200 (Mon, 04
+ May 2009) | 8 lines Ensure that by default only one console
+ channel driver is loaded This configuration file was changed to
+ ensure that only one console channel driver (chan_oss) is loaded
+ by default, but the change would only work if chan_console was
+ not built. Now it will work as expected; if chan_alsa or
+ chan_console are built and installed, they will not be loaded
+ unless explicity requested. ........
+
+2009-05-02 18:45 +0000 [r191777] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, main/logger.c: Merged revisions 191775 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r191775 |
+ kpfleming | 2009-05-02 20:39:48 +0200 (Sat, 02 May 2009) | 5
+ lines Fix an error in queue_log file rotation optimization code
+ This code was copy-and-pasted without properly changing
+ references to event_rotate into queue_rotate, so under some
+ conditions the log rotation would rotate queue_log even though it
+ was not necessary. ........
+
+2009-05-02 15:52 +0000 [r191702] Sean Bright <sean@malleable.com>
+
+ * main/asterisk.c, /: Merged revisions 191700 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r191700 |
+ seanbright | 2009-05-02 11:45:07 -0400 (Sat, 02 May 2009) | 1
+ line Update copyright year to 2009 ........
+
+2009-05-01 20:02 +0000 [r191553-191562] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 191560 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r191560 | tilghman | 2009-05-01 15:01:21 -0500 (Fri, 01 May 2009)
+ | 13 lines Merged revisions 191559 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r191559 | tilghman | 2009-05-01 15:00:23 -0500 (Fri, 01 May 2009)
+ | 6 lines SIP Response 410 maps to cause code 22 (or 23), not 1.
+ (closes issue #14993) Reported by: BigJimmy Patches: causepatch
+ uploaded by BigJimmy (license 371) ........ ................
+
+ * channels/chan_iax2.c, /: Merged revisions 191494 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r191494 | tilghman | 2009-05-01 13:18:00 -0500 (Fri, 01 May 2009)
+ | 4 lines Set debug message back to DEBUG level. (closes issue
+ #15007) Reported by: hulber ........
+
+2009-05-01 18:20 +0000 [r191505] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 191489 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r191489 | jpeeler | 2009-05-01 13:09:23 -0500 (Fri, 01 May 2009)
+ | 15 lines Merged revisions 191488 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r191488 | jpeeler | 2009-05-01 12:40:46 -0500 (Fri, 01 May 2009)
+ | 9 lines Fix DTMF not being sent to other side after a partial
+ feature match This fixes a regression from commit 176701. The
+ issue was that ast_generic_bridge never exited after the feature
+ digit timeout had elapsed, which prevented the queued DTMF from
+ being sent to the other side. This issue was reported to me
+ directly. ........ ................
+
+2009-05-01 16:26 +0000 [r191454] Sean Bright <sean@malleable.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. (Related to revision 191041 in branches/1.4)
+
+2009-04-30 17:45 +0000 [r191223-191369] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Merged revisions 191367 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r191367 |
+ tilghman | 2009-04-30 12:40:58 -0500 (Thu, 30 Apr 2009) | 3 lines
+ Detect eaccess (or euidaccess) before using it. Reported by
+ Andrew Lindh via the -dev list. ........
+
+ * main/asterisk.c, /: Merged revisions 191283 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r191283 |
+ tilghman | 2009-04-30 01:47:13 -0500 (Thu, 30 Apr 2009) | 11
+ lines Change working directory to / under certain conditions. If
+ backgrounding and no core will be produced, then changing the
+ directory won't break anything; likewise, if the CWD isn't
+ accessible by the current user, then a core wasn't possible
+ anyway. (closes issue #14831) Reported by: chris-mac Patches:
+ 20090428__bug14831.diff.txt uploaded by tilghman (license 14)
+ 20090430__bug14831.diff.txt uploaded by tilghman (license 14)
+ Tested by: chris-mac ........
+
+ * /, channels/h323/ast_h323.cxx, channels/chan_h323.c: Merged
+ revisions 191219 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r191219 |
+ tilghman | 2009-04-29 18:06:56 -0500 (Wed, 29 Apr 2009) | 2 lines
+ Make H.323 compile with FDLEAK detection code enabled ........
+
+2009-04-29 18:40 +0000 [r191138] David Brooks <dbrooks@digium.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 191136 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r191136 |
+ dbrooks | 2009-04-29 13:32:58 -0500 (Wed, 29 Apr 2009) | 3 lines
+ Removing crufty code that is no longer necessary. Code cleanup.
+ ........
+
+2009-04-29 08:45 +0000 [r190988] TransNexus OSP Development <support@transnexus.com>
+
+ * apps/app_osplookup.c, /: Merged revisions 190830 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r190830 | transnexus | 2009-04-28 17:10:42 +0800 (Tue, 28 Apr
+ 2009) | 2 lines Updated for OSP Toolkit 3.5. ........
+
+2009-04-28 17:33 +0000 [r190906] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/tex/cdrdriver.tex, /: Merged revisions 190904 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r190904 | tilghman | 2009-04-28 12:31:43 -0500 (Tue, 28 Apr 2009)
+ | 2 lines UniqueID column has a maximum size of 150 ........
+
+2009-04-28 14:13 +0000 [r190731-190863] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, Makefile.rules: Merged revisions 190861 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r190861 |
+ kpfleming | 2009-04-28 09:12:09 -0500 (Tue, 28 Apr 2009) | 5
+ lines Remove Makefile rules for bison and flex sources We never,
+ ever want these files to processed automatically, because we
+ store the output files in Subversion and users should never need
+ to rebuild them. ........
+
+ * /, configure, include/asterisk/autoconfig.h.in: Merged revisions
+ 190725 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r190725 | kpfleming | 2009-04-27 14:30:54 -0500 (Mon, 27 Apr
+ 2009) | 13 lines Merged revisions 190721 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r190721 | kpfleming | 2009-04-27 14:29:46 -0500 (Mon, 27 Apr
+ 2009) | 7 lines Fix 'inconsistent line endings' when autoconf
+ 2.63 is used Attempt to make configure script regeneration 'safe'
+ using autoconf 2.63, which embeds a bare CR into the script, thus
+ making Subversion complain about inconsistent line endings This
+ commit changes the MIME type of the configure script to be
+ 'binary' thus making Subversion no longer inspect line endings,
+ and as a bonus 'svn diff' will no longer try to generate diff
+ output for it, which is not generally useful anyway. ........
+ ................
+
+2009-04-27 19:36 +0000 [r190728] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 190726 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r190726 |
+ tilghman | 2009-04-27 14:34:48 -0500 (Mon, 27 Apr 2009) | 4 lines
+ Don't warn on pipe in the System call. (closes issue #14979)
+ Reported by: pj ........
+
+2009-08-10 Tilghman Lesher <tlesher@digium.com>
+
+ * Asterisk 1.6.1.4 released
+
+ * AST-2009-005
+
+2009-07-27 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.1.2 released
+
+ * AST-2009-004
+
+2009-06-05 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.1.1 released
+
+2009-06-04 David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: Additional updates for AST-2009-001
+
+2009-06-04 David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: REGAUTH loop fix related to AST-2009-001
+
+2009-04-27 Leif Madsen <lmadsen@digium.com>
+
+ * Create Asterisk 1.6.1.0
+
+2009-04-20 Leif Madsen <lmadsen@digium.com>
+
+ * Create Asterisk 1.6.1.0-rc5
+
+2009-04-20 17:08 +0000 [r189352] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 189350 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r189350 |
+ file | 2009-04-20 14:05:15 -0300 (Mon, 20 Apr 2009) | 10 lines
+ Fix a bug with non-UDP connections that caused dialogs to not get
+ freed. This issue crept up because of a reference count issue on
+ non-UDP based dialogs. The dialog reference count was increased
+ when transmitting a packet reliably but never decreased. This
+ caused the dialog structure to hang around despite being unlinked
+ from the dialogs container. (closes issue #14919) Reported by:
+ vrban ........
+
+2009-04-20 14:06 +0000 [r189280] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 189278 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r189278 | mmichelson | 2009-04-20 09:05:27 -0500 (Mon, 20 Apr
+ 2009) | 18 lines Merged revisions 189277 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189277 | mmichelson | 2009-04-20 09:04:41 -0500 (Mon, 20 Apr
+ 2009) | 12 lines Move the check for chan->fdno == -1 to after the
+ zombie/hangup check. Many users were finding that their hung up
+ channels were staying up and causing 100% CPU usage. (issue
+ #14723) Reported by: seadweller Patches: 14723_1-4-tip.patch
+ uploaded by mmichelson (license 60) Tested by: falves11, bamby
+ ........ ................
+
+2009-04-18 01:38 +0000 [r189206] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_agent.c: Merged revisions 189204 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r189204 | dvossel | 2009-04-17 20:28:45 -0500
+ (Fri, 17 Apr 2009) | 18 lines Merged revisions 189203 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17 Apr 2009)
+ | 12 lines Fixed autologoff in agents.conf not working when agent
+ logs in via AgentLogin app An agent logs in by calling an
+ extension that calls the AgentLogin app. In agents.conf
+ ackcall=always is set, so when they get a call they have the
+ choice to either acknowledge it or ignore it. autologoff=10 is
+ set as well, so if the agent ignores the call over 10sec one may
+ assume that the agent should be logged out (and in this case
+ hungup on as well), but this was not happening. (closes issue
+ #14091) Reported by: evandro Patches: autologoff.diff uploaded by
+ dvossel (license 671) Review:
+ http://reviewboard.digium.com/r/225/ ........ ................
+
+2009-04-17 21:55 +0000 [r189139] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
+ revisions 189137 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r189137 | rmudgett | 2009-04-17 16:48:10 -0500 (Fri, 17 Apr 2009)
+ | 17 lines Merged revisions 188833,189134 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009)
+ | 4 lines Only disable mISDN DSP if Asterisk DSP is enabled.
+ Leave jitter setting alone. JIRA ABE-1835 ........ r189134 |
+ rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines
+ Modifed/added some debug messages. JIRA ABE-1835 ........
+ ................
+
+2009-04-17 20:21 +0000 [r189103] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 189097 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r189097 |
+ mmichelson | 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) | 13
+ lines Prevent a crash when SIP blonde transferring an unbridged
+ call. If one attempts to use the attended transfer button on a
+ SIP phone to transfer an unbridged call (such as a call to an
+ IVR) but hangs up while the target of the transfer is still
+ ringing, we need to not crash. The problem was that ast_hangup
+ was called from outside the channel thread. AST-211 ........
+
+2009-04-17 19:46 +0000 [r189080] Sean Bright <sean.bright@gmail.com>
+
+ * main/asterisk.c, /: Merged revisions 189077 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r189077 |
+ seanbright | 2009-04-17 15:36:38 -0400 (Fri, 17 Apr 2009) | 1
+ line Fix copy/paste error with 'transmit silence' flag. ........
+
+2009-04-17 17:33 +0000 [r189069] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 189010 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r189010 | mnicholson | 2009-04-17 10:44:18 -0500 (Fri, 17 Apr
+ 2009) | 12 lines Merged revisions 189009 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189009 | mnicholson | 2009-04-17 10:43:09 -0500 (Fri, 17 Apr
+ 2009) | 5 lines Make Busy() application set the CDR disposition
+ to BUSY. (closes issue #14306) Reported by: cristiandimache
+ ........ ................
+
+2009-04-17 14:48 +0000 [r188940-188949] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 188947 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) |
+ 22 lines Merged revisions 188946 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) |
+ 15 lines Fix a bug where a value used to create the channel name
+ was bogus. This commit fixes the scenario where an incoming call
+ is authenticated using a peer entry. Previously the channel name
+ was created using either the username setting from the sip.conf
+ entry or the IP address that the call came from. Now the channel
+ name will be created using the peer name itself. This commit will
+ not change the way the channel name is generated for users or
+ friends. (closes issue #14256) Reported by: Nick_Lewis Patches:
+ chan_sip.c-chname.patch uploaded by Nick (license 657) Tested by:
+ Nick_Lewis, file ........ ................
+
+ * channels/chan_dahdi.c, /: Merged revisions 188938 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r188938 | file | 2009-04-17 11:26:53 -0300 (Fri,
+ 17 Apr 2009) | 11 lines Merged revisions 188937 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188937 | file | 2009-04-17 11:25:57 -0300 (Fri, 17 Apr 2009) | 4
+ lines Fix a situation where the DAHDI channel private structure
+ lock was not unlocked when it should have been. (issue AST-210)
+ ........ ................
+
+2009-04-16 22:05 +0000 [r188776-188838] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 188836 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r188836 | tilghman | 2009-04-16 16:57:37 -0500 (Thu, 16 Apr 2009)
+ | 14 lines Merged revisions 188835 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188835 | tilghman | 2009-04-16 16:41:13 -0500 (Thu, 16 Apr 2009)
+ | 7 lines Only update realtime, if global option rtupdate !=
+ false (closes issue #14885) Reported by: deepesh Patches:
+ 20090413__bug14885.diff.txt uploaded by tilghman (license 14)
+ Tested by: deepesh ........ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 188774 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r188774 | tilghman | 2009-04-16 16:03:31 -0500
+ (Thu, 16 Apr 2009) | 11 lines Merged revisions 188773 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188773 | tilghman | 2009-04-16 16:02:29 -0500 (Thu, 16 Apr 2009)
+ | 4 lines Umask should not be exported into global namespace.
+ (closes issue #14912) Reported by: jcapp ........
+ ................
+
+2009-04-15 22:12 +0000 [r188649] David Vossel <dvossel@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 188647 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r188647 | dvossel | 2009-04-15 17:10:04 -0500
+ (Wed, 15 Apr 2009) | 18 lines Merged revisions 188646 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15 Apr 2009)
+ | 12 lines National prefix inserted even when caller ID not
+ available When the caller ID is restricted, the expected behavior
+ is for the caller id to be blank. In chan_dahdi, the national
+ prefix is placed onto the callers number even if its restricted
+ (empty) causing the caller id to be the national prefix rather
+ than blank. (closes issue #13207) Reported by: shawkris Patches:
+ national_prefix.diff uploaded by dvossel (license 671) Review:
+ http://reviewboard.digium.com/r/220/ ........ ................
+
+2009-04-15 20:20 +0000 [r188473-188596] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/file.c: Merged revisions 188585 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r188585 | mmichelson | 2009-04-15 15:17:33 -0500 (Wed, 15 Apr
+ 2009) | 13 lines Merged revisions 188582 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188582 | mmichelson | 2009-04-15 15:04:20 -0500 (Wed, 15 Apr
+ 2009) | 7 lines Update ast_readvideo_callback to match
+ ast_readaudio_callback. This fixes potential refcount errors that
+ may occur on ast_filestreams. AST-208 ........ ................
+
+ * apps/app_queue.c, /: Merged revisions 188470 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188470 |
+ mmichelson | 2009-04-14 18:28:13 -0500 (Tue, 14 Apr 2009) | 3
+ lines Fix a couple of queue member reference leaks. ........
+
+2009-04-14 17:43 +0000 [r188254-188415] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c, /: Merged revisions 188413 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188413 |
+ file | 2009-04-14 14:40:50 -0300 (Tue, 14 Apr 2009) | 5 lines Fix
+ an incorrect clock rate when sending T140 text. (closes issue
+ #14029) Reported by: epicac ........
+
+ * /, channels/chan_sip.c: Merged revisions 188247 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188247 |
+ file | 2009-04-14 10:14:21 -0300 (Tue, 14 Apr 2009) | 7 lines Fix
+ a bug with the change I made yesterday to outbound proxy support.
+ Per discussion with oej on IRC we need the actual IP address, not
+ the outbound proxy IP address, in the sa field. Upon further
+ inspection this should make the behaviour of all other uses of
+ the outbound proxy in the code. ........
+
+2009-04-14 05:46 +0000 [r188208-188212] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 188210 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188210 |
+ tilghman | 2009-04-14 00:45:13 -0500 (Tue, 14 Apr 2009) | 2 lines
+ As suggested by Russell, warn users when their dialplan arguments
+ contain pipes, but not commas. ........
+
+ * /, utils/smsq.c: Merged revisions 188206 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188206 |
+ tilghman | 2009-04-14 00:27:53 -0500 (Tue, 14 Apr 2009) | 6 lines
+ Application delimiter is ',', not '|'. (closes issue #14881)
+ Reported by: stegro Patches: smsq.patch uploaded by stegro
+ (license 752) ........
+
+2009-04-13 19:33 +0000 [r188104] Mark Michelson <mmichelson@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 188102 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r188102 | mmichelson | 2009-04-13 14:31:48 -0500 (Mon, 13 Apr
+ 2009) | 5 lines Fix another crash related to cached realtime
+ music on hold. This was another off-by-one problem caused by
+ moh_register. ........
+
+2009-04-13 16:32 +0000 [r188069] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 188067 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188067 |
+ file | 2009-04-13 13:28:06 -0300 (Mon, 13 Apr 2009) | 10 lines
+ Fix a bug where using an outbound proxy would cause the local
+ address to be 127.0.0.1. Copy the outbound proxy IP address into
+ the SIP dialog structure as the IP address we will be sending to.
+ This has to be done because the logic that determines what local
+ IP address to use in the SIP messages is not aware of an outbound
+ proxy being in place. It only knows what IP address we are
+ sending to. (closes issue #12006) Reported by: mnicholson
+ ........
+
+2009-04-13 14:20 +0000 [r188038] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 188032 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188032 |
+ mmichelson | 2009-04-13 09:17:56 -0500 (Mon, 13 Apr 2009) | 6
+ lines Set all queue variables on both the caller and member
+ channels. This allows for the variables to be accessed if a
+ member macro is run. Thanks to Grigoriy Puzankin for bringing
+ this up on the -dev list. ........
+
+2009-04-10 20:28 +0000 [r187914] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/Makefile, /: Merged revisions 187906 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r187906 |
+ jpeeler | 2009-04-10 15:26:46 -0500 (Fri, 10 Apr 2009) | 12 lines
+ Fix module embedding for chan_h323. Include libchanh323.a in the
+ modules.link file so that all the symbols can be resolved at link
+ time. (closes issue #11966) Reported by: dome Patches:
+ issue_11966.patch uploaded by kpfleming (license 421) Tested by:
+ jpeeler ........
+
+2009-04-10 17:30 +0000 [r187767] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/sip-friends.sql,
+ contrib/scripts/realtime_pgsql.sql, /: Merged revisions 187764
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r187764 | tilghman | 2009-04-10 12:29:34 -0500
+ (Fri, 10 Apr 2009) | 9 lines Merged revisions 187763 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r187763 | tilghman | 2009-04-10 12:28:46 -0500 (Fri, 10
+ Apr 2009) | 2 lines Add lastms column to the contributed table
+ designs ........ ................
+
+2009-04-10 16:54 +0000 [r187723] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, build_tools/embed_modules.xml: Merged revisions 187721 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r187721 | kpfleming | 2009-04-10 11:51:44 -0500 (Fri, 10
+ Apr 2009) | 5 lines clean up some patterns for files to remove
+ add embedding support for bridge and test modules ........
+
+2009-04-10 16:03 +0000 [r187678] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 187674 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r187674 |
+ tilghman | 2009-04-10 10:59:40 -0500 (Fri, 10 Apr 2009) | 4 lines
+ Ensure pvt is not NULL before dereferencing it. (closes issue
+ #14784) Reported by: pj ........
+
+2009-04-10 16:00 +0000 [r187676] Russell Bryant <russell@digium.com>
+
+ * tests/test_heap.c, /: Merged revisions 187675 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r187675 |
+ russell | 2009-04-10 11:00:29 -0500 (Fri, 10 Apr 2009) | 2 lines
+ Disable test modules by default. ........
+
+2009-04-10 03:56 +0000 [r187600] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, main/pbx.c, main/manager.c, /,
+ include/asterisk/linkedlists.h, main/features.c, main/http.c,
+ main/app.c, include/asterisk/lock.h, main/audiohook.c: Merged
+ revisions 187599 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r187599 |
+ tilghman | 2009-04-09 22:55:27 -0500 (Thu, 09 Apr 2009) | 2 lines
+ Modify headers and macros, according to Russell's suggestions on
+ the -dev list ........
+
+2009-04-09 19:14 +0000 [r187495] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 187488 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r187488 | mmichelson | 2009-04-09 13:58:41 -0500 (Thu, 09 Apr
+ 2009) | 24 lines Merged revisions 187484 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r187484 | mmichelson | 2009-04-09 13:51:20 -0500 (Thu, 09 Apr
+ 2009) | 18 lines Handle a SIP race condition (reinvite before an
+ ACK) properly. RFC 5047 explains the proper course of action to
+ take if a reINVITE is received before the ACK from a previous
+ invite transaction. What we are to do is to treat the reINVITE as
+ if it were both an ACK and a reINVITE and process it normally.
+ Later, when we receive the ACK we had been expecting, we will
+ ignore it since its CSeq is less than the current iseqno of the
+ sip_pvt representing this dialog. (closes issue #13849) Reported
+ by: klaus3000 Patches: 13849_v2.patch uploaded by mmichelson
+ (license 60) Tested by: mmichelson, klaus3000 ........
+ ................
+
+2009-04-09 18:54 +0000 [r187486] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, /, include/asterisk/linkedlists.h,
+ include/asterisk/lock.h: Merged revisions 187483 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r187483 | tilghman | 2009-04-09 13:40:01 -0500
+ (Thu, 09 Apr 2009) | 15 lines Merged revisions 187428 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09 Apr 2009)
+ | 8 lines Race condition between ast_cli_command() and 'module
+ unload' could cause a deadlock. Add lock timeouts to avoid this
+ potential deadlock. (closes issue #14705) Reported by: jamessan
+ Patches: 20090320__bug14705.diff.txt uploaded by tilghman
+ (license 14) Tested by: jamessan ........ ................
+
+2009-04-09 17:43 +0000 [r187427] Mark Michelson <mmichelson@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 187421,187424 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r187421 | mmichelson | 2009-04-09 12:30:39 -0500 (Thu,
+ 09 Apr 2009) | 21 lines Fix a crash in res_musiconhold when using
+ cached realtime moh. The moh_register function links an mohclass
+ and then immediately unrefs the class since the container now has
+ a reference. The problem with using realtime music on hold is
+ that the class is allocated, registered, and started in one fell
+ swoop. The refcounting logic resulted in the count being off by
+ one. The same problem did not happen when using a static config
+ because the allocation and registration of an mohclass is a
+ separate operation from starting moh. This also did not affect
+ non-cached realtime moh because the classes are not registered at
+ all. I also have modified res_musiconhold to use the _t_ variants
+ of the ao2_ functions so that more info can be gleaned when
+ attempting to trace the refcounts. I found this to be incredibly
+ helpful for debugging this issue and there's no good reason to
+ remove it. (closes issue #14661) Reported by: sum ........
+ r187424 | mmichelson | 2009-04-09 12:34:39 -0500 (Thu, 09 Apr
+ 2009) | 3 lines Use safe macro practices even though they really
+ aren't necessary. ........
+
+2009-04-09 17:22 +0000 [r187305-187388] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 187381 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r187381 |
+ tilghman | 2009-04-09 12:20:49 -0500 (Thu, 09 Apr 2009) | 4 lines
+ Allow '/' in username portion of register; this is a regression.
+ (closes issue #14668) Reported by: Netview ........
+
+ * /, channels/chan_sip.c, apps/app_sendtext.c: Merged revisions
+ 187363 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r187363 | tilghman | 2009-04-09 11:39:43 -0500 (Thu, 09 Apr 2009)
+ | 10 lines Merged revisions 187362 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r187362 | tilghman | 2009-04-09 11:38:37 -0500 (Thu, 09 Apr 2009)
+ | 3 lines Permit zero-length text messages in SIP. (Related to an
+ issue posted to the -users list, subject "AEL2, BASE64_DECODE and
+ hexadecimal") ........ ................
+
+ * main/asterisk.c, agi/Makefile, build_tools/cflags.xml,
+ utils/Makefile, include/asterisk.h, /, main/Makefile,
+ main/file.c, main/astfd.c (added): Merged revisions 187302 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r187302 | tilghman | 2009-04-08 23:59:05 -0500
+ (Wed, 08 Apr 2009) | 14 lines Merged revisions 187300-187301 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009)
+ | 3 lines Add debugging mode for diagnosing file descriptor
+ leaks. (Related to issue #14625) ........ r187301 | tilghman |
+ 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines Oops,
+ missed this file in the last commit. ........ ................
+
+2009-04-08 16:53 +0000 [r186987-187048] Mark Michelson <mmichelson@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 187046 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r187046 | mmichelson | 2009-04-08 11:52:20 -0500
+ (Wed, 08 Apr 2009) | 16 lines Merged revisions 187045 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed, 08 Apr
+ 2009) | 10 lines Fix a small logical error when loading moh
+ classes. We were unconditionally incrementing the number of
+ mohclasses registered. However, we should actually only increment
+ if the call to moh_register was successful. While this probably
+ has never caused problems, I noticed it and decided to fix it
+ anyway. ........ ................
+
+ * main/channel.c, /: Merged revisions 186985 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186985 | mmichelson | 2009-04-08 10:27:41 -0500 (Wed, 08 Apr
+ 2009) | 30 lines Merged revisions 186984 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr
+ 2009) | 24 lines Make a couple of changes with regards to a new
+ message printed in ast_read(). "ast_read() called with no
+ recorded file descriptor" is a new message added after a bug was
+ discovered. Unfortunately, it seems there are a bunch of places
+ that potentially make such calls to ast_read() and trigger this
+ error message to be displayed. This commit does two things to
+ help to make this message appear less. First, the message has
+ been downgraded to a debug level message if dev mode is not
+ enabled. The message means a lot more to developers than it does
+ to end users, and so developers should take an effort to be sure
+ to call ast_read only when a channel is ready to be read from.
+ However, since this doesn't actually cause an error in operation
+ and is not something a user can easily fix, we should not spam
+ their console with these messages. Second, the message has been
+ moved to after the check for any pending masquerades. ast_read()
+ being called with no recorded file descriptor should not
+ interfere with a masquerade taking place. This could be seen as a
+ simple way of resolving issue #14723. However, I still want to
+ try to clear out the existing ways of triggering this message,
+ since I feel that would be a better resolution for the issue.
+ ........ ................
+
+2009-04-08 05:07 +0000 [r186900] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 186899 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r186899 |
+ tilghman | 2009-04-08 00:06:22 -0500 (Wed, 08 Apr 2009) | 2 lines
+ Add lastms to the require API call. ........
+
+2009-04-08 00:10 +0000 [r186835-186844] Mark Michelson <mmichelson@digium.com>
+
+ * /, formats/format_wav.c, formats/format_wav_gsm.c: Merged
+ revisions 186842 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186842 | mmichelson | 2009-04-07 19:09:28 -0500 (Tue, 07 Apr
+ 2009) | 14 lines Merged revisions 186841 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr
+ 2009) | 8 lines Fix a few typos of the word "frequency." (closes
+ issue #14842) Reported by: jvandal Patches: frequency-typo.diff
+ uploaded by jvandal (license 413) ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 186837 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r186837 |
+ mmichelson | 2009-04-07 19:01:49 -0500 (Tue, 07 Apr 2009) | 7
+ lines Fix bad merge from fix for issue 13867. (closes issue
+ #14686) Reported by: davidw ........
+
+ * main/channel.c, /: Merged revisions 186833 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186833 | mmichelson | 2009-04-07 18:50:56 -0500 (Tue, 07 Apr
+ 2009) | 15 lines Merged revisions 186832 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr
+ 2009) | 8 lines Set the AST_FEATURE_WARNING_ACTIVE flag when a
+ p2p bridge returns AST_BRIDGE_RETRY. Without this flag set,
+ warning sounds will not be properly played to either party of the
+ bridge. (closes issue #14845) Reported by: adomjan ........
+ ................
+
+2009-04-07 22:33 +0000 [r186806] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_macro.c: Merged revisions 186799 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186799 | tilghman | 2009-04-07 17:23:46 -0500 (Tue, 07 Apr 2009)
+ | 10 lines Merged revisions 186775 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186775 | tilghman | 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009)
+ | 3 lines Fix Macro documentation to match current (and intended)
+ behavior. (See -dev mailing list) ........ ................
+
+2009-04-07 20:53 +0000 [r186722] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c, /: Merged revisions 186720 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186720 | mmichelson | 2009-04-07 15:46:18 -0500 (Tue, 07 Apr
+ 2009) | 12 lines Merged revisions 186719 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186719 | mmichelson | 2009-04-07 15:43:49 -0500 (Tue, 07 Apr
+ 2009) | 6 lines Ensure that \r\n is printed after the ActionID in
+ an OriginateResponse. (closes issue #14847) Reported by: kobaz
+ ........ ................
+
+2009-04-03 20:21 +0000 [r186466] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 186461 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r186461 | kpfleming | 2009-04-03 15:20:01 -0500
+ (Fri, 03 Apr 2009) | 11 lines Merged revisions 186458 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03 Apr
+ 2009) | 5 lines Fix a bug where DAHDI/Zaptel channels would not
+ properly switch formats when requested Don't offer
+ AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could
+ provide a slight performance benefit, the translation core in
+ Asterisk has some flaws when a channel driver offers multiple raw
+ formats. this fix is much simpler than fixing the translation
+ core to solve that issue (although that will be done later).
+ ........ ................
+
+2009-04-03 20:04 +0000 [r186448] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged
+ revisions 186444,186447 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186444 | tilghman | 2009-04-03 14:30:34 -0500 (Fri, 03 Apr 2009)
+ | 14 lines Merged revisions 186415 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009)
+ | 7 lines Distinguish in a sent email between simple sends and
+ forwards. (closes issue #11678) Reported by: jamessan Patches:
+ 20090330__bug11678.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman, lmadsen ........ ................ r186447 |
+ tilghman | 2009-04-03 14:59:55 -0500 (Fri, 03 Apr 2009) | 9 lines
+ Merged revisions 186445 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186445 | tilghman | 2009-04-03 14:56:48 -0500 (Fri, 03 Apr 2009)
+ | 2 lines Found a conflict in the last commit, due to multiple
+ targets ........ ................
+
+2009-04-03 16:38 +0000 [r186381] David Vossel <dvossel@digium.com>
+
+ * /, main/audiohook.c: Merged revisions 186379 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r186379 |
+ dvossel | 2009-04-03 11:29:47 -0500 (Fri, 03 Apr 2009) | 4 lines
+ audio_audiohook_write_list() did not correctly update sample size
+ after ast_translate. audio_audiohook_write_list() did not take
+ into account that the sample size may change after translation
+ depending on if the original frame is is 8khz or 16khz. the
+ sample size is now updated after translating to reflect this
+ possibility. This caused the audio on the receiving end to sound
+ terrible. Thanks to jcolp and mmichelson for helping me work this
+ out. (issue AST-197) ........
+
+2009-04-03 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.1.0-rc4 released.
+
+2009-04-03 15:54 +0000 [r186323] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/crypto.h, /: Merged revisions 186321 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r186321 | file | 2009-04-03 12:52:50 -0300 (Fri,
+ 03 Apr 2009) | 12 lines Merged revisions 186320 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5
+ lines Fix a problem with the crypto variable definitions not
+ actually being defined properly. (closes issue #14804) Reported
+ by: jvandal ........ ................
+
+2009-04-03 14:33 +0000 [r186288] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 186286 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r186286 | mmichelson | 2009-04-03 09:32:05 -0500 (Fri, 03 Apr
+ 2009) | 20 lines Fix the ability to retrieve voicemail messages
+ from IMAP. A recent change made interactive vm_states no longer
+ get added to the list of vm_states and instead get stored in
+ thread-local storage. In trunk and all the 1.6.X branches, the
+ problem is that when we search for messages in a voicemail box,
+ we would attempt to update the appropriate vm_state struct by
+ directly searching in the list of vm_states instead of using the
+ get_vm_state_by_imap_user function. This meant we could not find
+ the interactive vm_state that we wanted. (closes issue #14685)
+ Reported by: BlargMaN Patches: 14685.patch uploaded by mmichelson
+ (license 60) Tested by: BlargMaN, qualleyiv, mmichelson ........
+
+2009-04-03 02:06 +0000 [r186232] Russell Bryant <russell@digium.com>
+
+ * cdr/cdr_radius.c, /: Merged revisions 186230 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186230 | russell | 2009-04-02 21:03:48 -0500 (Thu, 02 Apr 2009)
+ | 29 lines Merged revisions 186229 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009)
+ | 21 lines Fix a memory leak in cdr_radius. I came across this
+ while doing some testing of my ast_channel_ao2 branch. After
+ running a test overnight that generated over 5 million calls,
+ Asterisk had taken up about 1 GB of my system memory. So, I
+ re-ran the test with MALLOC_DEBUG turned on. However, it showed
+ no leaks in Asterisk during the test, even though Asterisk was
+ still consuming it somehow. Instead, I turned to valgrind, which
+ when run with --leak-check=full, told me exactly where the leak
+ came from, which was from allocations inside the radiusclient-ng
+ library. This explains why MALLOC_DEBUG did not report it. After
+ a bit of analysis, I found that we were leaking a little bit of
+ memory every time a CDR record was passed to cdr_radius. I don't
+ actually have a radius server set up to receive CDR records.
+ However, I always have my development systems compile and install
+ all modules. In addition to making sure there are not build
+ errors across modules, always loading modules helps find bugs
+ like this, too, so it is strongly recommend for all developers.
+ ........ ................
+
+2009-04-02 21:59 +0000 [r186177] Mark Michelson <mmichelson@digium.com>
+
+ * configs/features.conf.sample, /: Merged revisions 186175 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r186175 | mmichelson | 2009-04-02 16:56:21 -0500
+ (Thu, 02 Apr 2009) | 11 lines Merged revisions 186174 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr
+ 2009) | 5 lines Fix instructions in one-step parking comment to
+ make more sense. Changed a capital K to a lowercase k. ........
+ ................
+
+2009-04-02 17:27 +0000 [r186108] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 186101 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r186101 | kpfleming | 2009-04-02 12:26:07 -0500
+ (Thu, 02 Apr 2009) | 9 lines Merged revisions 186081 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02
+ Apr 2009) | 3 lines ensure that the buffer passed to
+ DAHDI_SET_BUFINFO is fully initialized ........ ................
+
+2009-04-02 17:14 +0000 [r186062] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+ 186060 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009)
+ | 16 lines Merged revisions 186059 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500
+ (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02
+ Apr 2009) | 2 lines Fix for AST-2009-003 ........
+ ................ ................
+
+2009-04-02 13:53 +0000 [r185956] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 185953 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r185953 | kpfleming | 2009-04-02 08:51:44 -0500
+ (Thu, 02 Apr 2009) | 11 lines Merged revisions 185952 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02 Apr
+ 2009) | 5 lines the DAHDI_GETCONF, DAHDI_SETCONF and
+ DAHDI_GET_PARAMS ioctls were recently corrected to show that they
+ do, in fact, read data from userspace as part of their work. due
+ to this fix, valgrind now reports a number of cases where
+ chan_dahdi passed an uninitialized (or partially) buffer to these
+ ioctls, which could lead to unexpected behavior. this patch
+ corrects chan_dahdi to ensure that buffers passed to these ioctls
+ are always fully initialized. ........ ................
+
+2009-04-01 19:06 +0000 [r185848] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 185846 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185846 | dvossel | 2009-04-01 14:03:32 -0500 (Wed, 01 Apr 2009)
+ | 16 lines Merged revisions 185845 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009)
+ | 10 lines Fixes issue with dropped calles due to re-Invite glare
+ and re-Invites never executing after a 491 Acknowledgement for
+ 491 responses were never being processed because it didn't match
+ our pending invite's seqno. Since the ACK was never processed,
+ the 491 frame would continue to be retransmitted until eventually
+ the call was dropped due to max retries. Now during a pending
+ invite, if we receive another invite, we send an 491 and hold on
+ to that glare invite's seqno in the "glareinvite" variable for
+ that sip_pvt struct. When ACK's are received, we first check to
+ see if it is in response to our pending invite, if not we check
+ to see if it is in response to a glare invite. In this case, it
+ is in response to the glare invite and must be dealt with or the
+ call is dropped. I've changed the wait time for resending the
+ re-Invite after receving a 491 response to comply with RFC 3261.
+ Before this patch the scheduled re-Invite would only change a
+ flag indicating that the re-Invite should be sent out, now it
+ actually sends it out as well. (closes issue #12013) Reported by:
+ alx Review: http://reviewboard.digium.com/r/213/ ........
+ ................
+
+2009-04-01 13:50 +0000 [r185774] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 185772 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185772 | russell | 2009-04-01 08:48:26 -0500 (Wed, 01 Apr 2009)
+ | 14 lines Merged revisions 185771 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009)
+ | 6 lines Fix a case where DTMF could bypass audiohooks. This
+ change fixes a situation where an audiohook that wants DTMF would
+ not actually get it. This is in the code path where we end DTMF
+ digit length emulation while handling a NULL frame. ........
+ ................
+
+2009-03-31 22:38 +0000 [r185666] Kevin P. Fleming <kpfleming@digium.com>
+
+ * utils, /: Merged revisions 185664 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r185664 |
+ kpfleming | 2009-03-31 17:35:07 -0500 (Tue, 31 Mar 2009) | 1 line
+ ignore copied (generated) file ........
+
+2009-03-31 22:05 +0000 [r185471-185602] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 185600 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185600 | mmichelson | 2009-03-31 17:02:48 -0500 (Tue, 31 Mar
+ 2009) | 12 lines Merged revisions 185599 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185599 | mmichelson | 2009-03-31 17:00:01 -0500 (Tue, 31 Mar
+ 2009) | 6 lines Fix crash that would occur if an empty member was
+ specified in queues.conf. (closes issue #14796) Reported by: pida
+ ........ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 185469 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r185469 | mmichelson | 2009-03-31 14:46:18 -0500
+ (Tue, 31 Mar 2009) | 14 lines Merged revisions 185468 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185468 | mmichelson | 2009-03-31 14:45:30 -0500 (Tue, 31 Mar
+ 2009) | 8 lines Fix Russian voicemail intro to say the word
+ "messages" properly. (closes issue #14736) Reported by: chappell
+ Patches: voicemail_no_messages.diff uploaded by chappell (license
+ 8) ........ ................
+
+2009-03-31 17:48 +0000 [r185427] David Brooks <dbrooks@digium.com>
+
+ * /, channels/chan_gtalk.c: Merged revisions 185363 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r185363 | dbrooks | 2009-03-31 11:46:57 -0500
+ (Tue, 31 Mar 2009) | 44 lines Merged revisions 185362 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009)
+ | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains
+ extra whitespaces To drill into the xmpp to find the capabilities
+ between channels, chan_gtalk calls iks_child() and iks_next().
+ iks_child() and iks_next() are functions in the iksemel xml
+ parsing library that traverse xml nodes. The bug here is that
+ both iks_child() and iks_next() will return the next iks_struct
+ node *regardless* of type. chan_gtalk expects the next node to be
+ of type IKS_TAG, which in most cases, it is, but in this case (a
+ call being made from the Empathy IM client), there exists
+ iks_struct nodes which are not IKS_TAG data (they are extraneous
+ whitespaces), and chan_gtalk doesn't handle that case, so
+ capabilities don't match, and a call cannot be made.
+ iks_first_tag() and iks_next_tag(), on the other hand, will not
+ return the very next iks_struct, but will check to see if the
+ next iks_struct is of type IKS_TAG. If it isn't, it will be
+ skipped, and the next struct of type IKS_TAG it finds will be
+ returned. This assures that chan_gtalk will find the iks_struct
+ it is looking for. This fix simply changes all calls to
+ iks_child() and iks_next() to become calls to iks_first_tag() and
+ iks_next_tag(), which resolves the capability matching. The
+ following is a payload listing from Empathy, which, due to the
+ extraneous whitespace, will not be parsed correctly by iksemel:
+ <iq from='dbrooksjab@235-22-24-10/Telepathy'
+ to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'>
+ <session xmlns='http://www.google.com/session'
+ initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate'
+ id='1837267342'> <description
+ xmlns='http://www.google.com/session/phone'> <payload-type
+ clockrate='16000' name='speex' id='96'/> <payload-type
+ clockrate='8000' name='PCMA' id='8'/> <payload-type
+ clockrate='8000' name='PCMU' id='0'/> <payload-type
+ clockrate='90000' name='MPA' id='97'/> <payload-type
+ clockrate='16000' name='SIREN' id='98'/> <payload-type
+ clockrate='8000' name='telephone-event' id='99'/> </description>
+ </session> </iq> Review: http://reviewboard.digium.com/r/181/
+ ........ ................
+
+2009-03-31 14:57 +0000 [r185263] Russell Bryant <russell@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 185261 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r185261 |
+ russell | 2009-03-31 09:53:45 -0500 (Tue, 31 Mar 2009) | 5 lines
+ Don't free() an astobj2 object. (closes issue #14672) Reported
+ by: makoto ........
+
+2009-03-31 14:10 +0000 [r185199] Joshua Colp <jcolp@digium.com>
+
+ * /, main/audiohook.c: Merged revisions 185197 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185197 | file | 2009-03-31 11:07:36 -0300 (Tue, 31 Mar 2009) |
+ 15 lines Merged revisions 185196 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8
+ lines Fix crash when moving audiohooks between channels. Handle
+ the scenario where we are called to move audiohooks between
+ channels and the source channel does not actually have any on it.
+ (closes issue #14734) Reported by: corruptor ........
+ ................
+
+2009-03-30 20:50 +0000 [r185126-185127] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn_config.c, /, configs/misdn.conf.sample: Merged
+ revisions 185123 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185123 | rmudgett | 2009-03-30 15:42:14 -0500 (Mon, 30 Mar 2009)
+ | 9 lines Merged revisions 185121 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009)
+ | 1 line Update the channel allocation method documentation.
+ ........ ................
+
+ * channels/misdn/isdn_lib.c, /: Merged revisions 185122 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r185122 | rmudgett | 2009-03-30 15:41:24 -0500
+ (Mon, 30 Mar 2009) | 26 lines Merged revisions 185120 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009)
+ | 19 lines Make chan_misdn BRI TE side normally defer channel
+ selection to the NT side. Channel allocation collisions are not
+ handled by chan_misdn very well. This patch simply avoids the
+ problem for BRI only. For PRI, allocation collisions are still
+ possible but less likely since there are simply more channels
+ available and each end could use a different allocation strategy.
+ misdn.conf options available: te_choose_channel - Use to force
+ the TE side to allocate channels. method - Specify the channel
+ allocation strategy. (closes issue #13488) Reported by:
+ Christian_Pinedo Patches: isdn_lib.patch.txt uploaded by crich
+ Tested by: crich, siepkes, festr ........ ................
+
+2009-03-30 16:47 +0000 [r185088] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 185072 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185072 | mmichelson | 2009-03-30 11:26:48 -0500 (Mon, 30 Mar
+ 2009) | 45 lines Merged revisions 185031 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar
+ 2009) | 39 lines Fix queue weight behavior so that calls in
+ low-weight queues are not inappropriately blocked. (This is
+ copied and pasted from the review request I made for this patch)
+ Asterisk has some odd behavior when queue weights are used. The
+ current logic used when potentially calling a queue member is: If
+ the member we are going to call is part of another queue and
+ _that other queue has any callers in it_ and has a higher weight
+ than the queue we are calling from, then don't try to contact
+ that member. The issue here is what I have marked with
+ underscores. If the higher-weighted queue has any callers in it
+ at all, then the queue member will be unreachable from the
+ lower-weighted queue. This has the potential to be really really
+ bad if using a queue strategy, such as leastrecent or
+ fewestcalls, with the potential to call the same member
+ repeatedly. The fix proposed by garychen on issue 13220 is very
+ simple and, as far as I can see, works well for this situation.
+ With this set of changes, the logic used becomes: If the member
+ we are going to call is part of another queue, the other queue
+ has a higher weight than the queue we are calling from, and the
+ higher weight queue has at least as many callers as available
+ members, then do not try to contact the queue member. If the
+ higher weighted queue has fewer callers than available members,
+ then there is no reason to deny the call to this member since the
+ other queue can afford to spare a member. Since the fix involved
+ writing a generic function for determining the number of
+ available members in the queue, I also modified the is_our_turn
+ function to make use of the new num_available_members function to
+ determine if it is our turn to try calling a member. There is one
+ small behavior change. Before writing this patch, if you had
+ autofill disabled, then if you were the head caller in a queue,
+ you would automatically be told that it was your turn to try
+ calling a member. This did not take into account whether there
+ were actually any queue members available to take the call. Now
+ we actually make sure there is at least one member available to
+ take the call if autofill is disabled. (closes issue #13220)
+ Reported by: garychen Review:
+ http://reviewboard.digium.com/r/202/ ........ ................
+
+2009-03-30 14:41 +0000 [r184950] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 184948 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184948 | file | 2009-03-30 11:37:47 -0300 (Mon, 30 Mar 2009) |
+ 21 lines Merged revisions 184947 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) |
+ 14 lines Improve our handling of T38 in the initial INVITE from a
+ device. We now answer with matching media streams to what is
+ requested. If an INVITE is received with both a T38 and RTP media
+ stream this means we answer with both. For any outgoing calls
+ created as a result of this inbound one no T38 is requested in
+ the initial INVITE. Instead if we start receiving udptl packets
+ we trigger a reinvite on the outbound side. (closes issue #12437)
+ Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu
+ Review: http://reviewboard.digium.com/r/208/ ........
+ ................
+
+2009-03-30 13:57 +0000 [r184912] Russell Bryant <russell@digium.com>
+
+ * channels/h323/Makefile.in, /: Merged revisions 184910 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r184910 | russell | 2009-03-30 08:55:44 -0500 (Mon, 30
+ Mar 2009) | 4 lines Fix build error when chan_h323 is not being
+ built. (reported by cai1982 in #asterisk-dev) ........
+
+2009-03-29 05:52 +0000 [r184840-184845] Russell Bryant <russell@digium.com>
+
+ * apps/app_followme.c, /: Merged revisions 184843 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184843 | russell | 2009-03-29 00:52:20 -0500 (Sun, 29 Mar 2009)
+ | 13 lines Merged revisions 184842 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009)
+ | 5 lines Ensure targs variable is fully initialized. (closes
+ issue #14758) Reported by: tim_ringenbach ........
+ ................
+
+ * channels/Makefile, /: Merged revisions 184838 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184838 |
+ russell | 2009-03-29 00:32:04 -0500 (Sun, 29 Mar 2009) | 8 lines
+ Simplify chan_h323 build to not require a second run of "make".
+ (closes issue #14715) Reported by: jthurman Patches:
+ h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license
+ 614) Tested by: tzafrir, russell ........
+
+2009-03-27 19:17 +0000 [r184765] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c, main/timing.c, main/channel.c, /,
+ include/asterisk/timing.h, include/asterisk/channel.h: Merged
+ revisions 184762 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184762 |
+ kpfleming | 2009-03-27 14:10:32 -0500 (Fri, 27 Mar 2009) | 12
+ lines Improve timing interface to remember which provider
+ provided a timer The ability to load/unload timing interfaces is
+ nice, but it means that when a timer is allocated, it may come
+ from provider A, but later provider B becomes the 'preferred'
+ provider. If this happens, all timer API calls on the timer that
+ was provided by provider A will actually be handed to provider B,
+ which will say WTF and return an error. This patch changes the
+ timer API to include a pointer to the provider of the timer
+ handle so that future operations on the timer will be forwarded
+ to the proper provider. (closes issue #14697) Reported by: moy
+ Review: http://reviewboard.digium.com/r/211/ ........
+
+2009-03-27 18:09 +0000 [r184728] Russell Bryant <russell@digium.com>
+
+ * main/manager.c, /, apps/app_minivm.c: Merged revisions 184726 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r184726 | russell | 2009-03-27 13:04:43 -0500 (Fri, 27
+ Mar 2009) | 2 lines Use ast_random() instead of rand() to ensure
+ we use the best RNG available. ........
+
+2009-03-27 15:54 +0000 [r184675] Joshua Colp <jcolp@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 184673 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184673 |
+ file | 2009-03-27 12:46:46 -0300 (Fri, 27 Mar 2009) | 7 lines Fix
+ speech structure leak in the AGI speech recognition integration.
+ The AGI dialplan applications did not destroy the speech
+ structure automatically if it was not destroyed by the running
+ AGI script. They will now do this. (issue LUMENVOX-15) ........
+
+2009-03-27 14:04 +0000 [r184631] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c, include/asterisk/utils.h, main/pbx.c, /,
+ res/ais/evt.c, main/event.c, pbx/pbx_dundi.c: Merged revisions
+ 184630 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184630 |
+ russell | 2009-03-27 09:00:18 -0500 (Fri, 27 Mar 2009) | 2 lines
+ Change g_eid to ast_eid_default. ........
+
+2009-03-27 13:22 +0000 [r184587] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 184566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184566 | file | 2009-03-27 10:15:26 -0300 (Fri, 27 Mar 2009) |
+ 16 lines Merged revisions 184565 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9
+ lines Fix an issue where nat=yes would not always take effect for
+ the RTP session on outgoing calls. If calls were placed using an
+ IP address or hostname the global nat setting was copied over but
+ was not set on the RTP session itself. This caused the RTP stack
+ to not perform symmetric RTP actions. (closes issue #14546)
+ Reported by: acunningham ........ ................
+
+2009-03-27 02:25 +0000 [r184513-184547] Russell Bryant <russell@digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 184531 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r184531 | russell | 2009-03-26 21:20:23 -0500 (Thu, 26 Mar 2009)
+ | 20 lines Fix some issues with rwlock corruption that caused
+ deadlock like symptoms. When dvossel and I were doing some load
+ testing last week, we noticed that we could make Asterisk trunk
+ lock up instantly when we started generating a bunch of calls.
+ The backtraces of locked threads were bizarre, and many were
+ stuck on an _unlock_ of an rwlock. The changes are: 1) Fix a
+ number of places where a backtrace would be loaded into an
+ invalid index of the backtrace array. It's an off by one error,
+ which ends up writing over the rwlock itself. 2) Ensure that in
+ the array of held locks, we NULL out an index once it is not
+ being used so that it's not confusing when analyzing its
+ contents. 3) Remove a bunch of logging referring to an rwlock
+ operating being done with "deep reentrancy". It is normal for
+ _many_ threads to hold a read lock on an rwlock. ........
+
+ * /, main/file.c: Merged revisions 184515 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184515 |
+ russell | 2009-03-26 20:40:28 -0500 (Thu, 26 Mar 2009) | 2 lines
+ Don't act surprised if we get a -1 indication. ........
+
+ * include/asterisk/heap.h, /, main/heap.c: Merged revisions 184512
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r184512 | russell | 2009-03-26 20:35:56 -0500 (Thu, 26
+ Mar 2009) | 2 lines Pass more useful information through to lock
+ tracking when DEBUG_THREADS is on. ........
+
+2009-03-26 22:19 +0000 [r184451] Kevin P. Fleming <kpfleming@digium.com>
+
+ * sounds/Makefile, /: Merged revisions 184448 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184448 | kpfleming | 2009-03-26 17:18:14 -0500 (Thu, 26 Mar
+ 2009) | 9 lines Merged revisions 184447 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184447 | kpfleming | 2009-03-26 17:17:32 -0500 (Thu, 26 Mar
+ 2009) | 3 lines use new, improved 8kHz prompts ........
+ ................
+
+2009-03-26 21:18 +0000 [r184394] David Vossel <dvossel@digium.com>
+
+ * /, apps/app_test.c: Merged revisions 184389 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184389 | dvossel | 2009-03-26 16:09:37 -0500 (Thu, 26 Mar 2009)
+ | 14 lines Merged revisions 184388 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184388 | dvossel | 2009-03-26 16:07:32 -0500 (Thu, 26 Mar 2009)
+ | 8 lines pri loop TestClient/TestServer fails: server SEND DTMF
+ 8 app_test was failing when sending the last DTMF digit, 8,
+ because of the 100ms pause issued after DTMF is sent. During this
+ pause the other side would hang up causing the test to look like
+ it failed. Now the other side waits a second before hanging up.
+ (closes issue #12442) Reported by: tzafrir ........
+ ................
+
+2009-03-25 22:13 +0000 [r184325-184345] Russell Bryant <russell@digium.com>
+
+ * /, main/event.c: Merged revisions 184344 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184344 |
+ russell | 2009-03-25 17:11:35 -0500 (Wed, 25 Mar 2009) | 2 lines
+ Remove unneeded AST_LIST_ENTRY() and comment on the purpose of
+ ast_event_ref. ........
+
+ * channels/chan_iax2.c, channels/chan_dahdi.c,
+ include/asterisk/event.h, channels/chan_skinny.c, res/ais/evt.c,
+ main/event.c, include/asterisk/strings.h, main/asterisk.c,
+ channels/chan_mgcp.c, apps/app_voicemail.c,
+ channels/chan_unistim.c, include/asterisk/devicestate.h, /,
+ channels/chan_sip.c, main/devicestate.c,
+ include/asterisk/_private.h: Merged revisions 184339 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r184339 | russell | 2009-03-25 16:57:19 -0500 (Wed, 25 Mar 2009)
+ | 35 lines Improve performance of the ast_event cache
+ functionality. This code comes from
+ svn/asterisk/team/russell/event_performance/. Here is a summary
+ of the changes that have been made, in order of both invasiveness
+ and performance impact, from smallest to largest. 1) Asterisk
+ 1.6.1 introduces some additional logic to be able to handle
+ distributed device state. This functionality comes at a cost. One
+ relatively minor change in this patch is that the extra
+ processing required for distributed device state is now
+ completely bypassed if it's not needed. 2) One of the things that
+ I noticed when profiling this code was that a _lot_ of time was
+ spent doing string comparisons. I changed the way strings are
+ represented in an event to include a hash value at the front. So,
+ before doing a string comparison, we do an integer comparison on
+ the hash. 3) Finally, the code that handles the event cache has
+ been re-written. I tried to do this in a such a way that it had
+ minimal impact on the API. I did have to change one API call,
+ though - ast_event_queue_and_cache(). However, the way it works
+ now is nicer, IMO. Each type of event that can be cached (MWI,
+ device state) has its own hash table and rules for hashing and
+ comparing objects. This by far made the biggest impact on
+ performance. For additional details regarding this code and how
+ it was tested, please see the review request. (closes issue
+ #14738) Reported by: russell Review:
+ http://reviewboard.digium.com/r/205/ ........
+
+ * /: add reviewboard:url property.
+
+2009-03-25 19:26 +0000 [r184282] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 184280 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184280 |
+ file | 2009-03-25 16:22:06 -0300 (Wed, 25 Mar 2009) | 5 lines Fix
+ issue with a T38 reinvite being sent even if not configured to do
+ so. If we receive a T38 request negotiate control frame we should
+ only attempt to do so if the option is enabled on the dialog.
+ ........
+
+2009-03-25 15:12 +0000 [r184223] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/asterisk.c, /: Merged revisions 184220 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184220 | eliel | 2009-03-25 10:38:19 -0400 (Wed, 25 Mar 2009) |
+ 19 lines Merged revisions 184188 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) |
+ 13 lines Avoid destroying the CLI line when moving the cursor
+ backward and trying to autocomplete. When moving the cursor
+ backward and pressing TAB to autocomplete, a NULL is put in the
+ line and we are loosing what we have already wrote after the
+ actual cursor position. (closes issue #14373) Reported by: eliel
+ Patches: asterisk.c.patch uploaded by eliel (license 64) Tested
+ by: lmadsen ........ ................
+
+2009-03-25 01:55 +0000 [r184149] Russell Bryant <russell@digium.com>
+
+ * main/timing.c, utils/Makefile, /, include/asterisk/compat.h:
+ Merged revisions 184147 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184147 |
+ russell | 2009-03-24 20:42:10 -0500 (Tue, 24 Mar 2009) | 5 lines
+ Fix build issues on Mac OSX. (closes issue #14714) Reported by:
+ ygor ........
+
+2009-03-24 22:42 +0000 [r184081] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_senddtmf.c, /: Merged revisions 184079 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184079 | mmichelson | 2009-03-24 17:40:39 -0500 (Tue, 24 Mar
+ 2009) | 15 lines Merged revisions 184078 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar
+ 2009) | 9 lines Change NULL pointer check to be ast_strlen_zero.
+ The 'digit' variable is guaranteed to be non-NULL, so the if
+ statement could never evaluate true. Changing to ast_strlen_zero
+ makes the logic correct. This was found while reviewing
+ ast_channel_ao2 code review. ........ ................
+
+2009-03-24 21:47 +0000 [r184039] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 184037 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r184037 | russell | 2009-03-24 16:40:44 -0500 (Tue, 24 Mar 2009)
+ | 6 lines Exclude slin16, siren7, and siren14 from bandwidth=low
+ and =medium The default codec configuration for chan_iax2 is
+ bandwidth=low. I noticed slin16 being negotiated as the codec in
+ some test calls, but that no longer happens after this change.
+ ........
+
+2009-03-24 15:28 +0000 [r183867-183916] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configs/voicemail.conf.sample: Merged revisions 183914 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r183914 | tilghman | 2009-03-24 10:26:42 -0500
+ (Tue, 24 Mar 2009) | 10 lines Merged revisions 183913 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009)
+ | 3 lines Additionally note that the operator option needs an 'o'
+ extension. (Related to issue #14731) ........ ................
+
+ * /, main/http.c: Merged revisions 183865 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r183865 |
+ tilghman | 2009-03-23 18:28:20 -0500 (Mon, 23 Mar 2009) | 2 lines
+ Allow browsers to cache images and other static content. (This is
+ a regression over 1.4) ........
+
+2009-03-23 18:59 +0000 [r183768] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 183766 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r183766 | mmichelson | 2009-03-23 13:58:03 -0500 (Mon, 23 Mar
+ 2009) | 13 lines Merged revisions 183700 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar
+ 2009) | 7 lines Fix a memory leak in res_monitor.c The only way
+ that this leak would occur is if Monitor were started using the
+ Manager interface and no File: header were given. Discovered
+ while reviewing the ast_channel_ao2 review request. ........
+ ................
+
+2009-03-23 18:12 +0000 [r183703] Leif Madsen <lmadsen@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 183701 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r183701 | lmadsen | 2009-03-23 14:06:40 -0400 (Mon, 23 Mar 2009)
+ | 7 lines Fixes a documentation error introduced during the CLI
+ cleanup at AstriDevCon 2008. (closes issue #14655) Reported by:
+ ulogic Patches: chan_dahdi.patch uploaded by ulogic (license 728)
+ Tested by: lmadsen ........
+
+2009-03-20 17:08 +0000 [r183563] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 183560 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r183560 | russell | 2009-03-20 12:00:58 -0500
+ (Fri, 20 Mar 2009) | 10 lines Merged revisions 183559 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20 Mar 2009)
+ | 2 lines Fix a crash in IAX2 registration handling found during
+ load testing with dvossel. ........ ................
+
+2009-03-19 20:33 +0000 [r183438] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/features.h, apps/app_dial.c, /, main/features.c:
+ Merged revisions 183436 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r183436 | dvossel | 2009-03-19 15:30:39 -0500 (Thu, 19 Mar 2009)
+ | 13 lines Merged revisions 183386 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009)
+ | 6 lines Cleaning up a few things in detect disconnect patch
+ Initialized ast_call_feature in detect_disconnect to avoid
+ accessing uninitialized memory. Cleaned up /param tags in
+ features.h. No longer send dynamic features in
+ ast_feature_detect. issue #11583 ........ ................
+
+2009-03-19 19:19 +0000 [r183333] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 183321 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r183321 | tilghman | 2009-03-19 14:17:31 -0500
+ (Thu, 19 Mar 2009) | 15 lines Merged revisions 183319 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183319 | tilghman | 2009-03-19 14:15:33 -0500 (Thu, 19 Mar 2009)
+ | 8 lines Delay signalling progress until a PRI channel really
+ signals progress. (closes issue #13034) Reported by: klaus3000
+ Patches: 20090316__bug13034.diff.txt uploaded by tilghman
+ (license 14) patch_trunk_183progress_klaus3000.txt uploaded by
+ klaus3000 (license 65) Tested by: klaus3000 ........
+ ................
+
+2009-03-19 18:14 +0000 [r183249] Russell Bryant <russell@digium.com>
+
+ * main/loader.c, /, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Merged revisions 183242 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r183242 | russell | 2009-03-19 13:00:15 -0500 (Thu, 19 Mar 2009)
+ | 10 lines Merged revisions 183241 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009)
+ | 2 lines Remove the use of RTLD_NOLOAD, as it is not behaving
+ like expected. ........ ................
+
+2009-03-19 18:11 +0000 [r183246] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 183244 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r183244 |
+ mmichelson | 2009-03-19 13:10:34 -0500 (Thu, 19 Mar 2009) | 16
+ lines Fix a memory leak associated with queues. For every attempt
+ that app_queue made to place an outbound call to a queue member,
+ we would allocate a queue_end_bridge structure. When the bridge
+ for the call had completed, we would free the structure.
+ Unfortunately not all call attempts actually end up bridged to a
+ member, so we need to be more selective of when to allocate the
+ structure. With this change, the allocation occurs in an area
+ where we can guarantee that the call will be bridged. (closes
+ issue #14680) Reported by: caspy Patches: 14680.patch uploaded by
+ mmichelson (license 60) Tested by: caspy ........
+
+2009-03-19 17:08 +0000 [r183198] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/features.h, apps/app_dial.c, /, main/features.c:
+ Merged revisions 183172 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r183172 | dvossel | 2009-03-19 11:28:33 -0500 (Thu, 19 Mar 2009)
+ | 20 lines Merged revisions 183126 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009)
+ | 17 lines Allow disconnect feature before a call is bridged
+ feature.conf has a disconnect option. By default this option is
+ set to '*', but it could be anything. If a user wishes to
+ disconnect a call before the other side answers, only '*' will
+ work, regardless if the disconnect option is set to something
+ else. This is because features are unavailable until bridging
+ takes place. The default disconnect option, '*', was hardcoded in
+ app_dial, which doesn't make any sense from a user perspective
+ since they may expect it to be something different. This patch
+ allows features to be detected from outside of the bridge, but
+ not operated on. In this case, the disconnect feature can be
+ detected before briding and handled outside of features.c.
+ (closes issue #11583) Reported by: sobomax Patches:
+ patch-apps__app_dial.c uploaded by sobomax (license 359)
+ 11583.latest-patch uploaded by murf (license 17)
+ detect_disconnect.diff uploaded by dvossel (license 671) Tested
+ by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/
+ ........ ................
+
+2009-03-19 16:09 +0000 [r183121] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 183117 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r183117 | mmichelson | 2009-03-19 11:07:54 -0500 (Thu, 19 Mar
+ 2009) | 20 lines Merged revisions 183115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar
+ 2009) | 14 lines Fix an issue where cancelled outgoing SIP calls
+ would erroneously report the device as "in use." A user was
+ having an issue where if an outgoing SIP call was canceled, the
+ SIP device would remain in use if we had not received any
+ response to the initial INVITE we sent out. The SIP device would
+ remain in use until the autocongestion timer was exhausted. I
+ tracked down the cause of this to be the section of code I am
+ removing here. I asked several people what the purpose of this
+ code was meant to be, but no one could give me any sort of answer
+ as to why this was here. The person who was having this issue has
+ been using this patch for several months and it has stopped the
+ problems they have had. AST-196 ........ ................
+
+2009-03-19 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.1.0-rc3
+
+2009-03-19 15:43 +0000 [r183067-183110] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 183108 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r183108 |
+ file | 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) | 11 lines
+ Improve our triggering of a T38 switchover internally when
+ triggered by a received reinvite. Previously we reached across
+ the channel bridge to get the other party's SIP dialog structure
+ in order to trigger an outgoing reinvite. This is extremely
+ dangerous to do and only works if bridged to another SIP channel.
+ This patch changes this to use the T38 control frame method of
+ requesting a switchover. This change also causes the SIP channel
+ driver to propogate back whether the switchover worked or not
+ instead of blindly accepting the incoming T38 reinvite. Review:
+ http://reviewboard.digium.com/r/200/ ........
+
+ * main/channel.c, /: Merged revisions 183057 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r183057 |
+ file | 2009-03-18 19:22:56 -0300 (Wed, 18 Mar 2009) | 6 lines Fix
+ an issue where a T38 control frame would get dropped. If two
+ channels were bridged together using a generic bridge the T38
+ control frame would get passed up instead of being indicated on
+ the other channel. ........
+
+2009-03-18 21:19 +0000 [r183030] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/h323/ast_h323.cxx: Merged revisions 183028 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r183028 | jpeeler | 2009-03-18 16:18:27 -0500 (Wed, 18
+ Mar 2009) | 4 lines Add some code removed by mistake from commit
+ 182722 that works around a file descriptor leak in versions of
+ PWLib prior to 1.12.0. ........
+
+2009-03-18 14:32 +0000 [r182946] Russell Bryant <russell@digium.com>
+
+ * main/poll.c, main/io.c, main/channel.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,
+ include/asterisk/io.h, main/utils.c, include/asterisk/channel.h:
+ Merged revisions 182847 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009)
+ | 52 lines Merged revisions 182810 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009)
+ | 44 lines Fix cases where the internal poll() was not being used
+ when it needed to be. We have seen a number of problems caused by
+ poll() not working properly on Mac OSX. If you search around,
+ you'll find a number of references to using select() instead of
+ poll() to work around these issues. In Asterisk, we've had poll.c
+ which implements poll() using select() internally. However, we
+ were still getting reports of problems. vadim investigated a bit
+ and realized that at least on his system, even though we were
+ compiling in poll.o, the system poll() was still being used. So,
+ the primary purpose of this patch is to ensure that we're using
+ the internal poll() when we want it to be used. The changes are:
+ 1) Remove logic for when internal poll should be used from the
+ Makefile. Instead, put it in the configure script. The logic in
+ the configure script is the same as it was in the Makefile.
+ Ideally, we would have a functionality test for the problem, but
+ that's not actually possible, since we would have to be able to
+ run an application on the _target_ system to test poll()
+ behavior. 2) Always include poll.o in the build, but it will be
+ empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll()
+ throughout the source tree to ast_poll(). I feel that it is good
+ practice to give the API call a new name when we are changing its
+ behavior and not using the system version directly in all cases.
+ So, normally, ast_poll() is just redefined to poll(). On systems
+ where AST_POLL_COMPAT is defined, ast_poll() is redefined to
+ ast_internal_poll(). 4) Change poll() in main/poll.c to be
+ ast_internal_poll(). It's worth noting that any code that still
+ uses poll() directly will work fine (if they worked fine before).
+ So, for example, out of tree modules that are using poll() will
+ not stop working or anything. However, for modules to work
+ properly on Mac OSX, ast_poll() needs to be used. (closes issue
+ #13404) Reported by: agalbraith Tested by: russell, vadim
+ http://reviewboard.digium.com/r/198/ ........ ................
+
+2009-03-17 20:52 +0000 [r182724] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/h323/chan_h323.h, 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: Merged revisions
+ 182722 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r182722 |
+ jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines
+ Allow H.323 Plus library to be used in addition to the OpenH323
+ library Chan_h323 can now be compiled against both the previously
+ supported versions of OpenH323 as well as the current H.323 Plus
+ (version 1.20.2). The configure script has been modified to look
+ in the default install location of h323 to hopefully help avoid
+ using the environment variables OPENH323DIR and PWLIBDIR. Also,
+ the CLI command "h323 show version" has been added which
+ indicates which version of h323 is in use. (closes issue #11261)
+ Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch
+ uploaded by jthurman (license 614) ........
+
+2009-03-17 15:31 +0000 [r182570] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 182553 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r182553 |
+ russell | 2009-03-17 10:22:12 -0500 (Tue, 17 Mar 2009) | 5 lines
+ Tweak the handling of the frame list inside of ast_answer(). This
+ does not change any behavior, but moves the frames from the local
+ frame list back to the channel read queue using an O(n) algorithm
+ instead of O(n^2). ........
+
+2009-03-17 15:00 +0000 [r182527-182533] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/channel.c, /: Merged revisions 182530 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r182530 |
+ kpfleming | 2009-03-17 09:59:33 -0500 (Tue, 17 Mar 2009) | 2
+ lines correct logic flaw in ast_answer() changes in r182525
+ ........
+
+ * main/channel.c, /, main/features.c, include/asterisk/channel.h:
+ Merged revisions 182525 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r182525 |
+ kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11
+ lines Improve behavior of ast_answer() to not lose incoming
+ frames ast_answer(), when supplied a delay before returning to
+ the caller, use ast_safe_sleep() to implement the delay.
+ Unfortunately during this time any incoming frames are discarded,
+ which is problematic for T.38 re-INVITES and other sorts of
+ channel operations. When a delay is not passed to ast_answer(),
+ it still delays for up to 500 milliseconds, waiting for media to
+ arrive. Again, though, it discards any control frames, or
+ non-voice media frames. This patch rectifies this situation, by
+ storing all incoming frames during the delay period on a list,
+ and then requeuing them onto the channel before returning to the
+ caller. http://reviewboard.digium.com/r/196/ ........
+
+2009-03-17 05:54 +0000 [r182452] Tilghman Lesher <tlesher@digium.com>
+
+ * main/db.c, /: Merged revisions 182450 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r182450 | tilghman | 2009-03-17 00:51:54 -0500 (Tue, 17 Mar 2009)
+ | 14 lines Merged revisions 182449 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009)
+ | 7 lines Fix race in astdb The underlying db1 implementation
+ does not fully isolate the pages retrieved from astdb, so the
+ lock protecting accesses needs to be extended until the copy from
+ the shared memory structure is done. (closes issue #14682)
+ Reported by: makoto ........ ................
+
+2009-03-16 17:53 +0000 [r182284] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 182282 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r182282 | dvossel | 2009-03-16 12:49:58 -0500
+ (Mon, 16 Mar 2009) | 13 lines Merged revisions 182281 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009)
+ | 7 lines Randomize IAX2 encryption padding The 16-32 byte random
+ padding at the beginning of an encrypted IAX2 frame turns out to
+ not be all that random at all. This patch calls ast_random to
+ fill the padding buffer with random data. The padding is
+ randomized at the beginning of every encrypted call and for every
+ encrypted retransmit frame. Review:
+ http://reviewboard.digium.com/r/193/ ........ ................
+
+2009-03-16 17:38 +0000 [r182280] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /, funcs/func_env.c: Merged revisions
+ 182211,182278 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r182211 | tilghman | 2009-03-16 10:50:55 -0500 (Mon, 16 Mar 2009)
+ | 14 lines Merged revisions 182208 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r182208 | tilghman | 2009-03-16 10:39:15 -0500 (Mon, 16 Mar 2009)
+ | 7 lines Fixup glare detection, to fix a memory leak of a local
+ pvt structure. (closes issue #14656) Reported by: caspy Patches:
+ 20090313__bug14656__2.diff.txt uploaded by tilghman (license 14)
+ Tested by: caspy ........ ................ r182278 | tilghman |
+ 2009-03-16 12:33:38 -0500 (Mon, 16 Mar 2009) | 7 lines Fix an
+ off-by-one error in the FILE() function, and extend FILE()'s
+ length parameter to work like variable substitution. Previously,
+ FILE() returned one less character than specified, due to the
+ terminating NULL. Both the offset and length parameters now
+ behave identically to the way variable substitution offsets and
+ lengths also work. (closes issue #14670) Reported by: BMC
+ ................
+
+2009-03-16 14:00 +0000 [r182173] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c, /: Merged revisions 182171 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r182171 |
+ file | 2009-03-16 10:58:24 -0300 (Mon, 16 Mar 2009) | 7 lines Fix
+ a memory leak in the ast_answer / __ast_answer API call. For a
+ channel that is not yet answered this API call will wait until a
+ voice frame is received on the channel before returning. It does
+ this by waiting for frames on the channel and reading them in.
+ The frames read in were not freed when they should have been.
+ ........
+
+2009-03-13 21:27 +0000 [r182068-182123] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 182121 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r182121 |
+ mmichelson | 2009-03-13 16:26:20 -0500 (Fri, 13 Mar 2009) | 6
+ lines Change faulty comparison used when announcing average hold
+ minutes and seconds (closes issue #14227) Reported by: caspy
+ ........
+
+ * /, main/features.c: Merged revisions 182029 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r182029 | mmichelson | 2009-03-13 12:26:43 -0500 (Fri, 13 Mar
+ 2009) | 41 lines Merged revisions 181990 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar
+ 2009) | 35 lines Check the DYNAMIC_FEATURES of both the chan and
+ peer when interpreting DTMF. Dynamic features defined in the
+ applicationmap section of features.conf allow one to specify
+ whether the caller, callee, or both have the ability to use the
+ feature. The documentation in the features.conf.sample file could
+ be interpreted to mean that one only needs to set the
+ DYNAMIC_FEATURES channel variable on the calling channel in order
+ to allow for the callee to be able to use the features which he
+ should have permission to use. However, the DYNAMIC_FEATURES
+ variable would only be read from the channel of the participant
+ that pressed the DTMF sequence to activate the feature. The
+ result of this was that the callee was unable to use dynamic
+ features unless the dialplan writer had taken measures to be sure
+ that the DYNAMIC_FEATURES variable was set on the callee's
+ channel. This commit changes the behavior of
+ ast_feature_interpret to concatenate the values of
+ DYNAMIC_FEATURES from both parties involved in the bridge. The
+ features themselves determine who has permission to use them, so
+ there is no reason to believe that one side of the bridge could
+ gain the ability to perform an action that they should not have
+ the ability to perform. Kevin Fleming pointed out on the
+ asterisk-users list that the typical way that this was worked
+ around in the past was by setting _DYNAMIC_FEATURES on the
+ calling channel so that the value would be inherited by the
+ called channel. While this works, the documentation alone is not
+ enough to figure out why this is necessary for the callee to be
+ able to use dynamic features. In this particular case, changing
+ the code to match the documentation is safe, easy, and will
+ generally make things easier for people for future installations.
+ This bug was originally reported on the asterisk-users list by
+ David Ruggles. (closes issue #14657) Reported by: mmichelson
+ Patches: 14657.patch uploaded by mmichelson (license 60) ........
+ ................
+
+2009-03-13 17:29 +0000 [r182042] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 182022 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r182022 |
+ file | 2009-03-13 14:25:09 -0300 (Fri, 13 Mar 2009) | 7 lines Fix
+ an issue with requesting a T38 reinvite before the call is
+ answered. The code responsible for sending the T38 reinvite did
+ not check if an INVITE was already being handled. This caused
+ things to get confused and the call to fail. The code now defers
+ sending the T38 reinvite until the current INVITE is done being
+ handled. (issue AST-191) ........
+
+2009-03-13 16:58 +0000 [r181987] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 181985 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r181985 |
+ kpfleming | 2009-03-13 11:55:38 -0500 (Fri, 13 Mar 2009) | 1 line
+ improve a bit of suboptimal code ........
+
+2009-03-12 21:45 +0000 [r181771-181849] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 181846 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r181846 |
+ mmichelson | 2009-03-12 16:43:51 -0500 (Thu, 12 Mar 2009) | 3
+ lines Run the macro on the queue member's channel when he
+ answers, not the caller's channel. ........
+
+ * /, channels/chan_sip.c: Merged revisions 181769 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r181769 | mmichelson | 2009-03-12 13:30:58 -0500 (Thu, 12 Mar
+ 2009) | 28 lines Merged revisions 181768 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar
+ 2009) | 22 lines Properly send a 487 on an INVITE we have not
+ responded to if we receive a BYE. If we receive an INVITE from an
+ endpoint and then later receive a BYE from that same endpoint
+ before we have sent a final response for the INVITE, then we need
+ to respond to the INVITE with a 487. There was logic in the code
+ prior to this commit which seemed to exist solely to handle this
+ situation, but there was one condition in an if statement which
+ was incorrect. The only way we would send a 487 was if the
+ sip_pvt had no owner channel. This made no sense since we created
+ the owner channel when we received the INVITE, meaning that the
+ majority of the time we would never send the 487. The 487 being
+ sent should not rely on whether we have created a channel. Its
+ delivery should be dependent on the current state of the initial
+ INVITE transaction. With this commit, that logic is now correctly
+ in place. (closes issue #14149) Reported by: legranjl Patches:
+ 14149.patch uploaded by mmichelson (license 60) Tested by:
+ legranjl ........ ................
+
+2009-03-12 18:07 +0000 [r181733] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/translate.c: Merged revisions 181731 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r181731 |
+ tilghman | 2009-03-12 12:32:13 -0500 (Thu, 12 Mar 2009) | 9 lines
+ Adjust translation table column widths based upon the translation
+ times. Previously, only 5 columns were displayed, and if a
+ translation time exceeded 99,999 useconds, it would be displayed
+ as 0, instead of its actual time. (closes issue #14532) Reported
+ by: pj Patches: 20090311__bug14532.diff.txt uploaded by tilghman
+ (license 14) Tested by: pj ........
+
+2009-03-12 16:58 +0000 [r181614-181667] Joshua Colp <jcolp@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 181665 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r181665 | file | 2009-03-12 13:56:58 -0300 (Thu,
+ 12 Mar 2009) | 9 lines Merged revisions 181664 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181664 | file | 2009-03-12 13:56:20 -0300 (Thu, 12 Mar 2009) | 2
+ lines Fix incorrect usage of strncasecmp... I really meant to use
+ strcasecmp. ........ ................
+
+ * /, res/res_musiconhold.c: Merged revisions 181661 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r181661 | file | 2009-03-12 13:53:52 -0300 (Thu,
+ 12 Mar 2009) | 19 lines Merged revisions 181659-181660 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8
+ lines Fix another scenario where depending on configuration the
+ stream would not get read. For custom commands we don't know
+ whether the audio is coming from a stream or not so we are going
+ to have to read the data despite no channels. (closes issue
+ #14416) Reported by: caspy ........ r181660 | file | 2009-03-12
+ 13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines Fix logic flaw in
+ previous commit. ........ ................
+
+ * /, res/res_musiconhold.c: Merged revisions 181656 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r181656 | file | 2009-03-12 13:32:20 -0300 (Thu,
+ 12 Mar 2009) | 17 lines Merged revisions 181655 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar 2009) |
+ 10 lines Fix issue with streaming MOH failing if nobody is
+ listening. When a music class is setup to actually provide music
+ on hold from a stream we need to constantly read audio from it
+ since it will constantly be providing audio. This is now done
+ despite there being no channels listening to it. (closes issue
+ #14416) Reported by: caspy ........ ................
+
+ * apps/app_dial.c, /: Merged revisions 181612 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r181612 |
+ file | 2009-03-12 10:24:12 -0300 (Thu, 12 Mar 2009) | 5 lines Fix
+ crash when sleep and retries argument was not given to RetryDial
+ application. (closes issue #14647) Reported by: sherpya ........
+
+2009-03-12 01:05 +0000 [r181544] Richard Mudgett <rmudgett@digium.com>
+
+ * /, build_tools/make_version: Merged revisions 181542 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r181542 | rmudgett | 2009-03-11 20:00:29 -0500 (Wed, 11 Mar 2009)
+ | 1 line Use the correct branch integrated property when
+ generating the version string ........
+
+2009-03-11 23:21 +0000 [r181521] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, configs/sip.conf.sample: Merged revisions 181499 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk Provide
+ correct hint to debug SIP trouble in the default config (closes
+ issue #14646) Reported by: strk
+
+2009-03-11 22:27 +0000 [r181474] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 181465 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r181465 |
+ russell | 2009-03-11 17:25:57 -0500 (Wed, 11 Mar 2009) | 2 lines
+ Make handling of the BRIDGE_PLAY_SOUND variable thread-safe.
+ ........
+
+2009-03-11 22:23 +0000 [r181457] Jason Parker <jparker@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 181444 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r181444 | qwell | 2009-03-11 17:20:13 -0500
+ (Wed, 11 Mar 2009) | 11 lines Merged revisions 181436 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181436 | qwell | 2009-03-11 17:18:42 -0500 (Wed, 11 Mar 2009) |
+ 4 lines Allow prefix to set localstatedir (when used and
+ different from the default). This is similar to the /etc change
+ that was made for the non-FreeBSD case. ........ ................
+
+2009-03-11 22:16 +0000 [r181426-181430] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 181428 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r181428 |
+ russell | 2009-03-11 17:14:55 -0500 (Wed, 11 Mar 2009) | 2 lines
+ Make handling of the BRIDGEPVTCALLID variable thread-safe.
+ ........
+
+ * main/channel.c, /: Merged revisions 181424 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r181424 | russell | 2009-03-11 16:49:29 -0500 (Wed, 11 Mar 2009)
+ | 17 lines Merged revisions 181423 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009)
+ | 9 lines Make code that updates BRIDGEPEER variable thread-safe.
+ It is not safe to read the name field of an ast_channel without
+ the channel locked. This patch fixes some places in channel.c
+ where this was being done, and lead to crashes related to
+ masquerades. (closes issue #14623) Reported by: guillecabeza
+ ........ ................
+
+2009-03-11 17:40 +0000 [r181373] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, channels/iax2-parser.h, /: Merged revisions
+ 181371 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r181371 | dvossel | 2009-03-11 12:34:57 -0500 (Wed, 11 Mar 2009)
+ | 17 lines Merged revisions 181340 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009)
+ | 11 lines encrypted IAX2 during packet loss causes decryption to
+ fail on retransmitted frames If an iax channel is encrypted, and
+ a retransmit frame is sent, that packet's iseqno is updated while
+ it is encrypted. This causes the entire frame to be corrupted.
+ When the corrupted frame is sent, the other side decrypts it and
+ sends a VNAK back because the decrypted frame doesn't make any
+ sense. When we get the VNAK, we look through the sent queue and
+ send the same corrupted frame causing a loop. To fix this,
+ encrypted frames requiring retransmission are decrypted, updated,
+ then re-encrypted. Since key-rotation may change the key held by
+ the pvt struct, the keys used for encryption/decryption are held
+ within the iax_frame to guarantee they remain correct. (closes
+ issue #14607) Reported by: stevenla Tested by: dvossel Review:
+ http://reviewboard.digium.com/r/192/ ........ ................
+
+2009-03-11 17:29 +0000 [r181298-181359] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 181345 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r181345 | file | 2009-03-11 14:26:40 -0300 (Wed, 11 Mar 2009) |
+ 21 lines Merged revisions 181328 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) |
+ 14 lines Fix issue where an attended transfer could not be
+ completed under a rare scenario. When completing an attended
+ transfer chan_sip does a check to make sure the extension in the
+ URI portion of the Refer-To header is a local valid extension. We
+ don't actually need to check this since we know for sure the
+ other channel is already up and talking to the extension. Some
+ devices do not put the extension in the Refer-To header either,
+ which can cause the extension check to fail. We now no longer do
+ this check if it is an attended transfer. (closes issue #14628)
+ Reported by: sverre Patches: 14628.diff uploaded by file (license
+ 11) ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 181296 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r181296 | file | 2009-03-11 13:40:48 -0300 (Wed, 11 Mar 2009) |
+ 16 lines Merged revisions 181295 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9
+ lines Fix a problem with inband DTMF detection on outgoing SIP
+ calls when dtmfmode=auto. When dtmfmode was set to auto the
+ inband DTMF detector was not setup on outgoing SIP calls. This
+ caused inband DTMF detection to fail. The inband DTMF detector is
+ now setup for both dtmfmode inband and auto. (closes issue
+ #13713) Reported by: makoto ........ ................
+
+2009-03-11 15:54 +0000 [r181199-181283] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/h323/ast_h323.cxx: add missing header file
+
+ * pbx/pbx_config.c, utils/Makefile, include/asterisk/utils.h,
+ include/asterisk/astmm.h, /, channels/chan_sip.c,
+ channels/h323/ast_h323.cxx, main/features.c, utils/extconf.c:
+ Merged revisions 181135 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r181135 |
+ jpeeler | 2009-03-10 23:06:44 -0500 (Tue, 10 Mar 2009) | 20 lines
+ Fix malloc debug macros to work properly with h323. The main
+ problem here was that cstdlib was undefining free thereby causing
+ the proper debug macros to not be used. ast_h323.cxx has been
+ changed to call ast_free instead to avoid the issue. A few other
+ issues were addressed: - There were a few instances of functions
+ improperly passing ast_free instead of ast_free_ptr. - Some clean
+ up was done to avoid the debug macros intentionally being
+ redefined. (copied below from Kevin's commit, appreciate the
+ help) - disable astmm.h from doing anything when STANDALONE is
+ defined, which is used by the tools in the utils/ directory that
+ use parts of Asterisk header files in hackish ways; also ensure
+ that utils/extconf.c and utils/conf2ael.c are compiled with
+ STANDALONE defined. (closes issue #13593) Reported by: pj
+ ........
+
+2009-03-11 01:04 +0000 [r181035] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 181032-181033 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r181032 | mmichelson | 2009-03-10 19:46:47 -0500
+ (Tue, 10 Mar 2009) | 19 lines Merged revisions 181029,181031 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar
+ 2009) | 9 lines Fix incorrect tag checking on transfers when
+ pedantic=yes is enabled. (closes issue #14611) Reported by:
+ klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt
+ uploaded by klaus3000 (license 65) Tested by: klaus3000 ........
+ r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar
+ 2009) | 3 lines Remove unused variables. ........
+ ................ r181033 | mmichelson | 2009-03-10 19:49:00 -0500
+ (Tue, 10 Mar 2009) | 3 lines Add missing comment that quotes RFC
+ 3891 ................
+
+2009-03-10 22:07 +0000 [r180947] Jason Parker <jparker@digium.com>
+
+ * /, configure, configure.ac, autoconf/ast_prog_sed.m4,
+ autoconf/ast_check_gnu_make.m4: Merged revisions 180944 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r180944 | qwell | 2009-03-10 17:03:41 -0500
+ (Tue, 10 Mar 2009) | 9 lines Merged revisions 180941 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r180941 | qwell | 2009-03-10 17:02:18 -0500 (Tue, 10 Mar
+ 2009) | 1 line Make things happier when using autoconf 2.62+
+ ........ ................
+
+2009-03-10 14:42 +0000 [r180802] Joshua Colp <jcolp@digium.com>
+
+ * main/manager.c, /: Merged revisions 180800 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r180800 |
+ file | 2009-03-10 11:40:38 -0300 (Tue, 10 Mar 2009) | 5 lines
+ Reset the thread local string buffer when handling the UserEvent
+ action. (closes issue #14593) Reported by: JimDickenson ........
+
+2009-03-09 21:22 +0000 [r180740] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/heap.h, include/asterisk/http.h,
+ include/asterisk/logger.h, main/tcptls.c,
+ include/asterisk/res_odbc.h, include/asterisk/doxyref.h,
+ include/asterisk/event.h, include/asterisk/audiohook.h,
+ include/asterisk/dsp.h, include/asterisk/lock.h,
+ include/asterisk/udptl.h, include/asterisk/dnsmgr.h,
+ include/asterisk/utils.h, include/asterisk/devicestate.h, /,
+ include/asterisk/taskprocessor.h, include/asterisk/astobj2.h,
+ include/asterisk/channel.h, include/asterisk/tcptls.h,
+ include/asterisk/manager.h, main/enum.c,
+ include/asterisk/callerid.h, include/asterisk/app.h,
+ include/asterisk/linkedlists.h, include/asterisk/sched.h,
+ include/asterisk/datastore.h, include/asterisk/timing.h,
+ include/asterisk/dlinkedlists.h, include/asterisk/pbx.h,
+ include/asterisk/enum.h, include/asterisk/config.h,
+ include/asterisk/rtp.h, include/asterisk/extconf.h,
+ main/devicestate.c: Merged revisions 180719 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r180719 |
+ jpeeler | 2009-03-09 15:58:17 -0500 (Mon, 09 Mar 2009) | 16 lines
+ Add Doxygen documentation for API changes from 1.6.0 to 1.6.1
+ Copied from my review board description: This is a continuation
+ of the API changes documentation started for describing changes
+ between releases. Most of the API changes were pretty simple
+ needing only to be brought to attention via the new "Asterisk API
+ Changes" list. However, if you see anything that needs further
+ explanation feel free to supplement what is there. The current
+ method of documenting is to add (in the header file): \version
+ <ver number> <description of changes> and then to add the
+ function to the change list in doxyref.h on the AstAPIChanges
+ page. I also made sure all the functions that were newly added
+ were tagged with \since 1.6.1. I think this is a good habit to
+ start both for the historical aspect as well as for the future
+ ability to easily add a "New Asterisk API" page. Review:
+ http://reviewboard.digium.com/r/190/ ........
+
+2009-03-06 18:26 +0000 [r180585] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 180579 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r180579 | mmichelson | 2009-03-06 12:25:44 -0600
+ (Fri, 06 Mar 2009) | 9 lines Merged revisions 180567 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r180567 | mmichelson | 2009-03-06 12:23:09 -0600 (Fri,
+ 06 Mar 2009) | 2 lines Make compilation succeed in dev-mode when
+ IMAP storage is enabled. ........ ................
+
+2009-03-06 17:35 +0000 [r180537] David Vossel <dvossel@digium.com>
+
+ * main/enum.c, /: Merged revisions 180534 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r180534 | dvossel | 2009-03-06 11:26:38 -0600 (Fri, 06 Mar 2009)
+ | 15 lines Merged revisions 180532 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r180532 | dvossel | 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009)
+ | 9 lines 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 23:28 +0000 [r180425-180467] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 180465 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r180465 | mmichelson | 2009-03-05 17:26:58 -0600
+ (Thu, 05 Mar 2009) | 22 lines Merged revisions 180464 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r180464 | mmichelson | 2009-03-05 17:26:11 -0600 (Thu, 05 Mar
+ 2009) | 16 lines [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: Merged
+ revisions 180383 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r180383 | mmichelson | 2009-03-05 13:14:14 -0600 (Thu, 05 Mar
+ 2009) | 31 lines Merged revisions 180380 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar
+ 2009) | 25 lines 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 18:40 +0000 [r180378] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/frame.h, main/rtp.c, main/frame.c, /: Merged
+ revisions 180373 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r180373 | kpfleming | 2009-03-05 12:29:38 -0600 (Thu, 05 Mar
+ 2009) | 15 lines Merged revisions 180372 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar
+ 2009) | 9 lines 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 Leif Madsen <lmadsen@digium.com>
+
+ * Released Asterisk 1.6.1.0-rc2
+
+2009-03-04 21:09 +0000 [r180263] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 180261 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r180261 |
+ russell | 2009-03-04 15:01:05 -0600 (Wed, 04 Mar 2009) | 54 lines
+ Resolve object matching issues related to the removal of the
+ sip_user object. Previously, chan_sip had both sip_peer and
+ sip_user objects in memory. A patch went in to remove sip_user to
+ simplify the code, since everything could be done with just
+ sip_peer. This patch resolves some regressions found that were
+ introduced by those changes. This code comes from
+ svn/asterisk/team/group/sip-object-matching/. Here is a list of
+ the changes that have been made: 1) When doing a match by name
+ with the find_peer() function, make it much easier to specify
+ which objects should be matched by having a parameter that
+ specifies exactly which object types should be considered. Also,
+ update find_by_name() to handle this parameter. Finally, update
+ all code to use the new option values. 2) When looking up an
+ object for an outbound request by name, consider peers only.
+ (create_addr()) 3) Only match peers on an incoming registration
+ request. 4) When doing authentication (except for SUBSCRIBE),
+ look up users by name, instead of all objects by name. 5) When
+ doing authentication (except for SUBSCRIBE), after looking for a
+ user by name, look for a peer by IP address, instead of all
+ objects by IP address. 6) When handling the SIP qualify CLI
+ command or manager action, look for a peer by name, instead of
+ any object by name. 7) When handling the SIP unregister CLI
+ command, look for a peer by name, instead of any object by name.
+ 9) In sip_do_debug_peer(), search for a peer by name, instead of
+ any object by name. 9) When handling the SIPPEER() dialplan
+ function, search for a peer by name, instead of any object by
+ name. 10) In the following session timer related functions,
+ st_get_se(), st_get_refresher(), and st_get_mode(), when looking
+ for an object for a given sip_pvt using pvt->peername, look for a
+ peer by name, instead of any object by name. 11) Fix build_peer()
+ to properly handle the case where separate type=peer and
+ type=user entries were specified in sip.conf. (closes issue
+ #14505) Reported by: lmadsen Review:
+ http://reviewboard.digium.com/r/172/ ........
+
+2009-03-04 19:27 +0000 [r180122-180197] Joshua Colp <jcolp@digium.com>
+
+ * /, main/callerid.c: Merged revisions 180195 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r180195 | file | 2009-03-04 15:24:59 -0400 (Wed, 04 Mar 2009) |
+ 11 lines Merged revisions 180194 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r180194 | file | 2009-03-04 15:22:50 -0400 (Wed, 04 Mar 2009) | 4
+ lines 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) ........ ................
+
+ * apps/app_dial.c, /: Merged revisions 180120 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r180120 |
+ file | 2009-03-04 10:39:28 -0400 (Wed, 04 Mar 2009) | 7 lines
+ Remove duplicate 'k' and 'K' Dial options. (closes issue #14601)
+ Reported by: alecdavis Patches: app_dial.optionk.diff.txt
+ uploaded by alecdavis (license 585) ........
+
+2009-03-03 23:39 +0000 [r180080] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, include/asterisk/app.h, apps/app_read.c, /,
+ main/app.c: Merged revisions 180032 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r180032 |
+ dvossel | 2009-03-03 17:21:18 -0600 (Tue, 03 Mar 2009) | 14 lines
+ app_read does not break from prompt loop with user terminated
+ empty string In app.c, ast_app_getdata is called to stream the
+ prompts and receive DTMF input. If ast_app_getdata() receives an
+ empty string caused by the user inputing the end of string
+ character, in this case '#', it should break from the prompt loop
+ and return to app_read, but instead it cycles through all the
+ prompts. I've added a return value for this special case in
+ ast_readstring() which uses an enum I've delcared in apps.h. This
+ enum is now used as a return value for ast_app_getdata(). (closes
+ issue #14279) Reported by: Marquis Patches: fix_app_read.patch
+ uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded
+ by dvossel (license 671) Tested by: Marquis, dvossel Review:
+ http://reviewboard.digium.com/r/177/ ........
+
+2009-03-03 23:31 +0000 [r180077] 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: Merged revisions 179973 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179973 | murf | 2009-03-03 15:12:02 -0700 (Tue, 03 Mar 2009) |
+ 33 lines Merged revisions 179807 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 I had some
+ work to do to port these changes to trunk; the check_expr stuff
+ hasn't been updated here for quite some time, it appears. I added
+ some more tests to the check_expr2 suite. I had to play around
+ with the makefile a bit, etc. I added STANDALONE2 #ifdefs to
+ ast_expr2.y so as not to conflict structure with aelparse.
+ ........ r179807 | murf | 2009-03-03 11:11:34 -0700 (Tue, 03 Mar
+ 2009) | 19 lines 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 22:49 +0000 [r179939-180009] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /, configs/queues.conf.sample: Merged revisions
+ 180007 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r180007 | mmichelson | 2009-03-03 16:49:07 -0600 (Tue, 03 Mar
+ 2009) | 22 lines Merged revisions 180006 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar
+ 2009) | 17 lines 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 ........ ................
+
+ * doc/timing.txt (added), /, res/res_timing_dahdi.c: Merged
+ revisions 179937 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r179937 |
+ mmichelson | 2009-03-03 14:59:16 -0600 (Tue, 03 Mar 2009) | 20
+ lines Add documentation for timing modules used in Asterisk This
+ document specifies the timing modules available in Asterisk
+ beginning with Asterisk 1.6.1. The document goes into detail
+ about the differences between each and gives a general overview
+ of what timing is used for in Asterisk. There is also a section
+ which can be used to help customize your setup or to troubleshoot
+ timing issues you may have. I also added messages to the DAHDI
+ timing test used in res_timing_dahdi.c that points to this new
+ documentation if people experience problems. Big thanks to all
+ who contributed comments on this. (closes issue #14490) Reported
+ by: mmichelson Patches: timing.txt uploaded by mmichelson
+ (license 60) Review: http://reviewboard.digium.com/r/164/
+ ........
+
+2009-03-03 20:09 +0000 [r179905] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_directed_pickup.c: Merged revisions 179903 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r179903 | bmd | 2009-03-03 14:02:20 -0600 (Tue, 03 Mar
+ 2009) | 1 line fix a leaked channel lock (and future deadlock)
+ when we try to pick up our own channel ........
+
+2009-03-03 18:30 +0000 [r179843] Joshua Colp <jcolp@digium.com>
+
+ * /, main/features.c: Merged revisions 179841 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179841 | file | 2009-03-03 14:28:46 -0400 (Tue, 03 Mar 2009) |
+ 16 lines Merged revisions 179840 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9
+ lines 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 16:48 +0000 [r179744] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 179742 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179742 | russell | 2009-03-03 10:47:28 -0600 (Tue, 03 Mar 2009)
+ | 14 lines Merged revisions 179741 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009)
+ | 6 lines 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 14:41 +0000 [r179674] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c, /: Merged revisions 179672 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179672 | file | 2009-03-03 10:40:04 -0400 (Tue, 03 Mar 2009) |
+ 10 lines Merged revisions 179671 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3
+ lines 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 13:56 +0000 [r179611] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 179609 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179609 | russell | 2009-03-03 07:54:41 -0600 (Tue, 03 Mar 2009)
+ | 17 lines Merged revisions 179608 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009)
+ | 9 lines 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. ........ ................
+
+2009-03-03 00:04 +0000 [r179539] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 179537 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179537 | jpeeler | 2009-03-02 18:01:51 -0600 (Mon, 02 Mar 2009)
+ | 21 lines Merged revisions 179536 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009)
+ | 15 lines 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 23:39 +0000 [r179535] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 179533 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179533 | russell | 2009-03-02 17:36:38 -0600 (Mon, 02 Mar 2009)
+ | 48 lines Merged revisions 179532 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009)
+ | 40 lines 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 23:12 +0000 [r179471] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/app.c: Merged revisions 179469 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179469 | tilghman | 2009-03-02 17:10:18 -0600 (Mon, 02 Mar 2009)
+ | 17 lines Merged revisions 179468 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009)
+ | 10 lines 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 23:04 +0000 [r179464] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 179462 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179462 | russell | 2009-03-02 17:00:30 -0600 (Mon, 02 Mar 2009)
+ | 16 lines Merged revisions 179461 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009)
+ | 8 lines 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 20:18 +0000 [r179407] Jason Parker <jparker@digium.com>
+
+ * /, main/editline/configure, main/editline/np/unvis.c,
+ main/editline/sys.h, main/editline/configure.in: Merged revisions
+ 179396 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179396 | qwell | 2009-03-02 14:16:51 -0600 (Mon, 02 Mar 2009) |
+ 9 lines Merged revisions 179395 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179395 | qwell | 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) |
+ 1 line Remove several silly warnings in editline. One about a
+ broken preprocessor directive, and another about strlcpy/strlcat.
+ (closes issue #14264) Reported by: dimas ........
+ ................
+
+2009-03-02 17:19 +0000 [r179362] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 179361 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r179361 | tilghman | 2009-03-02 11:18:48 -0600 (Mon, 02 Mar 2009)
+ | 2 lines Backport 1.6.0 fix to trunk (failsafe if db is not
+ loaded) ........
+
+2009-03-02 14:14 +0000 [r179293] Joshua Colp <jcolp@digium.com>
+
+ * /, main/audiohook.c: Merged revisions 179291 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r179291 |
+ file | 2009-03-02 10:13:45 -0400 (Mon, 02 Mar 2009) | 7 lines Fix
+ issue where changing the volume of both directions of audio did
+ not work. (closes issue #14574) Reported by: KNK Patches:
+ audiohook_volume_fix.diff uploaded by KNK (license 545) ........
+
+2009-03-01 23:28 +0000 [r179221-179256] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_speech_utils.c, /: Merged revisions 179254 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r179254 | mmichelson | 2009-03-01 17:25:23 -0600 (Sun, 01 Mar
+ 2009) | 5 lines Swap reversed timevals. This was pointed out by
+ ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ! ........
+
+ * /, channels/chan_sip.c: Merged revisions 179219 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r179219 |
+ mmichelson | 2009-03-01 15:45:08 -0600 (Sun, 01 Mar 2009) | 18
+ lines Properly free memory and remove scheduler entries when a
+ transmission failure occurs. Previously, only the "data" field of
+ the sip_pkt created during __sip_reliable_xmit was freed when
+ XMIT_ERROR was returned by __sip_xmit. When retrans_pkt was
+ called, this inevitably resulted in the reading and writing of
+ freed memory. XMIT_ERROR is a condition meaning that we don't
+ want to attempt resending the packet at all. The proper action to
+ take is to remove the scheduler entry we just created, free the
+ packet's data as well as the packet itself, and unlink it from
+ the list of packets on the sip_pvt structure. (closes issue
+ #14455) Reported by: Nick_Lewis Patches: 14455.patch uploaded by
+ mmichelson (license 60) Tested by: Nick_Lewis ........
+
+2009-02-27 21:48 +0000 [r179166] Russell Bryant <russell@digium.com>
+
+ * configs/ais.conf.sample, res/res_ais.c, /,
+ doc/distributed_devstate.txt: Merged revisions 179164 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r179164 | russell | 2009-02-27 15:47:18 -0600 (Fri, 27
+ Feb 2009) | 2 lines Mark res_ais as experimental, as the binary
+ event format is subject to change. ........
+
+2009-02-27 21:34 +0000 [r179163] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 179161 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r179161 | tilghman | 2009-02-27 15:32:13 -0600 (Fri, 27 Feb 2009)
+ | 3 lines If config file is blank, don't load module. (Closes
+ issue #14563) ........
+
+2009-02-27 21:25 +0000 [r179160] Russell Bryant <russell@digium.com>
+
+ * /, UPGRADE.txt: Merged revisions 179154 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r179154 |
+ russell | 2009-02-27 15:23:12 -0600 (Fri, 27 Feb 2009) | 2 lines
+ Add a note about the ordering of entries in sip.conf in 1.6.1.
+ ........
+
+2009-02-27 19:06 +0000 [r179059] Jason Parker <jparker@digium.com>
+
+ * /, doc/tex/channelvariables.tex: Merged revisions 179057 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r179057 | qwell | 2009-02-27 13:04:57 -0600 (Fri, 27 Feb
+ 2009) | 8 lines 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) ........
+
+2009-02-27 03:56 +0000 [r178988] Steve Murphy <murf@digium.com>
+
+ * configs/features.conf.sample, /, main/features.c: Merged
+ revisions 178986 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r178986 | murf | 2009-02-26 20:45:58 -0700 (Thu, 26 Feb 2009) |
+ 26 lines Merged revisions 178956 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 In this
+ case, it's just a matter of reducing the default timeouts from
+ 2000 to 1000 msec, as the max def feature digit timeout is no
+ longer halved. ........ r178956 | murf | 2009-02-26 14:27:32
+ -0700 (Thu, 26 Feb 2009) | 18 lines 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 17:50 +0000 [r178875] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 178871 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r178871 | dvossel | 2009-02-26 11:46:12 -0600 (Thu, 26 Feb 2009)
+ | 6 lines IAX2 prune realtime, minor tweak to last fix A return
+ statement was missing which caused unexpected cli output. issue
+ #14479 ........
+
+2009-02-26 17:38 +0000 [r178869] Steve Murphy <murf@digium.com>
+
+ * /, main/features.c: Merged revisions 178828 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r178828 | murf | 2009-02-26 10:22:11 -0700 (Thu, 26 Feb 2009) |
+ 34 lines Merged revisions 178804 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) |
+ 28 lines 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 ........ ................
+
+2009-02-26 16:44 +0000 [r178803] Joshua Colp <jcolp@digium.com>
+
+ * /, main/file.c: Merged revisions 178801 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r178801 |
+ file | 2009-02-26 12:42:36 -0400 (Thu, 26 Feb 2009) | 5 lines Fix
+ an issue where the timer for file playback would not be stopped
+ if DAHDI was not installed. (closes issue #14541) Reported by:
+ grant ........
+
+2009-02-26 16:07 +0000 [r178769] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 178767 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r178767 | dvossel | 2009-02-26 09:50:22 -0600 (Thu, 26 Feb 2009)
+ | 8 lines IAX2 prune realtime fix Iax2 prune realtime had issues.
+ If "iax2 prune realtime all" was called, it would appear like the
+ command was successful, but in reality nothing happened. This is
+ because the reload that was supposed to take place checks the
+ config files, sees no changes, and does nothing. If there had
+ been a change in the the config file, the realtime users would
+ have been marked for deletion and everything would have been
+ fine. 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/ ........
+
+2009-02-25 12:46 +0000 [r178511] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 178509 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r178509 | russell | 2009-02-25 06:45:30 -0600 (Wed, 25 Feb 2009)
+ | 10 lines Merged revisions 178508 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009)
+ | 2 lines Update the copyright year for the main page of the
+ doxygen documentation. ........ ................
+
+2009-02-24 23:28 +0000 [r178383-178448] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 178446 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r178446 | tilghman | 2009-02-24 17:27:23 -0600
+ (Tue, 24 Feb 2009) | 12 lines Merged revisions 178445 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009)
+ | 5 lines 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) ........
+ ................
+
+ * main/asterisk.c, /: Merged revisions 178381 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r178381 |
+ tilghman | 2009-02-24 14:52:44 -0600 (Tue, 24 Feb 2009) | 2 lines
+ Apparently, a void cast doesn't override warn_unused_result.
+ ........
+
+2009-02-24 20:44 +0000 [r178379-178380] Russell Bryant <russell@digium.com>
+
+ * Makefile: revert accidental Makefile change.
+
+ * main/rtp.c, Makefile, /: Merged revisions 178374 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r178374 | russell | 2009-02-24 14:39:57 -0600
+ (Tue, 24 Feb 2009) | 14 lines Merged revisions 178373 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009)
+ | 6 lines Only set dtmfcount on BEGIN, and ensure it gets reset
+ to 0 properly. (issue #14460) Reported by: moliveras Tested by:
+ russell ........ ................
+
+2009-02-24 20:41 +0000 [r178305-178377] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 178375 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r178375 |
+ tilghman | 2009-02-24 14:40:02 -0600 (Tue, 24 Feb 2009) | 2 lines
+ The 3 possible errors with pipe(2) are all impossible in this
+ situation. ........
+
+ * main/asterisk.c, /, utils/astcanary.c: Merged revisions 178342
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r178342 | tilghman | 2009-02-24 14:06:48 -0600 (Tue, 24
+ Feb 2009) | 2 lines Use a SIGPIPE to kill the process, instead of
+ depending upon the astcanary process being inherited by init.
+ ........
+
+ * /, utils/astcanary.c: Merged revisions 178303 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r178303 |
+ tilghman | 2009-02-24 11:51:36 -0600 (Tue, 24 Feb 2009) | 7 lines
+ Cause astcanary to exit if Asterisk exits abnormally and doesn't
+ kill astcanary. Also, add some documentation supporting the use
+ of astcanary. (closes issue #14538) Reported by: KNK Patches:
+ asterisk-1.6.x-astcanary.diff uploaded by KNK (license 545)
+ ........
+
+2009-02-24 15:22 +0000 [r178232] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 178213 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r178213 | file | 2009-02-24 11:18:38 -0400 (Tue, 24 Feb 2009) |
+ 16 lines Merged revisions 178205 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9
+ lines 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 23:22 +0000 [r178172] Russell Bryant <russell@digium.com>
+
+ * main/rtp.c, /: Merged revisions 178142 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r178142 | russell | 2009-02-23 17:11:37 -0600 (Mon, 23 Feb 2009)
+ | 22 lines Merged revisions 178141 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009)
+ | 14 lines 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 ........ ................
+
+2009-02-21 16:04 +0000 [r177945] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 177944 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177944 |
+ tilghman | 2009-02-21 09:59:49 -0600 (Sat, 21 Feb 2009) | 2 lines
+ On update, test against the existence of sipregs. ........
+
+2009-02-21 12:51 +0000 [r177851] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, channels/chan_sip.c: Merged revisions 177849 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177849 |
+ mvanbaak | 2009-02-21 13:22:32 +0100 (Sat, 21 Feb 2009) | 2 lines
+ make chan_sip.c compile on OpenBSD again. ........
+
+2009-02-20 23:05 +0000 [r177789] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 177787 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r177787 | tilghman | 2009-02-20 17:02:35 -0600 (Fri, 20 Feb 2009)
+ | 16 lines Merged revisions 177786 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r177786 | tilghman | 2009-02-20 16:59:52 -0600 (Fri, 20 Feb 2009)
+ | 9 lines 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 ........ ................
+
+2009-02-20 22:27 +0000 [r177785] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 177699 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177699 |
+ dhubbard | 2009-02-20 14:29:00 -0600 (Fri, 20 Feb 2009) | 9 lines
+ Make app_fax compatible with spandsp-0.0.6pre4 Prior to
+ spandsp-0.0.6pre4 the t30_stats_t structure used a
+ pages_transferred integer to indicate the number of pages
+ transferred (so far) during the fax session. The
+ spandsp-0.0.6pre4 release removed the pages_transferred integer
+ and replaced it with two different integers - pages_tx and
+ pages_rx. This revision uses the new integers for
+ spandsp-0.0.6pre4 while maintaining backwards compatibility for
+ previous spandsp releases. ........
+
+2009-02-20 22:15 +0000 [r177760-177764] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/strings.h: Oops, last merge broke 1.6.1 branch
+
+ * apps/app_system.c, include/asterisk/app.h, /, main/app.c: Merged
+ revisions 177664 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177664 |
+ tilghman | 2009-02-20 11:29:51 -0600 (Fri, 20 Feb 2009) | 8 lines
+ Allow semicolons to be escaped, when passing arguments to the
+ System command. (closes issue #14231) Reported by: jcovert
+ Patches: 20090113__bug14231__2.diff.txt uploaded by Corydon76
+ (license 14) corrected_20090113__bug14231__2.diff.txt uploaded by
+ jcovert (license 551) Tested by: jcovert ........
+
+ * include/asterisk/threadstorage.h, /: Merged revisions 177732 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r177732 | tilghman | 2009-02-20 15:25:37 -0600
+ (Fri, 20 Feb 2009) | 10 lines Merged revisions 177701 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r177701 | tilghman | 2009-02-20 15:15:01 -0600 (Fri, 20 Feb 2009)
+ | 3 lines 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 20:34 +0000 [r177700] 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
+ 1.6 only audio codec bits 0-12 are defined, leaving bits 13-14
+ undefined. By default all bits are enabled unless specified
+ otherwise. Since its a 2 byte field and 13-14 are not defined,
+ these bits are never turned off. In trunk, bits 13-14 are
+ defined, which means 1.6 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
+
+2009-02-20 17:28 +0000 [r177663] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 177661 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r177661 | tilghman | 2009-02-20 11:22:19 -0600 (Fri, 20 Feb 2009)
+ | 2 lines Oops, merge broke trunk ........
+
+2009-02-20 00:38 +0000 [r177626] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 177624 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177624 |
+ jpeeler | 2009-02-19 18:35:53 -0600 (Thu, 19 Feb 2009) | 7 lines
+ Set sip_request ast_str data to NULL so ast_str_copy allocates
+ space properly in copy_request (issue #14478) Reported by:
+ erik_dedecker ........
+
+2009-02-20 00:26 +0000 [r177623] Steve Murphy <murf@digium.com>
+
+ * /, main/Makefile, main/ast_expr2f.c: Merged revisions 177595 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r177595 | murf | 2009-02-19 16:56:50 -0700 (Thu,
+ 19 Feb 2009) | 32 lines Merged revisions 177540 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 Trunk was
+ already pretty 8-bit clean; but I'm still removing the --full
+ from the flex command so everything is uniform. ........ r177540
+ | murf | 2009-02-19 15:51:37 -0700 (Thu, 19 Feb 2009) | 21 lines
+ 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 22:35 +0000 [r177539] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 177537 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r177537 | tilghman | 2009-02-19 16:33:00 -0600
+ (Thu, 19 Feb 2009) | 14 lines Merged revisions 177536 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r177536 | tilghman | 2009-02-19 16:26:01 -0600 (Thu, 19 Feb 2009)
+ | 7 lines 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 16:46 +0000 [r177389] Jeff Peeler <jpeeler@digium.com>
+
+ * /, include/asterisk/channel.h: Merged revisions 177387 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r177387 | jpeeler | 2009-02-19 10:45:02 -0600 (Thu, 19
+ Feb 2009) | 3 lines Fix another merge error from 176708 ........
+
+2009-02-19 16:40 +0000 [r177386] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_speech_utils.c, /: Merged revisions 177384 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r177384 | file | 2009-02-19 12:38:41 -0400 (Thu,
+ 19 Feb 2009) | 10 lines Merged revisions 177383 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r177383 | file | 2009-02-19 12:37:25 -0400 (Thu, 19 Feb 2009) | 3
+ lines If we are able to create a speech structure unset the ERROR
+ variable in case it was previously set. (issue #LUMENVOX-13)
+ ........ ................
+
+2009-02-19 15:57 +0000 [r177358] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 177356 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177356 |
+ jpeeler | 2009-02-19 09:56:31 -0600 (Thu, 19 Feb 2009) | 4 lines
+ Fix mismerge from revision 176708 pointed out by Kaloyan Kovachev
+ on the asterisk-dev mailing list. Thanks! ........
+
+2009-02-19 00:17 +0000 [r177294] Steve Murphy <murf@digium.com>
+
+ * /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 177286 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r177286 | murf | 2009-02-18 16:50:57 -0700 (Wed,
+ 18 Feb 2009) | 39 lines Merged revisions 177225 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r177225 | murf | 2009-02-18 15:43:14 -0700 (Wed, 18 Feb 2009) |
+ 34 lines 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 23:15 +0000 [r177230] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/frame.c, /: Merged revisions 177229 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177229 |
+ kpfleming | 2009-02-18 17:09:58 -0600 (Wed, 18 Feb 2009) | 3
+ lines fix two very minor bugs: if anyone ever uses SLINEAR16 as a
+ format in RTP, ensure that the samples are byte-swapped to
+ network order if needed. also, when a smoother is operating on a
+ format that has a sample rate other than 8000 samples per second,
+ use the proper sample rate for computing delivery timestamps.
+ ........
+
+2009-02-18 23:03 +0000 [r177228] David Vossel <dvossel@digium.com>
+
+ * /, main/features.c: Merged revisions 177226 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177226 |
+ dvossel | 2009-02-18 16:51:38 -0600 (Wed, 18 Feb 2009) | 9 lines
+ Locking issue in action_bridge and bridge_exec action_bridge()
+ and bridge_exec() both search for the channels to bridge to, and
+ then immediately drop the lock. Instead, they should hold the
+ lock until the masquerade is complete. This will guarantee the
+ channel remains and prevent any other weirdness from occurring.
+ In action_bridge() some more weirdness comes into play. Both
+ channels are needlessly locked at the same time and perform the
+ exact same logic. It makes sense from a coding organizational
+ standpoint, but could cause a theoretical deadlock so I split the
+ code up. There is an issue associated with this, but since its a
+ rather complicated thing to reproduce I'm not certain this alone
+ will close it. issue# 14296 Review:
+ http://reviewboard.digium.com/r/167/ ........
+
+2009-02-18 20:16 +0000 [r177164] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/h323/chan_h323.h, 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_ptlib.h (added), channels/h323/ast_h323.cxx,
+ configure, channels/h323/compat_h323.h, configure.ac,
+ channels/h323/caps_h323.h, autoconf/ast_prog_sed.m4,
+ channels/h323/ast_h323.h: Merged revisions 177162 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r177162 | jpeeler | 2009-02-18 14:11:57 -0600 (Wed, 18 Feb 2009)
+ | 14 lines 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 19:30 +0000 [r177158] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 177101 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177101 |
+ russell | 2009-02-18 13:12:49 -0600 (Wed, 18 Feb 2009) | 8 lines
+ Re-add 'o' option to MeetMe, reverting rev 62297. Enabling this
+ option by default proved to be a bad idea, as the talker
+ detection is not very reliable. So, make it optional again, and
+ off by default. (issue #13801) Reported by: justdave ........
+
+2009-02-18 19:09 +0000 [r177100] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/config.h: Merged revisions 177098 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r177098 | tilghman | 2009-02-18 13:05:15 -0600
+ (Wed, 18 Feb 2009) | 9 lines Merged revisions 177096 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r177096 | tilghman | 2009-02-18 12:30:38 -0600 (Wed, 18
+ Feb 2009) | 2 lines Document the return value of the update
+ method (as requested on -dev list) ........ ................
+
+2009-02-18 17:26 +0000 [r177037] Doug Bailey <dbailey@digium.com>
+
+ * /, main/utils.c: Merged revisions 177035 via svnmerge 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. ........
+
+2009-02-18 17:14 +0000 [r177007] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 177005 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177005 |
+ file | 2009-02-18 13:11:52 -0400 (Wed, 18 Feb 2009) | 6 lines Fix
+ ordering of output for a ChannelUpdate manager event. (closes
+ issue #14497) Reported by: vinsik Patches:
+ chan_update_fix-chan_sip.c.diff uploaded by vinsik (license 623)
+ ........
+
+2009-02-18 16:20 +0000 [r176962] Doug Bailey <dbailey@digium.com>
+
+ * /, main/utils.c: Merged revisions 176948 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r176948 |
+ dbailey | 2009-02-18 10:09:12 -0600 (Wed, 18 Feb 2009) | 2 lines
+ Need to take into account the \0 terminator of the old string to
+ determine the amount available. ........
+
+2009-02-18 15:59 +0000 [r176946] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c, /: Merged revisions 176943 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r176943 |
+ murf | 2009-02-18 08:35:26 -0700 (Wed, 18 Feb 2009) | 45 lines
+ This patch fixes merge_contexts_and_delete so it does not
+ deadlock when hints are present. Reason: when I re-engineered the
+ merge_and_delete func to reduce its lock time, I failed to notice
+ that the functions it calls still also do locking as before. This
+ leads to deadlocks on dialplan reloads, when there are actually
+ living, subscribed hints registered in the system. While the
+ reporter come across this problem while using AEL, I might note
+ that these deadlocks should also happen if extensions.conf were
+ used. Here I added these routines to pbx.c:
+ ast_add_extension_nolock add_pri_lockopt
+ ast_add_extension2_lockopt find_context add_hint_nolock All of
+ the above routines are static and restricted to be used only
+ within pbx.c, and more specifically within the
+ merge_contexts_and_delete routine. They are pretty much the same
+ as their counterparts except they don't lock contexts or hints.
+ Most of them now do the real work of their name-alike, with
+ optional locking via extra arguments, and are called by their
+ name-alike. The goal was to have the original functions so they
+ would behave exactly as before. Both PJ and I tested these fixes,
+ and the deadlocking problem is no longer encountered. (closes
+ issue #14357) Reported by: pj Patches: 14357.diff uploaded by
+ murf (license 17) Tested by: pj, murf ........
+
+2009-02-18 06:15 +0000 [r176903-176906] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/heap.h, /: Merged revisions 176904 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r176904 | russell | 2009-02-18 00:14:47 -0600 (Wed, 18 Feb 2009)
+ | 2 lines Add example code for a heap traversal. ........
+
+ * main/pbx.c, /: Merged revisions 176901 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r176901 |
+ russell | 2009-02-18 00:00:40 -0600 (Wed, 18 Feb 2009) | 9 lines
+ Fix a number of incorrect uses of strncpy(). The big problem here
+ is that the 3rd argument provided in these uses of strncpy() did
+ not reserve a byte for the null terminator, leaving the potential
+ for writing one byte past the end of the buffer. Aside from this,
+ there were coding guidelines violations with regards to spacing,
+ as well as hard coded lengths being used instead of sizeof().
+ ........
+
+2009-02-18 00:23 +0000 [r176809] Shaun Ruffell <sruffell@digium.com>
+
+ * /, codecs/codec_dahdi.c: Merged revisions 176760 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r176760 | sruffell | 2009-02-17 16:28:41 -0600 (Tue, 17 Feb 2009)
+ | 10 lines 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-trunk-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 22:21 +0000 [r176731] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 176705 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r176705 |
+ dhubbard | 2009-02-17 15:59:38 -0600 (Tue, 17 Feb 2009) | 11
+ lines create a UDPTL structure in create_addr_from_peer() if it
+ does not already exist for T38 This is required to create a UDPTL
+ structure in create_addr_from_peer() to handle the scenario where
+ 't38pt_udptl=yes' is not defined in the [general] section of
+ sip.conf but is defined the peer's context. I tested this patch
+ by enabling t38pt_udptl in the [general] section on one system
+ and only enabling t38pt_udptl in a peer's context on the system
+ sending a fax. Without the patch, the sending system will fail to
+ initiate T38 negotiation with the warning message, "No way to add
+ SDP without an UDPTL structure". When this patch is applied the
+ sending side will successfully initiate T38 negotiation. ........
+
+2009-02-17 22:15 +0000 [r176711] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /, main/features.c, include/asterisk/channel.h:
+ Merged revisions 176708 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r176708 | jpeeler | 2009-02-17 16:08:00 -0600 (Tue, 17 Feb 2009)
+ | 23 lines Merged revisions 176701 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009)
+ | 17 lines 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 21:41 +0000 [r176699] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/frame.h, /: Merged revisions 176697 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r176697 | mmichelson | 2009-02-17 15:40:09 -0600 (Tue, 17 Feb
+ 2009) | 3 lines Clear up documentation of AST_FRIENDLY_OFFSET in
+ frame.h ........
+
+2009-02-17 21:24 +0000 [r176675] Russell Bryant <russell@digium.com>
+
+ * main/timing.c, main/channel.c, /, res/res_timing_pthread.c,
+ res/res_timing_dahdi.c, include/asterisk/timing.h: Merged
+ revisions 176666 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r176666 |
+ russell | 2009-02-17 15:22:40 -0600 (Tue, 17 Feb 2009) | 16 lines
+ Update the timing API to have better support for multiple timing
+ interfaces. 1) Add module use count handling so that timing
+ modules can be unloaded. 2) Implement unload_module() functions
+ for the timing interface modules. 3) Allow multiple timing
+ modules to be loaded, and use the one with the highest priority
+ value. 4) Report which timing module is being use in the "timing
+ test" CLI command. (closes issue #14489) Reported by: russell
+ Review: http://reviewboard.digium.com/r/162/ ........
+
+2009-02-17 21:16 +0000 [r176644] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_odbc.c, channels/chan_local.c, /: Merged revisions
+ 176592,176642 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r176592 |
+ tilghman | 2009-02-17 12:49:20 -0600 (Tue, 17 Feb 2009) | 4 lines
+ Add assertions in the quest to track down a refcount leak.
+ (closes issue #14485) Reported by: davevg ........ r176642 |
+ tilghman | 2009-02-17 15:14:18 -0600 (Tue, 17 Feb 2009) | 8 lines
+ 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 ........
+
+2009-02-17 20:57 +0000 [r176559-176637] Russell Bryant <russell@digium.com>
+
+ * tests/test_heap.c (added), /: Merged revisions 176635 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r176635 | russell | 2009-02-17 14:56:26 -0600 (Tue, 17
+ Feb 2009) | 4 lines Add a test module for the heap
+ implementation. Review: http://reviewboard.digium.com/r/160/
+ ........
+
+ * include/asterisk/heap.h (added), /, main/Makefile, main/heap.c
+ (added): Merged revisions 176632 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r176632 |
+ russell | 2009-02-17 14:51:10 -0600 (Tue, 17 Feb 2009) | 8 lines
+ Add an implementation of the heap data structure. A heap is a
+ convenient data structure for implementing a priority queue. Code
+ from svn/asterisk/team/russell/heap/. Review:
+ http://reviewboard.digium.com/r/160/ ........
+
+ * apps/app_queue.c, main/pbx.c, /: Merged revisions 176557 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r176557 | russell | 2009-02-17 11:33:38 -0600 (Tue, 17
+ Feb 2009) | 12 lines Fix a race condition that caused device
+ states to become incorrect for hints. The problem here is that
+ the hint processing code was subscribed to the wrong event type.
+ So, it started processing state for a hint too soon, before the
+ device state cache had been updated. Also, fix a similar bug in
+ app_queue, as it was also subscribed to the wrong event type.
+ (closes issue #14461) Reported by: alecdavis ........
+
+2009-02-17 14:48 +0000 [r176461-176503] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 176501 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r176501 |
+ tilghman | 2009-02-17 08:39:36 -0600 (Tue, 17 Feb 2009) | 3 lines
+ In this version, we can combine the queries, because we support
+ dropping nonexistent columns. ........
+
+ * /, channels/chan_sip.c: Merged revisions 176459 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r176459 | tilghman | 2009-02-16 19:58:39 -0600 (Mon, 16 Feb 2009)
+ | 17 lines Merged revisions 176426 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r176426 | tilghman | 2009-02-16 18:49:22 -0600 (Mon, 16 Feb 2009)
+ | 10 lines 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 23:57 +0000 [r176362] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 176355 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r176355 | dvossel | 2009-02-16 17:33:55 -0600
+ (Mon, 16 Feb 2009) | 13 lines Merged revisions 176354 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r176354 | dvossel | 2009-02-16 17:30:52 -0600 (Mon, 16 Feb 2009)
+ | 8 lines 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 23:17 +0000 [r176321] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_skinny.c: Merged revisions 176320 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r176320 | tilghman | 2009-02-16 17:14:08 -0600 (Mon, 16 Feb 2009)
+ | 7 lines Use the correct list macros for deleting an item from
+ the middle of a list. (issue #13777) Reported by: pj Patches:
+ 20090203__bug13777.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: pj ........
+
+2009-02-16 22:00 +0000 [r176259] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/stringfields.h, /, main/utils.c: Merged
+ revisions 176255 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r176255 | kpfleming | 2009-02-16 15:45:54 -0600 (Mon, 16 Feb
+ 2009) | 13 lines Merged revisions 176216 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r176216 | kpfleming | 2009-02-16 15:10:38 -0600 (Mon, 16 Feb
+ 2009) | 3 lines 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. ........ r176254 | kpfleming | 2009-02-16 15:41:46
+ -0600 (Mon, 16 Feb 2009) | 3 lines 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 21:50 +0000 [r176257] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 176253 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r176253 | mmichelson | 2009-02-16 15:40:40 -0600 (Mon, 16 Feb
+ 2009) | 24 lines Merged revisions 176249,176252 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r176249 | mmichelson | 2009-02-16 15:34:27 -0600 (Mon, 16 Feb
+ 2009) | 14 lines 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 ........ r176252 |
+ mmichelson | 2009-02-16 15:39:21 -0600 (Mon, 16 Feb 2009) | 3
+ lines Remove unused variable and make dev-mode compilation happy
+ ........ ................
+
+2009-02-16 21:36 +0000 [r176251] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 176248 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r176248 | dvossel | 2009-02-16 15:30:17 -0600
+ (Mon, 16 Feb 2009) | 11 lines Merged revisions 175597 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r175597 | dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13
+ Feb 2009) | 4 lines Fixed iax2 key rotation backwards
+ compatibility Turns key rotation back on by default. Added bit
+ into encryption IE to indicate whether or not key rotation is
+ supported or not. If it is not supported then it is not enabled,
+ which insures backwards compatibility. This eliminates the need
+ for the keyrotate option in iax.conf, so it has been removed.
+ ........ ................
+
+2009-02-16 18:38 +0000 [r176176] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/logger.c: Merged revisions 176174 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r176174 |
+ mmichelson | 2009-02-16 12:25:57 -0600 (Mon, 16 Feb 2009) | 11
+ lines Assist proper thread synchronization when stopping the
+ logger thread. I was finding that on my dev box, occasionally
+ attempting to "stop now" in trunk would cause Asterisk to hang. I
+ traced this to the fact that the logger thread was waiting on a
+ condition which had already been signalled. The logger thread
+ also need to be sure to check the value of the
+ close_logger_thread variable. The close_logger_thread variable is
+ only checked when the list of logmessages is empty. This allows
+ for the logger thread to print and free any pending messages
+ before exiting. ........
+
+2009-02-16 17:10 +0000 [r176102] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_features.c (removed): Merged revisions 176100
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r176100 | russell | 2009-02-16 11:09:24 -0600 (Mon, 16
+ Feb 2009) | 4 lines Remove chan_features. Review:
+ http://reviewboard.digium.com/r/161/ ........
+
+2009-02-16 17:07 +0000 [r176099] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/func_odbc.conf.sample: Eliminate mention of a variable
+ which exists only in trunk. (Thanks, jsmith)
+
+2009-02-16 15:38 +0000 [r176032] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 176030 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r176030 | file | 2009-02-16 11:36:19 -0400 (Mon, 16 Feb 2009) |
+ 16 lines Merged revisions 176029 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9
+ lines 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)
+ ........ ................
+
+2009-02-16 09:42 +0000 [r176023] Michiel van Baak <michiel@vanbaak.info>
+
+ * include/asterisk/manager.h, doc/unistim.txt,
+ channels/chan_unistim.c, /, channels/chan_sip.c: Merged revisions
+ 175952 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r175952 | mvanbaak | 2009-02-16 01:26:59 +0100 (Mon, 16 Feb 2009)
+ | 10 lines Merged revisions 175921 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009)
+ | 3 lines fix mis-spelling of the word registered. Reported by
+ De_Mon on #asterisk-dev. ........ ................
+
+2009-02-15 21:28 +0000 [r175831-175890] Russell Bryant <russell@digium.com>
+
+ * main/sched.c, /, include/asterisk/sched.h: Merged revisions
+ 175882 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r175882 |
+ russell | 2009-02-15 15:27:33 -0600 (Sun, 15 Feb 2009) | 2 lines
+ Make ast_sched_report() and ast_sched_dump() thread safe.
+ ........
+
+ * main/sched.c, /, channels/chan_sip.c, include/asterisk/sched.h:
+ Merged revisions 175829 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r175829 |
+ russell | 2009-02-15 14:56:27 -0600 (Sun, 15 Feb 2009) | 14 lines
+ Fix a number of problems with ast_sched_report(). 1) It had
+ numerous coding guidelines violations with regards to formatting.
+ 2) It allocated memory using ast_calloc() that was never freed.
+ 3) It didn't check for failure from the allocation. 4) It used
+ sprintf() and strcat() to build the result, doing zero checking
+ to prevent writing past the end of the provided buffer. The
+ function also lacks API documentation, but that has not been
+ addressed in this commit. ........
+
+2009-02-13 20:48 +0000 [r175662] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, configs/iax.conf.sample, channels/iax2.h:
+ Merged revisions 175597 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r175597 |
+ dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines
+ Fixed iax2 key rotation backwards compatibility Turns key
+ rotation back on by default. Added bit into encryption IE to
+ indicate whether or not key rotation is supported or not. If it
+ is not supported then it is not enabled, which insures backwards
+ compatibility. This eliminates the need for the keyrotate option
+ in iax.conf, so it has been removed. ........
+
+2009-02-13 19:52 +0000 [r175593] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 175591 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r175591 | mmichelson | 2009-02-13 13:49:38 -0600
+ (Fri, 13 Feb 2009) | 22 lines Merged revisions 175590 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175590 | mmichelson | 2009-02-13 13:47:48 -0600 (Fri, 13 Feb
+ 2009) | 16 lines 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) ........ ................
+
+2009-02-13 16:44 +0000 [r175551] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_record.c: Merged revisions 175549 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r175549 |
+ file | 2009-02-13 12:41:15 -0400 (Fri, 13 Feb 2009) | 4 lines Add
+ an option to keep the recorded file upon hangup. (closes issue
+ #14341) Reported by: fnordian ........
+
+2009-02-12 21:41 +0000 [r175370] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 175368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r175368 |
+ russell | 2009-02-12 15:41:01 -0600 (Thu, 12 Feb 2009) | 2 lines
+ Remove useless string copy, and make sscanf safe again ........
+
+2009-02-12 21:27 +0000 [r175342] Tilghman Lesher <tlesher@digium.com>
+
+ * main/udptl.c, /: Merged revisions 175334 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r175334 | tilghman | 2009-02-12 15:25:14 -0600 (Thu, 12 Feb 2009)
+ | 16 lines Merged revisions 175311 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009)
+ | 9 lines 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 20:51 +0000 [r175300] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 175298 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r175298 | jpeeler | 2009-02-12 14:48:56 -0600 (Thu, 12 Feb 2009)
+ | 15 lines Merged revisions 175294 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009)
+ | 9 lines 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 ........ ................
+
+2009-02-12 20:48 +0000 [r175257-175297] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 175295 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r175295 |
+ russell | 2009-02-12 14:45:47 -0600 (Thu, 12 Feb 2009) | 2 lines
+ Avoid using ast_strdupa() in a loop. ........
+
+ * build_tools/cflags.xml, /: Merged revisions 175255 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r175255 | russell | 2009-02-12 13:11:08 -0600 (Thu, 12 Feb 2009)
+ | 4 lines Don't enable something by default that has a dependency
+ on something _not_ enabled by default. menuselect was not happy
+ with this. ........
+
+2009-02-12 18:50 +0000 [r175251] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 175250 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r175250 | kpfleming | 2009-02-12 12:48:52 -0600 (Thu, 12 Feb
+ 2009) | 1 line correct warning message to not refer specifically
+ to DAHDI ........
+
+2009-02-12 18:01 +0000 [r175190] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 175188 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r175188 | jpeeler | 2009-02-12 12:00:11 -0600 (Thu, 12 Feb 2009)
+ | 12 lines Merged revisions 175187 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009)
+ | 6 lines 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 17:09 +0000 [r175130] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 175127 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r175127 | dvossel | 2009-02-12 11:07:17 -0600 (Thu, 12 Feb 2009)
+ | 4 lines Setting key rotation to be off by default Key rotation
+ breaks compatibility between (trunk/1.6.1) and (1.2/1.4/1.6.0).
+ As a follow up to this, I am investigating possible ways to allow
+ key rotation to be on by default and not affect the other
+ branches, but for now it must be turned off. ........
+
+2009-02-12 17:08 +0000 [r175129] Russell Bryant <russell@digium.com>
+
+ * main/rtp.c, /: Merged revisions 175125 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r175125 | russell | 2009-02-12 10:57:25 -0600 (Thu, 12 Feb 2009)
+ | 35 lines Merged revisions 175124 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009)
+ | 27 lines 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 16:35 +0000 [r174947-175123] Mark Michelson <mmichelson@digium.com>
+
+ * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions
+ 175121 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r175121 |
+ mmichelson | 2009-02-12 10:28:06 -0600 (Thu, 12 Feb 2009) | 11
+ lines Make lock information for ao2_trylock be more useful and
+ gnarly Core show locks information involving an ao2_trylock did
+ not show the function that called ao2_trylock, but would instead
+ show ao2_trylock as the source of the lock. This is not useful
+ when trying to debug locking issues. One bizarre note is that
+ this logic is already in 1.4 but somehow did not get merged to
+ trunk or the 1.6.X branches. ........
+
+ * apps/app_queue.c, /: Merged revisions 174951 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174951 |
+ mmichelson | 2009-02-11 17:12:57 -0600 (Wed, 11 Feb 2009) | 3
+ lines Fix a bit of odd logic for announcing position. Sync with
+ 1.6.0's logic ........
+
+ * apps/app_queue.c, /: Merged revisions 174948 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174948 |
+ mmichelson | 2009-02-11 17:03:08 -0600 (Wed, 11 Feb 2009) | 20
+ lines Fix odd "thank you" sound playing behavior in app_queue.c
+ If someone has configured the queue to play an position or
+ holdtime announcement, then it is odd and potentially unexpected
+ to hear a "Thank you for your patience" sound when no position or
+ holdtime was actually announced. This fixes the announcement so
+ that the "thanks" sound is only played in the case that a
+ position or holdtime was actually announced. There is a way that
+ the "thank you" sound can be played without a position or
+ holdtime, and that is to set announce-frequency to a value but
+ keep announce-position and announce-holdtime both turned off.
+ (closes issue #14227) Reported by: caspy Patches: 14227_v3.patch
+ uploaded by putnopvut (license 60) Tested by: caspy ........
+
+ * apps/app_dial.c, main/channel.c, main/pbx.c, /,
+ apps/app_dictate.c, apps/app_waitforsilence.c,
+ include/asterisk/channel.h: Merged revisions 174945 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r174945 | mmichelson | 2009-02-11 16:41:01 -0600 (Wed, 11 Feb
+ 2009) | 29 lines Fix 'd' option for app_dial and add new option
+ to Answer application The 'd' option would not work for channel
+ types which use RTP to transport DTMF digits. The only way to
+ allow for this to work was to answer the channel if we saw that
+ this option was enabled. I realized that this may cause issues
+ with CDRs, specifically with giving false dispositions and answer
+ times. I therefore modified ast_answer to take another parameter
+ which would tell if the CDR should be marked answered. I also
+ extended this to the Answer application so that the channel may
+ be answered but not CDRified if desired. I also modified
+ app_dictate and app_waitforsilence to only answer the channel if
+ it is not already up, to help not allow for faulty CDR answer
+ times. All of these changes are going into Asterisk trunk. For
+ 1.6.0 and 1.6.1, however, all the changes except for the change
+ to the Answer application will go in since we do not introduce
+ new features into stable branches (closes issue #14164) Reported
+ by: DennisD Patches: 14164.patch uploaded by putnopvut (license
+ 60) Tested by: putnopvut Review:
+ http://reviewboard.digium.com/r/145 ........
+
+2009-02-11 14:46 +0000 [r174846] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c, /: Merged revisions 174844 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174844 |
+ file | 2009-02-11 10:44:47 -0400 (Wed, 11 Feb 2009) | 10 lines
+ Tell the device state core a change happened when a channel is
+ freed but not a specific state. We need to do this because while
+ we know that the freeing of the channel may cause something to
+ become not in use we do not know this for sure. There may be
+ another channel that is still up which would cause it to be in
+ use. (closes issue #13238) Reported by: kowalma Patches:
+ 20090121__bug13238.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: alecdavis ........
+
+2009-02-10 23:21 +0000 [r174769-174823] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 174805 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174805 |
+ mmichelson | 2009-02-10 17:17:03 -0600 (Tue, 10 Feb 2009) | 11
+ lines Fix potential for stack overflows in app_chanspy.c When
+ using the 'g' or 'e' options, the stack allocations that were
+ used could cause a stack overflow if a spyer stayed on the line
+ long enough without actually successfully spying on anyone. The
+ problem has been corrected by using static buffers and copying
+ the contents of the appropriate strings into them instead of
+ using functions like alloca or ast_strdupa ........
+
+ * main/manager.c, /: Merged revisions 174764 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174764 |
+ mmichelson | 2009-02-10 15:45:14 -0600 (Tue, 10 Feb 2009) | 21
+ lines Fix an fd leak that would occur in HTTP AMI sessions The
+ explanation behind this fix is a bit complicated, and I've
+ already typed it up in the code as a huge comment inside of
+ manager.c, so I'll give the abridged version here. We needed a
+ way to separate action-specific data from session-specific data.
+ Unfortunately, the only way to maintain API compatibility and to
+ not have to change every single manager action was to rename the
+ current mansession structure and wrap it inside a new mansession
+ structure which actually contains action- specific data. (closes
+ issue #14364) Reported by: awk Patches: 14364_better.patch
+ uploaded by putnopvut (license 60) Tested by: putnopvut Review:
+ http://reviewboard.digium.com/r/148/ ........
+
+2009-02-10 20:17 +0000 [r174714] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 174710 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174710 |
+ file | 2009-02-10 16:15:43 -0400 (Tue, 10 Feb 2009) | 4 lines
+ Only decrease inringing count if above zero. (issue #13238)
+ Reported by: kowalma ........
+
+2009-02-10 18:18 +0000 [r174590] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/jitterbuf.c: Merged revisions 174584 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r174584 | mnicholson | 2009-02-10 12:16:31 -0600 (Tue, 10 Feb
+ 2009) | 25 lines Merged revisions 174583 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb
+ 2009) | 18 lines 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 17:49 +0000 [r174545-174582] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 174580 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174580 |
+ file | 2009-02-10 13:48:29 -0400 (Tue, 10 Feb 2009) | 4 lines Set
+ the type for the peer structure to be a peer as the default.
+ (closes issue #14447) Reported by: triccyx ........
+
+ * /, channels/chan_sip.c: Merged revisions 174543 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174543 |
+ file | 2009-02-10 11:37:07 -0400 (Tue, 10 Feb 2009) | 6 lines
+ Make the logic for inuse and inringing manipluation match that of
+ 1.4. The old broken logic would reset the values back to 0 during
+ certain scenarios causing the wrong state to be reported. (closes
+ issue #14399) Reported by: caspy (issue #13238) Reported by:
+ kowalma ........
+
+2009-02-10 07:07 +0000 [r174471-174504] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_stack.c, apps/app_voicemail.c, /: Merged revisions
+ 174503 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174503 |
+ tilghman | 2009-02-10 01:06:29 -0600 (Tue, 10 Feb 2009) | 2 lines
+ Fix0ring build ........
+
+ * apps/app_stack.c, /: Merged revisions 174470 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174470 |
+ tilghman | 2009-02-09 23:39:33 -0600 (Mon, 09 Feb 2009) | 2 lines
+ Remove the usage of the KeepAlive app, as it no longer exists.
+ ........
+
+2009-02-10 05:13 +0000 [r174428-174440] Steve Murphy <murf@digium.com>
+
+ * apps/app_osplookup.c: This patch corrects warnings which seem to
+ appear only on 64-bit compilers, gcc-4.3.2.
+
+ * apps/app_rpt.c: One final fix in the 1.6.1 release only; some
+ variables the compiler worries "may not be initialized".
+
+ * apps/app_rpt.c, /: Merged revisions 174435 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174435 |
+ murf | 2009-02-09 21:49:02 -0700 (Mon, 09 Feb 2009) | 8 lines
+ This patch removes the use of AST_PBX_KEEPALIVE from app_rpt.c.
+ (closes issue #14435) Reported by: D_McNaul ........
+
+ * apps/app_rpt.c, /: Merged revisions 174432 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174432 |
+ murf | 2009-02-09 21:36:22 -0700 (Mon, 09 Feb 2009) | 3 lines
+ More intptr_t work. ........
+
+ * apps/app_rpt.c, /: Merged revisions 174370 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r174370 | murf | 2009-02-09 19:45:56 -0700 (Mon, 09 Feb 2009) |
+ 10 lines Merged revisions 174369 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174369 | murf | 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) | 5
+ lines This patch solves some compiler complaints in both 32 and
+ 64-bit environments. ........ ................
+
+2009-02-09 17:47 +0000 [r174330] David Vossel <dvossel@digium.com>
+
+ * /, apps/app_externalivr.c: Merged revisions 174325 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r174325 | dvossel | 2009-02-09 11:26:02 -0600 (Mon, 09 Feb 2009)
+ | 9 lines Fixes issue with hangups not being sent and external
+ process never terminating. The ignore_hangup, run_dead, and
+ noanswer flags were never initilized to zero causing hangups to
+ never be issued. If the external script expects to be notified of
+ a hangup and never receives one, it runs indefinitely. (closes
+ issue #14251) Reported by: chris-mac Tested by: dvossel ........
+
+2009-02-09 17:30 +0000 [r174326-174329] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 174327 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174327 |
+ mmichelson | 2009-02-09 11:27:32 -0600 (Mon, 09 Feb 2009) | 3
+ lines Fix something I messed up in the merge I just did ........
+
+ * /, channels/chan_sip.c: Merged revisions 174301 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r174301 | mmichelson | 2009-02-09 11:20:55 -0600 (Mon, 09 Feb
+ 2009) | 20 lines Merged revisions 174282 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb
+ 2009) | 12 lines 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 14:50 +0000 [r174221] Joshua Colp <jcolp@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 174219 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r174219 | file | 2009-02-09 10:49:24 -0400 (Mon,
+ 09 Feb 2009) | 11 lines Merged revisions 174218 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb 2009) | 4
+ lines 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 16:18 +0000 [r174154] Russell Bryant <russell@digium.com>
+
+ * /, res/snmp/agent.c: Merged revisions 174149 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r174149 | russell | 2009-02-07 10:16:50 -0600 (Sat, 07 Feb 2009)
+ | 10 lines Merged revisions 174148 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009)
+ | 2 lines Fix a race condition that could cause a crash. ........
+ ................
+
+2009-02-07 00:09 +0000 [r174086] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 174084 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r174084 | dhubbard | 2009-02-06 17:51:56 -0600 (Fri, 06 Feb 2009)
+ | 13 lines Merged revisions 174082 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009)
+ | 5 lines 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 19:30 +0000 [r173994-174043] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 174041 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r174041 | file | 2009-02-06 15:28:53 -0400 (Fri, 06 Feb 2009) | 4
+ lines Don't subscribe to a mailbox on pseudo channels. It is
+ futile. This solves an issue where duplicated pseudo channels
+ would cause a crash because the first one would unsubscribe and
+ the next one would also try to unsubscribe the same subscription.
+ (closes issue #14322) Reported by: amessina ........
+
+ * /, channels/chan_sip.c: Merged revisions 173974 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r173974 | file | 2009-02-06 13:18:35 -0400 (Fri, 06 Feb 2009) |
+ 15 lines Merged revisions 173967-173968 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4
+ lines 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 ........ r173968 | file
+ | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines Remove a
+ debug message I put in by accident. ........ ................
+
+2009-02-06 17:05 +0000 [r173964-173966] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 173952 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r173952 | mnicholson | 2009-02-06 10:28:19 -0600 (Fri, 06 Feb
+ 2009) | 14 lines Merged revisions 173917 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb
+ 2009) | 7 lines Limit the addition of the Contact header in SIP
+ responses according to various SIP RFCs. (closes issue #13602)
+ Reported by: hjourdain Tested by: mnicholson ........
+ ................
+
+ * main/ast_expr2.c, /, channels/chan_sip.c, main/ast_expr2.h:
+ revert revision 173964
+
+ * main/ast_expr2.c, /, channels/chan_sip.c, main/ast_expr2.h:
+ Merged revisions 173952 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r173952 | mnicholson | 2009-02-06 10:28:19 -0600 (Fri, 06 Feb
+ 2009) | 14 lines Merged revisions 173917 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb
+ 2009) | 7 lines 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 16:01 +0000 [r173904] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_chanspy.c, /, main/audiohook.c: Merged revisions 173902
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r173902 | file | 2009-02-06 11:59:17 -0400 (Fri, 06 Feb
+ 2009) | 4 lines Always detach and destroy the whisper and barge
+ audiohooks. Additionally also allow an audiohook to be detached
+ if it has not been attached. (closes issue #14414) Reported by:
+ bluecrow76 ........
+
+2009-02-06 10:26 +0000 [r173850] Russell Bryant <russell@digium.com>
+
+ * main/manager.c, /: Merged revisions 173848 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r173848 |
+ russell | 2009-02-06 04:25:09 -0600 (Fri, 06 Feb 2009) | 2 lines
+ Resolve a memory leak that would occur on an invalid channel
+ given to Action: Status ........
+
+2009-02-05 23:53 +0000 [r173779] Mark Michelson <mmichelson@digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 173776 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r173776 | mmichelson | 2009-02-05 17:48:48 -0600 (Thu,
+ 05 Feb 2009) | 14 lines Update extensions.conf.sample to be
+ correct. In trunk, the only necessary change pointed out was that
+ the call to ChanIsAvail uses an option that has been removed. For
+ the 1.6.1 branch, however, it appears that the sample file is
+ badly in need of updating since there are |'s used all over the
+ place there. My tentative plan is just to copy trunk's sample