[asterisk-commits] lmadsen: tag 1.6.1.20-rc2 r262985 - /tags/1.6.1.20-rc2/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu May 13 12:35:59 CDT 2010
Author: lmadsen
Date: Thu May 13 12:35:54 2010
New Revision: 262985
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=262985
Log:
Importing files for 1.6.1.20-rc2 release.
Added:
tags/1.6.1.20-rc2/.lastclean (with props)
tags/1.6.1.20-rc2/.version (with props)
tags/1.6.1.20-rc2/ChangeLog (with props)
Added: tags/1.6.1.20-rc2/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.1.20-rc2/.lastclean?view=auto&rev=262985
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Added: tags/1.6.1.20-rc2/ChangeLog
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--- tags/1.6.1.20-rc2/ChangeLog (added)
+++ tags/1.6.1.20-rc2/ChangeLog Thu May 13 12:35:54 2010
@@ -1,0 +1,66231 @@
+2010-05-13 Leif Madsen <lmadsen at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * Asterisk 1.6.1.20-rc1 Released
+
+2010-05-06 14:02 +0000 [r261497] Russell Bryant <russell at 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 at 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 (×”', 06 מ××™ 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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
[... 65556 lines stripped ...]
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