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- Committer: Bazaar Package Importer
- Author(s): Lorenzo De Liso
- Date: 2010-10-15 22:24:34 UTC
- mfrom: (1.2.8 upstream) (8.3.8 sid)
- Revision ID: james.westby@ubuntu.com-20101015222434-iy328q8in3lajzlv
Tags: 1:1.6.2.9-2ubuntu1
* Merge from debian unstable, remaining changes:
- debian/control:
+ Build-depend on hardening-wrapper
+ Change Maintainer
+ Removed Uploaders field.
+ Removed Debian Vcs-Svn entry and replaced with ubuntu-voip Vcs-Bzr,
to reflect divergence in packages.
- debian/rules: Make use of hardening-wrapper
- debian/asterisk.init: chown /dev/dahdi
- debian/backports/hardy: add file
- debian/backports/asterisk.init.hardy: add file
- debian/control:
+ Build-depend on hardening-wrapper
+ Change Maintainer
+ Removed Uploaders field.
+ Removed Debian Vcs-Svn entry and replaced with ubuntu-voip Vcs-Bzr,
to reflect divergence in packages.
- debian/rules: Make use of hardening-wrapper
- debian/asterisk.init: chown /dev/dahdi
- debian/backports/hardy: add file
- debian/backports/asterisk.init.hardy: add file
- files added:
- .pc/man_hyphen
- .pc/man_hyphen/contrib
- .pc/man_hyphen/contrib/scripts
- .pc/man_hyphen/doc
- .pc/parser-mangles-include
- .pc/parser-mangles-include/main
- .pc/rtcp_cli_fix
- .pc/rtcp_cli_fix/main
- .pc/typos
- .pc/typos/apps
- .pc/typos/channels
- .pc/typos/res
- files removed:
- .pc/dahdi_fxs_false_ringing
- .pc/dahdi_fxs_false_ringing/channels
- .pc/fxs_ports_1626
- .pc/fxs_ports_1626/channels
- files modified:
- .pc/allow-tilde-destdir/Makefile
added
removed
ChangeLog
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2010-06-17 Leif Madsen <lmadsen@digium.com>
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* Asterisk 1.6.2.9 Released.
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2010-06-10 Leif Madsen <lmadsen@digium.com>
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* Asterisk 1.6.2.9-rc3 Released.
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2010-06-10 Tilghman Lesher <tlesher@digium.com>
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* Ensure signals are not blocked inside other signal handlers.
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This eliminates the annoying <beep> on the console.
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(closes issue 0017477)
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Reported by: jvandal
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Patches:
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20100610__issue17477.diff.txt uploaded by tilghman (license 14
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2010-06-09 Paul Belanger <paul.belanger@polybeacon.com>
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* Fix Debian init script to not use -c.
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When using the init script as-is currently, it could cause issues on Debian
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such as high CPU usage. This fix has worked for several people so I'm
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implementing the change. We now handle color displays properly.
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(closes issue 0016784)
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Reported by: pabelanger
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Patches:
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20100530__issue16784__2.diff.txt uploaded by tilghman (license 14)
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Tested by: pabelanger, tilghman
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2010-06-07 Leif Madsen <lmadsen@digium.com>
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* Asterisk 1.6.2.9-rc2 Released.
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2010-06-07 Tilghman Lesher <tlesher@digium.com>
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* Fix crash in DTMF detection.
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What I did not originally see in my previous commit was that even
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though the next digit could be detected before the previous was
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considered ended, the detection of the next digit effectively ends
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the detection of the previous. Therefore, the length moves in
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lockstep with the digit, and no separate counter is needed for the
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length alone.
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(closes issue 0017371)
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Reported by: alecdavis
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(closes issue 0017474)
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Reported by: kenner
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2010-06-01 Leif Madsen <lmadsen@digium.com>
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* Asterisk 1.6.2.9-rc1 Released.
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2010-06-01 15:20 +0000 [r266598] Tilghman Lesher <tlesher@digium.com>
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* main/asterisk.c, /: Merged revisions 266592 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ................
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r266592 | tilghman | 2010-06-01 10:18:59 -0500 (Tue, 01 Jun 2010)
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| 18 lines Merged revisions 266585 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
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r266585 | tilghman | 2010-06-01 10:17:46 -0500 (Tue, 01 Jun 2010)
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| 11 lines Prevent CLI prompt from distorting output of lines
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shorter than the prompt. Uses the VT100 method of clearing the
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line from the cursor position to the end of the line: Esc-0K
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(closes issue #17160) Reported by: coolmig Patches:
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20100531__issue17160.diff.txt uploaded by tilghman (license 14)
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Tested by: coolmig ........ ................
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2010-05-31 16:07 +0000 [r266570] Paul Belanger <paul.belanger@polybeacon.com>
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* res/res_agi.c: Fix typo in documentation (closes issue #17395)
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Reported by: pabelanger Patches: res_agi.c.patch uploaded by
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pabelanger (license 224)
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2010-05-30 04:45 +0000 [r266439] Tilghman Lesher <tlesher@digium.com>
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* contrib/init.d/rc.debian.asterisk, /: Merged revisions 266438 via
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svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
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................ r266438 | tilghman | 2010-05-29 23:44:28 -0500
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(Sat, 29 May 2010) | 9 lines Merged revisions 266437 via svnmerge
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from https://origsvn.digium.com/svn/asterisk/branches/1.4
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........ r266437 | tilghman | 2010-05-29 23:43:28 -0500 (Sat, 29
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May 2010) | 2 lines Reverting patch and reopening issue #16784,
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as patch breaks color display. ........ ................
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2010-05-28 20:55 +0000 [r266338] Tilghman Lesher <tlesher@digium.com>
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* main/asterisk.c, /: Merged revisions 266337 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ........ r266337 |
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tilghman | 2010-05-28 15:53:04 -0500 (Fri, 28 May 2010) | 1 line
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Only report swap on platforms which can examine those statistics
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........
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2010-05-28 17:57 +0000 [r266293] David Vossel <dvossel@digium.com>
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* /, channels/chan_sip.c: Merged revisions 266292 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ........ r266292 |
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dvossel | 2010-05-28 12:55:38 -0500 (Fri, 28 May 2010) | 9 lines
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fixes crash when creation of UDPTL fails (closes issue #17264)
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Reported by: falves11 Patches: issue_17264_reviewboard_fix.diff
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uploaded by dvossel (license 671)
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issue_17264_1.6.2_reviewboard_fix.diff uploaded by dvossel
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(license 671) Tested by: falves11 ........
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2010-05-26 21:19 +0000 [r266154] Tilghman Lesher <tlesher@digium.com>
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* utils/extconf.c, main/asterisk.c, /, main/logger.c: Merged
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revisions 266146 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ................
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r266146 | tilghman | 2010-05-26 16:17:46 -0500 (Wed, 26 May 2010)
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| 21 lines Merged revisions 266142 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
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r266142 | tilghman | 2010-05-26 16:11:44 -0500 (Wed, 26 May 2010)
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| 14 lines Use sigaction for signals which should persist past
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the initial trigger, not signal. If you call signal() in a
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Solaris signal handler, instead of just resetting the signal
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handler, it causes the signal to refire, because the signal is
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not marked as handled prior to the signal handler being called.
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This effectively causes Solaris to immediately exceed the
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threadstack in recursive signal handlers and crash. (closes issue
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#17000) Reported by: rmcgilvr Patches:
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20100526__issue17000.diff.txt uploaded by tilghman (license 14)
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Tested by: rmcgilvr ........ ................
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2010-05-26 18:37 +0000 [r266007] David Vossel <dvossel@digium.com>
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* /, channels/chan_sip.c: Merged revisions 266006 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ........ r266006 |
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dvossel | 2010-05-26 13:32:51 -0500 (Wed, 26 May 2010) | 8 lines
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fixes failed SIP Directed pickup resulting in dead channel
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(closes issue #17339) Reported by: one47 Patches:
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sip_magic_pickup2 uploaded by one47 (license 23) Tested by:
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one47, dvossel ........
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2010-05-26 16:31 +0000 [r265895-265959] Tilghman Lesher <tlesher@digium.com>
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* res/res_config_pgsql.c, /: Merged revisions 265923 via svnmerge
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from https://origsvn.digium.com/svn/asterisk/trunk
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................ r265923 | tilghman | 2010-05-26 11:23:28 -0500
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(Wed, 26 May 2010) | 14 lines Merged revisions 265910 via
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svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
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r265910 | tilghman | 2010-05-26 11:21:00 -0500 (Wed, 26 May 2010)
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| 7 lines Not finding rows in the DB does not rise to the level
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of a warning. (closes issue #17062) Reported by: drookie Patches:
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20100525__issue17062.diff.txt uploaded by tilghman (license 14)
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........ ................
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* configs/res_pgsql.conf.sample, res/res_config_pgsql.c, /: Merged
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revisions 265894 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ........ r265894 |
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tilghman | 2010-05-26 11:14:48 -0500 (Wed, 26 May 2010) | 8 lines
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Construct socket name, according to the Postgres docs, and
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document as such. (closes issue #17392) Reported by: dps Patches:
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20100525__issue17392.diff.txt uploaded by tilghman (license 14)
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Tested by: dps ........
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2010-05-26 15:52 +0000 [r265890] Mark Michelson <mmichelson@digium.com>
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* /, channels/chan_sip.c: Recorded merge of revisions 265842 via
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svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
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........ r265842 | mmichelson | 2010-05-26 09:41:55 -0500 (Wed,
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26 May 2010) | 9 lines Re-enable "always" option for videosupport
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option in sip.conf. (closes issue #17016) Reported by: twilson
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Patches: 17016.patch uploaded by mmichelson (license 60) Tested
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by: devmod ........
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2010-05-26 00:33 +0000 [r265748] Tilghman Lesher <tlesher@digium.com>
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* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
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pbx/pbx_lua.c: Merged revisions 265747 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ........ r265747 |
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tilghman | 2010-05-25 19:29:40 -0500 (Tue, 25 May 2010) | 8 lines
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Use configure to determine the prefixes and include directories
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properly. This ensures cross-platform compatibility, even among
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Linux distributions, which don't always put headers in the same
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place. (closes issue #17391) Reported by: loloski ........
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2010-05-25 21:05 +0000 [r265699] Mark Michelson <mmichelson@digium.com>
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* /, channels/chan_sip.c: Merged revisions 265698 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ........ r265698 |
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mmichelson | 2010-05-25 15:59:04 -0500 (Tue, 25 May 2010) | 12
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lines Properly use peer's outboundproxy for outbound REGISTERs.
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The logic used in transmit_register to get the outboundproxy for
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a peer was flawed since this value would be overridden shortly
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afterwards when create_addr was called. In addition, this also
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fixes some logic used when parsing users.conf so that the peer
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name is placed in the internally-generated register string so
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that an outboundproxy set in the Asterisk GUI will be used for
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outbound REGISTERs. ........
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2010-05-25 17:15 +0000 [r265615] David Vossel <dvossel@digium.com>
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* channels/chan_dahdi.c: fixes build issue with zaptel (closes
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issue 0017394) Reported by: aragon Patches: half_buffer_fix.diff
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uploaded by dvossel (license 671) Tested by: aragon
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2010-05-25 17:06 +0000 [r265612] Matthew Nicholson <mnicholson@digium.com>
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* apps/app_queue.c, /: Merged revisions 265611 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ................
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r265611 | mnicholson | 2010-05-25 12:00:11 -0500 (Tue, 25 May
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2010) | 15 lines Merged revisions 265610 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
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r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May
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2010) | 8 lines Don't mark the cdr records of unanswered queue
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calls with "NOANSWER". This restores the behavior prior to
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r258670. (closes issue #17334) Reported by: jvandal Patches:
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queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested
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by: aragon, jvandal ........ ................
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2010-05-24 23:52 +0000 [r265521] Terry Wilson <twilson@digium.com>
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* include/asterisk/options.h, main/asterisk.c, Makefile,
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doc/manager_1_1.txt, doc/tex/manager.tex, main/manager.c: Merged
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revisions 265320,265467 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ........ r265320 |
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twilson | 2010-05-24 14:06:40 -0500 (Mon, 24 May 2010) | 14 lines
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Add the FullyBooted AMI event It is possible to connect to the
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manager interface before all Asterisk modules are loaded. To
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ensure that an application does not send AMI actions that might
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require a module that has not yet loaded, the application can
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listen for the FullyBooted manager event. It will be sent upon
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connection if all modules have been loaded, or as soon as loading
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is complete. The event: Event: FullyBooted Privilege: system,all
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Status: Fully Booted Review:
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https://reviewboard.asterisk.org/r/639/ ........ r265467 |
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twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line
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Merge the rest of the FullyBooted patch ........
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2010-05-24 22:07 +0000 [r265450-265452] Mark Michelson <mmichelson@digium.com>
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* /, channels/h323/ast_h323.cxx: Merged revisions 265451 via
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svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
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........ r265451 | mmichelson | 2010-05-24 17:05:15 -0500 (Mon,
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24 May 2010) | 8 lines Print openh323 log to the Asterisk
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console. (closes issue #17109) Reported by: under Patches:
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logstream.diff uploaded by under (license 914) ........
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* /, channels/chan_sip.c: Merged revisions 265449 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ........ r265449 |
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mmichelson | 2010-05-24 16:44:30 -0500 (Mon, 24 May 2010) | 11
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lines Allow type=user SIP endpoints to be loaded properly from
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realtime. (closes issue #16021) Reported by: Guggemand Patches:
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realtime-type-fix.patch uploaded by Guggemand (license 897)
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(altered by me slightly to avoid ref leaks) Tested by: Guggemand
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........
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2010-05-24 19:30 +0000 [r265364] David Vossel <dvossel@digium.com>
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* main/channel.c, /: Merged revisions 265273 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ........ r265273 |
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dvossel | 2010-05-24 11:10:09 -0500 (Mon, 24 May 2010) | 2 lines
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fixes segfault when using generic plc ........
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2010-05-24 18:30 +0000 [r265318] Tilghman Lesher <tlesher@digium.com>
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* main/asterisk.c, /: Merged revisions 265316 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ........ r265316 |
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tilghman | 2010-05-24 13:19:08 -0500 (Mon, 24 May 2010) | 7 lines
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On systems with a LOT of RAM, a signed integer sometimes printed
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negative. (closes issue #16837) Reported by: jlpedrosa Patches:
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20100504__issue16837.diff.txt uploaded by tilghman (license 14)
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........
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2010-05-21 21:57 +0000 [r264998-265172] Mark Michelson <mmichelson@digium.com>
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* apps/app_queue.c: Fix memory hogging behavior of app_queue. From
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reviewboard: This review request is for the patch on issue 17081.
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A user reported that he saw increasing numbers of allocations
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stemming from app_queue.c when he would run the "queue show" CLI
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command. The user reported that he was using approximately 40
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realtime queues and as he ran the CLI command more and more, the
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memory usage would shoot up. As it turns out, there was a memory
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leak and a separate usage of memory that, while not really a
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leak, was very irresponsible. Both memory problems can be
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attributed to the function init_queue(). When the "queue show"
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command is run, all realtime queues have the init_queue()
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function called on the in-memory queue. The idea is to place the
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queue in its default state and then overwrite options specified
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in the realtime backend as we read them. The first problem, the
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memory leak, had to do with the fact that the string field for
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the name of the first periodic announcement file was being
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re-created every time init_queue was called. This patch corrects
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the behavior by only calling ast_str_create if the memory has not
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already been allocated. The other problem is a bit more
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complicated. The majority of the strings in the call_queue
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structure were changed to use the ast_string_fields API for 1.6.0
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and beyond. init_queue resets all string fields on the queue to
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their default values. Then, later in the realtime queue loading
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process, these string fields are set to their configured values.
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For those unfamiliar with string fields, frequent resizing of a
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string like this is not what the string fields API is designed
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for. The result of this constant resizing is that as the queue
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gets loaded, eventually space for the string runs out and so a
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new memory pool, at twice the size of the previously allocated
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one, is created for the string fields. The reporter of issue
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17081 wrote a script that ran the "queue show" CLI command 2100
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times. By the end, each of his 40 queues was taking about a
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megabyte of memory apiece just for their string fields. My fix
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for this problem is to revert the call_queue structure from using
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string fields. In my patch here, I have moved the queue back to
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using fixed-sized buffers. I ran the script provided by the
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reporter of 17081 and determined that I no longer saw the
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steadily-increasing memory usage that I had seen before applying
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the patch. (closes issue #17081) Reported by: wliegel Patches:
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17081v2.patch uploaded by mmichelson (license 60) Tested by:
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wliegel, mmichelson Review:
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https://reviewboard.asterisk.org/r/651/
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* apps/app_queue.c, include/asterisk/file.h, /: Merged revisions
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265090 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ................
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r265090 | mmichelson | 2010-05-21 16:08:51 -0500 (Fri, 21 May
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2010) | 15 lines Merged revisions 265089 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
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r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May
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2010) | 8 lines Don't hang up on a queue caller if the file we
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attempt to play does not exist. This also fixes a documentation
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mistake in file.h that made my original attempt to correct this
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problem not work correctly. (closes issue #17061) Reported by:
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RoadKill ........ ................
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* /, channels/chan_sip.c: Merged revisions 265087 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ........ r265087 |
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mmichelson | 2010-05-21 15:38:14 -0500 (Fri, 21 May 2010) | 7
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lines Be sure to set the sin_family on the proxy when allocating.
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(closes issue #17157) Reported by: stuarth ........
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* /, include/asterisk/channel.h: Merged revisions 265000 via
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svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
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................ r265000 | mmichelson | 2010-05-21 11:54:21 -0500
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(Fri, 21 May 2010) | 9 lines Merged revisions 264999 via svnmerge
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from https://origsvn.digium.com/svn/asterisk/branches/1.4
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........ r264999 | mmichelson | 2010-05-21 11:53:53 -0500 (Fri,
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21 May 2010) | 3 lines Fix grammatical error in comment. ........
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................
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* main/channel.c, main/autoservice.c, /,
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include/asterisk/channel.h: Merged revisions 264997 via svnmerge
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from https://origsvn.digium.com/svn/asterisk/trunk
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................ r264997 | mmichelson | 2010-05-21 11:44:27 -0500
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(Fri, 21 May 2010) | 38 lines Merged revisions 264996 via
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svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
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r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May
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2010) | 32 lines Allow ast_safe_sleep to defer specific frames
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until after the sleep has concluded. From reviewboard Background:
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A Digium customer discovered a somewhat odd bug. The setup is
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that parties A and B are bridged, and party A places party B on
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hold. While party B is listening to hold music, he mashes a bunch
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of DTMF. Party A takes party B off hold while this is happening,
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but party B continues to hear hold music. I could reproduce this
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about 1 in 5 times. The issue: When DTMF features are enabled and
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a user presses keys, the channel that the DTMF is streamed to is
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placed in an ast_safe_sleep for 100 ms, the duration of the
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emulated tone. If an AST_CONTROL_UNHOLD frame is read from the
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channel during the sleep, the frame is dropped. Thus the unhold
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indication is never made to the channel that was originally
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placed on hold. The fix: Originally, I discussed with Kevin
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possible ways of fixing the specific problem reported. However,
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we determined that the same type of problem could happen in other
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situations where ast_safe_sleep() is used. Using autoservice as a
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model, I modified ast_safe_sleep_conditional() to defer specific
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frame types so they can be re-queued once the sleep has finished.
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I made a common function for determining if a frame should be
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deferred so that there are not two identical switch blocks to
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maintain. Review: https://reviewboard.asterisk.org/r/674/
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........ ................
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2010-05-20 23:34 +0000 [r264829] Richard Mudgett <rmudgett@digium.com>
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* /, main/callerid.c: Merged revisions 264828 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ................
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r264828 | rmudgett | 2010-05-20 18:29:43 -0500 (Thu, 20 May 2010)
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| 13 lines Merged revisions 264820 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
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r264820 | rmudgett | 2010-05-20 18:23:21 -0500 (Thu, 20 May 2010)
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| 6 lines ast_callerid_parse() had a path that left name
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uninitialized. Several callers of ast_callerid_parse() do not
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initialize the name parameter before calling thus there is the
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potential to use an uninitialized pointer. ........
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................
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2010-05-20 22:24 +0000 [r264753-264783] Tilghman Lesher <tlesher@digium.com>
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* main/pbx.c, /: Merged revisions 264779 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ........ r264779 |
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tilghman | 2010-05-20 17:23:32 -0500 (Thu, 20 May 2010) | 8 lines
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Let ExtensionState resolve dynamic hints. (closes issue #16623)
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Reported by: tilghman Patches: 20100116__issue16623.diff.txt
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uploaded by tilghman (license 14) Tested by: lmadsen ........
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* apps/app_stack.c, /: Merged revisions 264752 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ........ r264752 |
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tilghman | 2010-05-20 16:28:53 -0500 (Thu, 20 May 2010) | 7 lines
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Error message fix. (closes issue #17356) Reported by: kenner
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Patches: app_stack.c.diff uploaded by kenner (license 1040)
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........
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2010-05-19 22:10 +0000 [r264453] Mark Michelson <mmichelson@digium.com>
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* include/asterisk/_private.h, include/asterisk/options.h,
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main/asterisk.c, main/loader.c, main/channel.c, /,
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channels/chan_sip.c: Merged revisions 264452 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ........ r264452 |
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mmichelson | 2010-05-19 16:29:08 -0500 (Wed, 19 May 2010) | 86
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lines Fix transcode_via_sln option with SIP calls and improve PLC
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usage. From reviewboard: The problem here is a bit complex, so
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try to bear with me... It was noticed by a Digium customer that
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generic PLC (as configured in codecs.conf) did not appear to
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actually be having any sort of benefit when packet loss was
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introduced on an RTP stream. I reproduced this issue myself by
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streaming a file across an RTP stream and dropping approx. 5% of
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the RTP packets. I saw no real difference between when PLC was
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enabled or disabled when using wireshark to analyze the RTP
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streams. After analyzing what was going on, it became clear that
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one of the problems faced was that when running my tests, the
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translation paths were being set up in such a way that PLC could
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not possibly work as expected. To illustrate, if packets are lost
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on channel A's read stream, then we expect that PLC will be
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applied to channel B's write stream. The problem is that generic
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PLC can only be done when there is a translation path that moves
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from some codec to SLINEAR. When I would run my tests, I found
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that every single time, read and write translation paths would be
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set up on channel A instead of channel B. There appeared to be no
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real way to predict which channel the translation paths would be
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set up on. This is where Kevin swooped in to let me know about
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the transcode_via_sln option in asterisk.conf. It is supposed to
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work by placing a read translation path on both channels from the
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channel's rawreadformat to SLINEAR. It also will place a write
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translation path on both channels from SLINEAR to the channel's
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rawwriteformat. Using this option allows one to predictably set
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up translation paths on all channels. There are two problems with
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this, though. First and foremost, the transcode_via_sln option
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did not appear to be working properly when I was placing a SIP
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call between two endpoints which did not share any common
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formats. Second, even if this option were to work, for PLC to be
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applied, there had to be a write translation path that would go
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from some format to SLINEAR. It would not work properly if the
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starting format of translation was SLINEAR. The one-line change
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presented in this review request in chan_sip.c fixed the first
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issue for me. The problem was that in sip_request_call, the
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jointcapability of the outbound channel was being set to the
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format passed to sip_request_call. This is nativeformats of the
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inbound channel. Because of this, when
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ast_channel_make_compatible was called by app_dial, both channels
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already had compatibly read and write formats. Thus, no
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translation path was set up at the time. My change is to set the
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jointcapability of the sip_pvt created during sip_request_call to
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the intersection of the inbound channel's nativeformats and the
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configured peer capability that we determined during the earlier
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call to create_addr. Doing this got the translation paths set up
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as expected when using transcode_via_sln. The changes presented
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in channel.c fixed the second issue for me. First and foremost,
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when Asterisk is started, we'll read codecs.conf to see the value
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of the genericplc option. If this option is set, and ast_write is
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called for a frame with no data, then we will attempt to fill in
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the missing samples for the frame. The implementation uses a
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channel datastore for maintaining the PLC state and for creating
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a buffer to store PLC samples in. Even when we receive a frame
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with data, we'll call plc_rx so that the PLC state will have
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knowledge of the previous voice frame, which it can use as a
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basis for when it comes time to actually do a PLC fill-in. So,
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reviewers, now I ask for your help. First off, there's the one
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line change in chan_sip that I have put in. Is it right? By my
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logic it seems correct, but I'm sure someone can tell me why it
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is not going to work. This is probably the change I'm least
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concerned about, though. What concerns me much more is the set of
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changes in channel.c. First off, am I even doing it right? When I
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run tests, I can clearly see that when PLC is activated, I see a
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significant increase in RTP traffic where I would expect it to
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be. However, in my humble opinion, the audio sounds kind of
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crappy whenever the PLC fill-in is done. It sounds worse to me
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than when no PLC is used at all. I need someone to review the
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logic I have used to be sure that I'm not misusing anything. As
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far as I can see my pointer arithmetic is correct, and my use of
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AST_FRIENDLY_OFFSET should be correct as well, but I'm sure
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someone can point out somewhere where I've done something
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incorrectly. As I was writing this review request up, I decided
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to give the code a test run under valgrind, and I find that for
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some reason, calls to plc_rx are causing some invalid reads.
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Apparently I'm reading past the end of a buffer somehow. I'll
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have to dig around a bit to see why that is the case. If it's
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obvious to someone reviewing, speak up! Finally, I have one other
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proposal that is not reflected in my code review. Since without
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transcode_via_sln set, one cannot predict or control where a
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translation path will be up, it seems to me that the current
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practice of using PLC only when transcoding to SLINEAR is not
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useful. I recommend that once it has been determined that the
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method used in this code review is correct and works as expected,
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then the code in translate.c that invokes PLC should be removed.
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Review: https://reviewboard.asterisk.org/r/622/ ........
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2010-05-19 20:31 +0000 [r264405] David Vossel <dvossel@digium.com>
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* main/udptl.c, /: Merged revisions 264400 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ........ r264400 |
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dvossel | 2010-05-19 15:30:33 -0500 (Wed, 19 May 2010) | 11 lines
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fixes infinite loop during udptl.c's decode_open_type When
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decode_length returns the length there is a check to see if that
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length is negative, if so the decode loop breaks as this means
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the limit has been reached. The problem here is that length is an
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unsigned int, so length can never be negative. This resulted in
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an infinite loop. (issue #17352) ........
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2010-05-19 20:27 +0000 [r264336-264388] Matthew Nicholson <mnicholson@digium.com>
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* main/udptl.c, /: Merged revisions 264379 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ........ r264379 |
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mnicholson | 2010-05-19 15:26:27 -0500 (Wed, 19 May 2010) | 4
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lines Cast an unsigned int to a signed int when comparing it with
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0. (AST-377) ........
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* apps/app_speech_utils.c, /: Merged revisions 264335 via svnmerge
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from https://origsvn.digium.com/svn/asterisk/trunk
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................ r264335 | mnicholson | 2010-05-19 15:02:57 -0500
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(Wed, 19 May 2010) | 12 lines Merged revisions 264334 via
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svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
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r264334 | mnicholson | 2010-05-19 15:01:38 -0500 (Wed, 19 May
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2010) | 5 lines Set quieted flag when receiving a dtmf tone
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during playback in speechbackground. (closes issue #16966)
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Reported by: asackheim ........ ................
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2010-05-19 19:25 +0000 [r264332] David Vossel <dvossel@digium.com>
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* /, channels/chan_sip.c: Merged revisions 264331 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ........ r264331 |
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dvossel | 2010-05-19 14:21:04 -0500 (Wed, 19 May 2010) | 13 lines
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fixes crash in check_rtp_timeout During deadlock avoidance the
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sip dialog pvt is locked and unlocked. When this occurs we have
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no guarantee the pvt's owner is still valid. We were trying to
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access the pvt's owner after this without checking to see if it
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still existed first. (closes issue #17271) Reported by: under
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Patches: check_rtp_timeout.diff uploaded by under (license 914)
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Tested by: dvossel ........
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2010-05-19 17:49 +0000 [r264205-264250] Tilghman Lesher <tlesher@digium.com>
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* include/asterisk/options.h, /, configure,
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include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
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264249 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ................
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r264249 | tilghman | 2010-05-19 12:48:31 -0500 (Wed, 19 May 2010)
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| 24 lines Merged revisions 264248 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
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r264248 | tilghman | 2010-05-19 12:41:29 -0500 (Wed, 19 May 2010)
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| 17 lines Internal timing is now on by default, if you're using
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DAHDI 2.3 or above. The reason for ensuring DAHDI 2.3 or above is
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that this version ensures that a timer is always available,
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whereas in previous versions, it was possible for DAHDI to be
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loaded, but have no drivers to actually generate timing. If
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internal_timing was turned on in this circumstance, a complete
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lack of audio would result. This is the reason why
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internal_timing was not on by default. However, now that DAHDI
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ensures the availability of a timer, there is no reason for this
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setting to be off (and in fact, it solves a great many initial
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user problems). (closes issue #15932) Reported by: dimas Patches:
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20100519__issue15932.diff.txt uploaded by tilghman (license 14)
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Tested by: tilghman ........ ................
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* main/dsp.c, /: Merged revisions 264204 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ........ r264204 |
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tilghman | 2010-05-19 11:42:20 -0500 (Wed, 19 May 2010) | 9 lines
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Keep track of digit duration, when we're decoding inband to pass
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DTMF frames. (closes issue #17235) Reported by: frawd Patches:
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new_dtmf_dsp_len.patch uploaded by frawd (license 610)
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20100518__issue17235.diff.txt uploaded by tilghman (license 14)
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Tested by: frawd ........
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2010-05-19 14:47 +0000 [r264115] David Vossel <dvossel@digium.com>
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* main/rtp.c, /: Merged revisions 264114 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ........ r264114 |
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dvossel | 2010-05-19 09:38:02 -0500 (Wed, 19 May 2010) | 13 lines
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fixes crash during dtmf During the processing of Cisco dtmf the
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dtmf samples were not being calculated correctly. In an attempt
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to determine what sample rate was being used, a NULL frame was
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processed which caused a crash. This patch resolves this. (closes
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issue #17248) Reported by: falves11 Patches: issue_17248.diff
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uploaded by dvossel (license 671) ........
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2010-05-19 08:15 +0000 [r264032] Alec L Davis <sivad.a@paradise.net.nz>
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* /, configs/indications.conf.sample: Merged revisions 264031 via
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svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
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........ r264031 | alecdavis | 2010-05-19 20:09:14 +1200 (Wed, 19
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May 2010) | 8 lines fix incorrectly typed indications for [nz]
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stutter and dialrecall (closes issue #17359) Reported by:
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alecdavis Patches: bug17359.diff.txt uploaded by alecdavis
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(license 585) ........
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2010-05-19 06:41 +0000 [r263951] Tilghman Lesher <tlesher@digium.com>
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* main/dsp.c, /: Merged revisions 263950 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ................
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r263950 | tilghman | 2010-05-19 01:41:04 -0500 (Wed, 19 May 2010)
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| 15 lines Merged revisions 263949 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
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r263949 | tilghman | 2010-05-19 01:32:27 -0500 (Wed, 19 May 2010)
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| 8 lines Because progress is called multiple times, across
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several frames, we must persist states when detecting multitone
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sequences. (closes issue #16749) Reported by: dant Patches:
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dsp.c-bug16749-1.patch uploaded by dant (license 670) Tested by:
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dant ........ ................
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2010-05-18 22:49 +0000 [r263906] David Vossel <dvossel@digium.com>
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* main/strings.c, /: Merged revisions 263904 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ........ r263904 |
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dvossel | 2010-05-18 17:48:51 -0500 (Tue, 18 May 2010) | 9 lines
619
fixes segfault on logging (closes issue #17331) Reported by:
620
under Patches: utils.diff uploaded by under (license 914)
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segfault_on_logging.diff uploaded by dvossel (license 671) Tested
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by: under, dvossel ........
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2010-05-18 19:41 +0000 [r263809] Jeff Peeler <jpeeler@digium.com>
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* apps/app_directory.c, /: Merged revisions 263807 via svnmerge
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from https://origsvn.digium.com/svn/asterisk/trunk
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................ r263807 | jpeeler | 2010-05-18 14:27:34 -0500
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(Tue, 18 May 2010) | 17 lines Merged revisions 263769 via
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svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
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r263769 | jpeeler | 2010-05-18 13:54:58 -0500 (Tue, 18 May 2010)
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| 10 lines Modify directory name reading to be interrupted with
634
operator or pound escape. In the case of accidentally entering
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the wrong first three letters for the reading, users could be
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very frustrated if the name listing is very long. This allows
637
interrupting the reading by pressing 0 or #. 0 will attempt to
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execute a configured operator (o) extension and # will exit and
639
proceed in the dialplan. ABE-2200 ........ ................
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2010-05-17 22:10 +0000 [r263642] Mark Michelson <mmichelson@digium.com>
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* /, main/devicestate.c: Merged revisions 263640 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ................
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r263640 | mmichelson | 2010-05-17 17:08:01 -0500 (Mon, 17 May
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2010) | 16 lines Merged revisions 263639 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
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r263639 | mmichelson | 2010-05-17 17:00:28 -0500 (Mon, 17 May
649
2010) | 10 lines Fix logic error when checking for a devstate
650
provider. When using strsep, if one of the list of specified
651
separators is not found, it is the first parameter to strsep
652
which is now NULL, not the pointer returned by strsep. This issue
653
isn't especially severe in that the worst it is likely to do is
654
waste some cycles when a device with no '/' and no ':' is passed
655
to ast_device_state. ........ ................
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2010-05-17 19:37 +0000 [r263587-263590] Tilghman Lesher <tlesher@digium.com>
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* apps/app_voicemail.c, /: Merged revisions 263589 via svnmerge
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from https://origsvn.digium.com/svn/asterisk/trunk ........
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r263589 | tilghman | 2010-05-17 14:31:15 -0500 (Mon, 17 May 2010)
662
| 9 lines With IMAP backend, messages in INBOX were counted twice
663
for MWI. (closes issue #17135) Reported by: edhorton Patches:
664
20100513__issue17135.diff.txt uploaded by tilghman (license 14)
665
17135_2.diff uploaded by ebroad (license 878) Tested by:
666
edhorton, ebroad ........
667
668
* main/app.c: Don't close 'n', just close 'above_n'. (closes issue
669
#17345) Reported by: wdoekes
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2010-05-17 14:41 +0000 [r263376-263458] Leif Madsen <lmadsen@digium.com>
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* main/manager.c, /: Merged revisions 263457 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ................
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r263457 | lmadsen | 2010-05-17 09:37:35 -0500 (Mon, 17 May 2010)
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| 19 lines Recorded merge of revisions 263456 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
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r263456 | lmadsen | 2010-05-17 09:35:18 -0500 (Mon, 17 May 2010)
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| 11 lines Manager cookies are not compatible with RFC2109. The
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Version field in the cookies we're setting contain quotes around
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the version number which is not compatible with RFC2109 and
682
breaks some implementations. (closes issue #17231) Reported by:
683
ecarruda Patches: manager_rfc2109-trunk-v1.patch uploaded by
684
ecarruda (license 559) manager_rfc2109-1.6.2-v1.patch uploaded by
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ecarruda (license 559) Tested by: ecarruda, russell ........
686
................
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* sounds/Makefile, /: Merged revisions 263375 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ................
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r263375 | lmadsen | 2010-05-17 09:05:33 -0500 (Mon, 17 May 2010)
691
| 16 lines Merged revisions 263374 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
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r263374 | lmadsen | 2010-05-17 09:04:57 -0500 (Mon, 17 May 2010)
694
| 8 lines Update link to new version of core sounds. The latest
695
version of the core sounds files 1.4.19 now includes the missing
696
queue-minute sound file which is called by app_queue but which
697
has been missing. (closes issue #17123) Reported by: n8ideas
698
........ ................
699
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2010-05-17 13:03 +0000 [r263293] David Vossel <dvossel@digium.com>
701
702
* CHANGES, channels/chan_dahdi.c: backport of DAHDI dynamic buffer
703
policy dialstring option
704
705
2010-05-15 23:41 +0000 [r263202] Paul Belanger <paul.belanger@polybeacon.com>
706
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* /, codecs/gsm/Makefile: Merged revisions 252488 via svnmerge from
708
https://origsvn.digium.com/svn/asterisk/trunk ........ r252488 |
709
tilghman | 2010-03-15 12:27:08 -0400 (Mon, 15 Mar 2010) | 9 lines
710
Make the Makefile logic more explicit and move the Snow Leopard
711
logic down to where it's not executed on non-Darwin systems.
712
(closes issue #17028) Reported by: pabelanger Patches:
713
issue17028_20100315.patch uploaded by seanbright (license 71)
714
20100315__issue17028.diff.txt uploaded by tilghman (license 14)
715
Tested by: tilghman, pabelanger ........
716
717
2010-05-13 22:13 +0000 [r263070] Richard Mudgett <rmudgett@digium.com>
718
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* channels/chan_dahdi.c, /: Merged revisions 263069 via svnmerge
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from https://origsvn.digium.com/svn/asterisk/trunk ........
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r263069 | rmudgett | 2010-05-13 17:01:36 -0500 (Thu, 13 May 2010)
722
| 1 line Fix inverted logic in cli command: ss7 set debug on/off
723
........
724
725
2010-05-13 15:36 +0000 [r262898] Russell Bryant <russell@digium.com>
726
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* channels/chan_console.c, /: Merged revisions 262897 via svnmerge
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from https://origsvn.digium.com/svn/asterisk/trunk ........
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r262897 | russell | 2010-05-13 10:36:12 -0500 (Thu, 13 May 2010)
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| 4 lines Fix an off by one error that causes a crash. Thanks to
731
Raymond Burke for pointing it out. ........
732
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2010-05-12 20:01 +0000 [r262801] Paul Belanger <paul.belanger@polybeacon.com>
734
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* main/loader.c, main/cli.c, /: Merged revisions 262800 via
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svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
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........ r262800 | pabelanger | 2010-05-12 15:59:16 -0400 (Wed,
738
12 May 2010) | 8 lines Notify CLI when modules is loaded /
739
unloaded (closes issue #17308) Reported by: pabelanger Patches:
740
cli.modules.patch uploaded by pabelanger (license 224) Tested by:
741
pabelanger, russell ........
742
743
2010-05-12 19:53 +0000 [r262797-262799] Leif Madsen <lmadsen@digium.com>
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* res/ael/pval.c, /: Merged revisions 262798 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ........ r262798 |
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lmadsen | 2010-05-12 14:53:10 -0500 (Wed, 12 May 2010) | 7 lines
748
Revert previous WARNING message removal. Marquis42 suggested a
749
better method of doing what I wanted because I ended up removing
750
the WARNING message for all instances when really I just wanted
751
to remove it for the 'return' keyword, not everything. (issue
752
#17145) ........
753
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* res/ael/pval.c, /: Merged revisions 262796 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ........ r262796 |
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lmadsen | 2010-05-12 14:31:42 -0500 (Wed, 12 May 2010) | 4 lines
757
Remove unnecessary WARNING message in ael/pval.c (closes issue
758
#17145) Reported by: okrief ........
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2010-05-12 18:03 +0000 [r262746] David Vossel <dvossel@digium.com>
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* /, apps/app_meetme.c: Merged revisions 262744 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk ................
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r262744 | dvossel | 2010-05-12 13:01:20 -0500 (Wed, 12 May 2010)
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| 17 lines Merged revisions 262662 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
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r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010)
768
| 11 lines fixes app_meetme dsp error We attempted to detect
769
silence after translating a frame from signed linear. This caused
770
a flooding of errors. To resolve this the code to detect silence
771
was moved before the translation. (closes issue #17133) Reported
772
by: jsdyer ........ ................
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2010-05-12 16:29 +0000 [r262516-262659] Tilghman Lesher <tlesher@digium.com>
775
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* /, apps/app_privacy.c: Merged revisions 262656 via svnmerge from
777
https://origsvn.digium.com/svn/asterisk/trunk ........ r262656 |
778
tilghman | 2010-05-12 11:23:26 -0500 (Wed, 12 May 2010) | 8 lines
779
Ensure the arguments are initialized. Also miscellaneous CG
780
cleanup. (closes issue #16576) Reported by: uxbod Patches:
781
20100505__issue16576.diff.txt uploaded by tilghman (license 14)
782
Tested by: uxbod ........
783
784
* /, include/asterisk/causes.h: Merged revisions 262513 via
785
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
786
........ r262513 | tilghman | 2010-05-11 16:25:05 -0500 (Tue, 11
787
May 2010) | 7 lines Move cause 200 to cause 26, as specified in
788
Q.850. Also cleanup the formatting and add a few more that seem
789
like good candidates. (closes issue #16157) Reported by: wimpy
790
........
791
792
2010-05-11 19:58 +0000 [r262425] Jason Parker <jparker@digium.com>
793
794
* /, res/Makefile: Merged revisions 262422 via svnmerge from
795
https://origsvn.digium.com/svn/asterisk/trunk ................
796
r262422 | qwell | 2010-05-11 14:57:24 -0500 (Tue, 11 May 2010) |
797
18 lines Merged revisions 262421 via svnmerge from
798
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
799
r262421 | qwell | 2010-05-11 14:55:42 -0500 (Tue, 11 May 2010) |
800
11 lines Use a less silly method for modifying a flex-generated
801
file. The sed syntax that was used wasn't actually valid, causing
802
some versions to choke. This is the method that is used in 1.6.x+
803
for similar changes. (closes issue #16696) Reported by: bklang
804
Patches: 16696-sedfix.diff uploaded by qwell (license 4) Tested
805
by: qwell ........ ................
806
807
2010-05-11 19:41 +0000 [r262415-262420] Paul Belanger <paul.belanger@polybeacon.com>
808
809
* pbx/pbx_config.c, /: Merged revisions 262419 via svnmerge from
810
https://origsvn.digium.com/svn/asterisk/trunk ........ r262419 |
811
pabelanger | 2010-05-11 15:40:37 -0400 (Tue, 11 May 2010) | 8
812
lines Improve logging by displaying line number (closes issue
813
#16303) Reported by: dant Patches: issue16303.patch.v2 uploaded
814
by pabelanger (license 224) Tested by: dant, lmadsen, pabelanger
815
........
816
817
* /, channels/chan_sip.c: Merged revisions 262414 via svnmerge from
818
https://origsvn.digium.com/svn/asterisk/trunk ........ r262414 |
819
pabelanger | 2010-05-11 15:26:17 -0400 (Tue, 11 May 2010) | 8
820
lines Improve logging information for misconfigured contexts
821
(closes issue #17238) Reported by: pprindeville Patches:
822
chan_sip-bug17238.patch uploaded by pprindeville (license 347)
823
Tested by: pprindeville ........
824
825
2010-05-11 17:25 +0000 [r262340] Tilghman Lesher <tlesher@digium.com>
826
827
* apps/app_voicemail.c, /, Makefile.rules: Merged revisions 262330
828
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
829
................ r262330 | tilghman | 2010-05-11 12:23:51 -0500
830
(Tue, 11 May 2010) | 9 lines Merged revisions 262321 via svnmerge
831
from https://origsvn.digium.com/svn/asterisk/branches/1.4
832
........ r262321 | tilghman | 2010-05-11 12:22:07 -0500 (Tue, 11
833
May 2010) | 2 lines Fix issue #17302 a slightly different way
834
(mad props to Qwell) ........ ................
835
836
2010-05-10 19:06 +0000 [r262237-262241] David Vossel <dvossel@digium.com>
837
838
* /, apps/app_directed_pickup.c: Merged revisions 262240 via
839
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
840
........ r262240 | dvossel | 2010-05-10 14:06:08 -0500 (Mon, 10
841
May 2010) | 9 lines fixes PickupChan application (closes issue
842
#16863) Reported by: schern Patches: app_directed_pickup.c.patch
843
uploaded by schern (license 995) for_trunk.diff uploaded by
844
cjacobsen (license 1029) Tested by: Graber, cjacobsen, lathama,
845
rickead2000, dvossel ........
846
847
* channels/chan_console.c, /: Merged revisions 262236 via svnmerge
848
from https://origsvn.digium.com/svn/asterisk/trunk ........
849
r262236 | dvossel | 2010-05-10 13:36:10 -0500 (Mon, 10 May 2010)
850
| 11 lines fixes crash in chan_console There is a race condition
851
between console_hangup() and start_stream(). It is possible for
852
console_hangup() to be called and then the stream thread to begin
853
after the hangup. To avoid this a check in start_stream() to make
854
sure the pvt-owner still exists while the pvt lock is held is
855
made. If the owner is gone that means the channel hung up and
856
start_stream should be aborted. ........
857
858
2010-05-10 16:39 +0000 [r262155] Tilghman Lesher <tlesher@digium.com>
859
860
* /, Makefile.rules: Merged revisions 262152 via svnmerge from
861
https://origsvn.digium.com/svn/asterisk/trunk ................
862
r262152 | tilghman | 2010-05-10 11:36:25 -0500 (Mon, 10 May 2010)
863
| 17 lines Merged revisions 262151 via svnmerge from
864
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
865
r262151 | tilghman | 2010-05-10 11:34:21 -0500 (Mon, 10 May 2010)
866
| 10 lines Allow compilation on Mac OS X 10.4 (Tiger) (closes
867
issue #17297) Reported by: jcovert Patches:
868
20100506__issue17297.diff.txt uploaded by tilghman (license 14)
869
(closes issue #17302) Reported by: jcovert ........
870
................
871
872
2010-05-09 02:17 +0000 [r261916-262105] Tilghman Lesher <tlesher@digium.com>
873
874
* autoconf/ast_ext_lib.m4, autoconf/ast_c_compile_check.m4,
875
autoconf/ast_c_define_check.m4, /, configure,
876
include/asterisk/autoconfig.h.in: Merged revisions 262102 via
877
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
878
........ r262102 | tilghman | 2010-05-08 21:14:04 -0500 (Sat, 08
879
May 2010) | 5 lines Cleanup a bit more by getting rid of useless
880
version defines. Also make library detection use passed CFLAGS.
881
(closes issue #17309) Reported by: stuarth ........
882
883
* /, configure, configure.ac: Merged revisions 262048 via svnmerge
884
from https://origsvn.digium.com/svn/asterisk/trunk ........
885
r262048 | tilghman | 2010-05-07 21:40:01 -0500 (Fri, 07 May 2010)
886
| 2 lines Use CPPFLAGS to pass PTHREAD_CFLAGS for vpb only
887
........
888
889
* /, funcs/func_odbc.c: Merged revisions 261917 via svnmerge from
890
https://origsvn.digium.com/svn/asterisk/trunk ........ r261917 |
891
tilghman | 2010-05-07 15:54:35 -0500 (Fri, 07 May 2010) | 8 lines
892
Double free crash (closes issue #17245) Reported by:
893
thedavidfactor Patches: 20100426__issue17245.diff.txt uploaded by
894
tilghman (license 14) Tested by: murraytm ........
895
896
* /, configure, include/asterisk/autoconfig.h.in, configure.ac:
897
Merged revisions 261913 via svnmerge from
898
https://origsvn.digium.com/svn/asterisk/trunk ........ r261913 |
899
tilghman | 2010-05-07 15:35:17 -0500 (Fri, 07 May 2010) | 14
900
lines Use the detected pthread building flags in every place,
901
instead of hardcoding -lpthread. We nicely detect the right flags
902
on each system for building Asterisk with pthreads, then ignore
903
it for every other build option that requires us to build with
904
pthreads. This caused some items to return a false negative. Also
905
cleanup some minor naming issues that caused "library library"
906
redundancy in the output. (closes issue #17303) Reported by:
907
stuarth Patches: 20100507__issue17303.diff.txt uploaded by
908
tilghman (license 14) Tested by: stuarth ........
909
910
2010-05-07 16:08 +0000 [r261868] Leif Madsen <lmadsen@digium.com>
911
912
* UPGRADE-1.6.txt, /: Merged revisions 261867 via svnmerge from
913
https://origsvn.digium.com/svn/asterisk/trunk ........ r261867 |
914
lmadsen | 2010-05-07 11:05:24 -0500 (Fri, 07 May 2010) | 6 lines
915
Update UPGRADE-1.6.txt stating insecure=very has been removed.
916
(closes issue #17282) Reported by: stuarth Tested by: stuarth
917
........
918
919
2010-05-06 20:13 +0000 [r261739] Jeff Peeler <jpeeler@digium.com>
920
921
* apps/app_voicemail.c, /: Merged revisions 261736 via svnmerge
922
from https://origsvn.digium.com/svn/asterisk/trunk
923
................ r261736 | jpeeler | 2010-05-06 15:11:53 -0500
924
(Thu, 06 May 2010) | 15 lines Merged revisions 261735 via
925
svnmerge from
926
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
927
r261735 | jpeeler | 2010-05-06 15:10:59 -0500 (Thu, 06 May 2010)
928
| 8 lines Only allow the operator key to be accepted after
929
leaving a voicemail. Or rather disallow the operator key from
930
being accepted when not offered, such as after finishing a
931
recording from within the mailbox options menu. ABE-2121 SWP-1267
932
........ ................
933
934
2010-05-06 17:08 +0000 [r261612] Jason Parker <jparker@digium.com>
935
936
* sounds/Makefile, /: Merged revisions 261609 via svnmerge from
937
https://origsvn.digium.com/svn/asterisk/trunk ................
938
r261609 | qwell | 2010-05-06 12:06:40 -0500 (Thu, 06 May 2010) |
939
11 lines Merged revisions 261608 via svnmerge from
940
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
941
r261608 | qwell | 2010-05-06 11:56:02 -0500 (Thu, 06 May 2010) |
942
4 lines Use the versioned MOH tarballs, now that we have them.
943
This makes for more reproducibility. Prompted by a discussion in
944
#asterisk-dev ........ ................
945
946
2010-05-06 15:43 +0000 [r261563] Tilghman Lesher <tlesher@digium.com>
947
948
* /, channels/chan_sip.c: Merged revisions 261560 via svnmerge from
949
https://origsvn.digium.com/svn/asterisk/trunk ........ r261560 |
950
tilghman | 2010-05-06 10:39:10 -0500 (Thu, 06 May 2010) | 8 lines
951
Permit more lines within a SIP body to be parsed. The example
952
given within the related issue showed 120 lines, which was mostly
953
a result of the body being XML. (closes issue #17179) Reported
954
by: khw ........
955
956
2010-06-01 Leif Madsen <lmadsen@digium.com>
957
958
* Asterisk 1.6.2.8 Released.
959
960
2010-05-26 Leif Madsen <lmadsen@digium.com>
961
962
* Asterisk 1.6.2.8-rc2 Released.
963
964
2010-05-26 10:56 -0500 [r265891] Matt Nicholson <mnicholson@digium.com>
965
966
* Merged r265610 from 1.4:
967
968
Don't mark the cdr records of unanswered queue calls with "NOANSWER".
969
This restores the behavior prior to r258670.
970
971
(closes issue #17334)
972
Reported by: jvandal
973
Patches:
974
queue-cdr-fixes1.diff uploaded by mnicholson (license 96)
975
Tested by: aragon, jvandal
976
977
2010-05-06 Leif Madsen <lmadsen@digium.com>
978
979
* Asterisk 1.6.2.8-rc1 Released
980
981
2010-05-06 14:07 +0000 [r261498-261499] Russell Bryant <russell@digium.com>
982
983
* tests/test_heap.c: Add test case that ensures the heap handles
984
arbitrary removals properly. (issue #17277) Reported by:
985
cappucinoking Patches: test_heap.diff uploaded by cappucinoking
986
(license 1036) Tested by: cappucinoking, russell
987
988
* /, main/heap.c: Merged revisions 261496 via svnmerge from
989
https://origsvn.digium.com/svn/asterisk/trunk ........ r261496 |
990
russell | 2010-05-06 08:58:07 -0500 (Thu, 06 May 2010) | 40 lines
991
Fix handling of removing nodes from the middle of a heap. This
992
bug surfaced in 1.6.2 and does not affect code in any other
993
released version of Asterisk. It manifested itself as SIP qualify
994
not happening when it should, causing peers to go unreachable.
995
This was debugged down to scheduler entries sometimes not getting
996
executed when they were supposed to, which was in turn caused by
997
an error in the heap code. The problem only sometimes occurs, and
998
it is due to the logic for removing an entry in the heap from an
999
arbitrary location (not just popping off the top). The scheduler
1000
performs this operation frequently when entries are removed
1001
before they run (when ast_sched_del() is used). In a normal pop
1002
off of the top of the heap, a node is taken off the bottom,
1003
placed at the top, and then bubbled down until the max heap
1004
property is restored (see max_heapify()). This same logic was
1005
used for removing an arbitrary node from the middle of the heap.
1006
Unfortunately, that logic is full of fail. This patch fixes that
1007
by fully restoring the max heap property when a node is thrown
1008
into the middle of the heap. Instead of just pushing it down as
1009
appropriate, it first pushes it up as high as it will go, and
1010
_then_ pushes it down. Lastly, fix a minor problem in
1011
ast_heap_verify(), which is only used for debugging. If a parent
1012
and child node have the same value, that is not an error. The
1013
only error is if a parent's value is less than its children. A
1014
huge thanks goes out to cappucinoking for debugging this down to
1015
the scheduler, and then producing an ast_heap test case that
1016
demonstrated the breakage. That made it very easy for me to focus
1017
on the heap logic and produce a fix. Open source projects are
1018
awesome. (closes issue #16936) Reported by: ib2 Tested by:
1019
cappucinoking, crjw (closes issue #17277) Reported by:
1020
cappucinoking Patches: heap-fix.rev2.diff uploaded by russell
1021
(license 2) Tested by: cappucinoking, russell ........
1022
1023
2010-05-06 07:43 +0000 [r261453] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
1024
1025
* channels/chan_dahdi.c, /: Merged revisions 261451 via svnmerge
1026
from https://origsvn.digium.com/svn/asterisk/trunk ........
1027
r261451 | tzafrir | 2010-05-06 10:27:31 +0300 (ה', 06 מאי 2010) |
1028
4 lines When failing to configure, don't destroy 'cfg' twice
1029
Fixes a crash when some config section had an incorrect channel
1030
config. ........
1031
1032
2010-05-05 19:08 +0000 [r261233-261315] Paul Belanger <paul.belanger@polybeacon.com>
1033
1034
* /, channels/chan_sip.c: Merged revisions 261314 via svnmerge from
1035
https://origsvn.digium.com/svn/asterisk/trunk ................
1036
r261314 | pabelanger | 2010-05-05 14:43:03 -0400 (Wed, 05 May
1037
2010) | 19 lines Merged revisions 261274 via svnmerge from
1038
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1039
r261274 | pabelanger | 2010-05-05 12:42:22 -0400 (Wed, 05 May
1040
2010) | 12 lines Registration fix for SIP realtime. Make sure
1041
realtime fields are not empty. (closes issue #17266) Reported by:
1042
Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick
1043
Lewis (license 657) Tested by: Nick_Lewis, sberney Review:
1044
https://reviewboard.asterisk.org/r/643/ ........ ................
1045
1046
* apps/app_queue.c, /: Merged revisions 261232 via svnmerge from
1047
https://origsvn.digium.com/svn/asterisk/trunk ........ r261232 |
1048
pabelanger | 2010-05-05 11:42:07 -0400 (Wed, 05 May 2010) | 10
1049
lines 'queue reset stats' erroneously clears wrapuptime
1050
configuration. Resets each member's lastcall to 0 now. (closes
1051
issue #17262, #16519) Reported by: rain Patches:
1052
wrapuptime_reset_fix.diff uploaded by rain (license 327) Tested
1053
by: rain ........
1054
1055
2010-05-04 23:55 +0000 [r261098] Tilghman Lesher <tlesher@digium.com>
1056
1057
* main/channel.c, /: Merged revisions 261095 via svnmerge from
1058
https://origsvn.digium.com/svn/asterisk/trunk ................
1059
r261095 | tilghman | 2010-05-04 18:51:52 -0500 (Tue, 04 May 2010)
1060
| 18 lines Merged revisions 261093-261094 via svnmerge from
1061
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1062
r261093 | tilghman | 2010-05-04 18:36:53 -0500 (Tue, 04 May 2010)
1063
| 7 lines Protect against overflow, when calculating how long to
1064
wait for a frame. (closes issue #17128) Reported by: under
1065
Patches: d.diff uploaded by under (license 914) ........ r261094
1066
| tilghman | 2010-05-04 18:47:08 -0500 (Tue, 04 May 2010) | 2
1067
lines Add a tiny corner case to the previous commit ........
1068
................
1069
1070
2010-05-04 19:01 +0000 [r260927] Jeff Peeler <jpeeler@digium.com>
1071
1072
* apps/app_voicemail.c, /: Merged revisions 260924 via svnmerge
1073
from https://origsvn.digium.com/svn/asterisk/trunk
1074
................ r260924 | jpeeler | 2010-05-04 13:51:28 -0500
1075
(Tue, 04 May 2010) | 18 lines Merged revisions 260923 via
1076
svnmerge from
1077
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1078
r260923 | jpeeler | 2010-05-04 13:46:46 -0500 (Tue, 04 May 2010)
1079
| 12 lines Voicemail transfer to operator should occur
1080
immediately, not after main menu. There were two scenarios in the
1081
advanced options that while using the operator=yes and review=yes
1082
options, the transfer occurred only after exiting the main menu
1083
(after sending a reply or leaving a message for an extension).
1084
Now after the audio is processed for the reply or message the
1085
transfer occurs immediately as expected. ABE-2107 ABE-2108
1086
........ ................
1087
1088
2010-05-04 16:58 +0000 [r260884] Matthew Nicholson <mnicholson@digium.com>
1089
1090
* configs/sip.conf.sample, include/asterisk/frame.h,
1091
main/channel.c, /, channels/chan_sip.c: Merged revisions 254450
1092
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
1093
........ r254450 | kpfleming | 2010-03-25 10:27:31 -0500 (Thu, 25
1094
Mar 2010) | 49 lines Improve handling of T.38 re-INVITEs that
1095
arrive before a T.38-capable application is executing on a
1096
channel. This patch addresses an issue found during working with
1097
end-users using res_fax. If an incoming call is answered in the
1098
dialplan, or jumps to the 'fax' extension due to reception of a
1099
CNG tone (with faxdetect enabled), and then the remote endpoint
1100
sends a T.38 re-INVITE, it is possible for the channel's T.38
1101
state to be 'T38_STATE_NEGOTIATING' when the application starts
1102
up. Unfortunately, even if the application wants to use T.38, it
1103
can't respond to the peer's negotiation request, because the
1104
AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent
1105
originally has been lost, and the application needs the content
1106
of that frame to be able to formulate a reply. This patch adds a
1107
new 'request' type to AST_CONTROL_T38_PARAMETERS,
1108
AST_T38_REQUEST_PARMS. If the application sends this request,
1109
chan_sip will re-send the original control frame (with
1110
AST_T38_REQUEST_NEGOTIATE as the request type), and the
1111
application can respond as normal. If this occurs within the five
1112
second timeout in chan_sip, the automatic cancellation of the
1113
peer reinvite will be stopped, and the application will 'own' the
1114
negotiation process from that point onwards. This also improves
1115
the code path in chan_sip to allow sip_indicate(), when called
1116
for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero
1117
response, which should have been in place before since the
1118
control frame *can* fail to be processed properly. It also
1119
modifies ast_indicate() to return whatever result the channel
1120
driver returned for this control frame, rather than converting
1121
all non-zero results into '-1'. Finally, the new request type
1122
intentionally returns a positive value, so that an application
1123
that sends AST_T38_REQUEST_PARMS can know for certain whether the
1124
channel driver accepted it and will be replying with a control
1125
frame of its own, or whether it was ignored (if the
1126
sip_indicate()/ast_indicate() path had properly supported failure
1127
responses before, this would not be necessary). This patch also
1128
modifies res_fax to take advantage of the new request. In
1129
addition, this patch makes sip_t38_abort() actually lock the
1130
private structure before doing its work... bad programmer, no
1131
donut. This patch also enhances chan_sip's 'faxdetect' support to
1132
allow triggering on T.38 re-INVITEs received as well as CNG tone
1133
detection. Review: https://reviewboard.asterisk.org/r/556/
1134
........
1135
1136
2010-05-04 15:51 +0000 [r260746-260805] Jason Parker <jparker@digium.com>
1137
1138
* /, build_tools/make_build_h: Merged revisions 260802 via svnmerge
1139
from https://origsvn.digium.com/svn/asterisk/trunk
1140
................ r260802 | qwell | 2010-05-04 10:49:57 -0500
1141
(Tue, 04 May 2010) | 9 lines Merged revisions 260801 via svnmerge
1142
from https://origsvn.digium.com/svn/asterisk/branches/1.4
1143
........ r260801 | qwell | 2010-05-04 10:49:27 -0500 (Tue, 04 May
1144
2010) | 1 line Fix fallout from removing from configure script.
1145
Pointed out by philipp64 on #asterisk-dev ........
1146
................
1147
1148
* /: Fix merge props
1149
1150
2010-05-03 17:42 +0000 [r260743] Paul Belanger <paul.belanger@polybeacon.com>
1151
1152
* Makefile, /: Merged revisions 260661-260662 via svnmerge from
1153
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1154
r260661 | pabelanger | 2010-05-03 12:41:30 -0400 (Mon, 03 May
1155
2010) | 10 lines non-root make install PREFIX=/tmp fails. Prepend
1156
libdir when executing mkpkgconfig allowing non-root installs to
1157
work. (closes issue #17268) Reported by: pabelanger Patches:
1158
issue17268.patch uploaded by pabelanger (license 224) Tested by:
1159
pabelanger ........ r260662 | pabelanger | 2010-05-03 12:54:41
1160
-0400 (Mon, 03 May 2010) | 3 lines Should have removed /usr/lib/
1161
part. Thanks Qwell. ........
1162
1163
2010-05-03 14:59 +0000 [r260571] Leif Madsen <lmadsen@digium.com>
1164
1165
* doc/HOWTO_collect_debug_information.txt: Merged revisions 260570
1166
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
1167
................ r260570 | lmadsen | 2010-05-03 09:58:23 -0500
1168
(Mon, 03 May 2010) | 9 lines Merged revisions 260569 via svnmerge
1169
from https://origsvn.digium.com/svn/asterisk/branches/1.4
1170
........ r260569 | lmadsen | 2010-05-03 09:57:39 -0500 (Mon, 03
1171
May 2010) | 1 line Minor typo pointed out by pabelanger on IRC.
1172
........ ................
1173
1174
2010-04-30 22:48 +0000 [r260441] Jeff Peeler <jpeeler@digium.com>
1175
1176
* channels/chan_dahdi.c, /: Merged revisions 260437 via svnmerge
1177
from https://origsvn.digium.com/svn/asterisk/trunk
1178
................ r260437 | jpeeler | 2010-04-30 17:36:49 -0500
1179
(Fri, 30 Apr 2010) | 18 lines Merged revisions 260434 via
1180
svnmerge from
1181
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1182
r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010)
1183
| 11 lines Ensure channel state is not incorrectly set in the
1184
case of a very early answer. The needringing bit was being read
1185
in dahdi_read after answering thereby setting the state to
1186
ringing from up. This clears needringing upon answering so that
1187
is no longer possible. (closes issue #17067) Reported by: tzafrir
1188
Patches: needringing.diff uploaded by tzafrir (license 46)
1189
........ ................
1190
1191
2010-04-30 20:22 +0000 [r260373] Mark Michelson <mmichelson@digium.com>
1192
1193
* res/res_musiconhold.c, /: Merged revisions 260346 via svnmerge
1194
from https://origsvn.digium.com/svn/asterisk/trunk
1195
................ r260346 | mmichelson | 2010-04-30 15:11:02 -0500
1196
(Fri, 30 Apr 2010) | 24 lines Merged revisions 260345 via
1197
svnmerge from
1198
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1199
r260345 | mmichelson | 2010-04-30 15:08:15 -0500 (Fri, 30 Apr
1200
2010) | 18 lines Fix potential crash from race condition due to
1201
accessing channel data without the channel locked. In
1202
res_musiconhold.c, there are several places where a channel's
1203
stream's existence is checked prior to calling ast_closestream on
1204
it. The issue here is that in several cases, the channel was not
1205
locked while checking the stream. The result was that if two
1206
threads checked the state of the channel's stream at
1207
approximately the same time, then there could be a situation
1208
where both threads attempt to call ast_closestream on the
1209
channel's stream. The result here is that the refcount for the
1210
stream would go below 0, resulting in a crash. I have added
1211
proper channel locking to res_musiconhold.c to ensure that we do
1212
not try to check chan->stream without the channel locked. A
1213
Digium customer has been using this patch for several weeks and
1214
has not had any crashes since applying the patch. ABE-2147
1215
........ ................
1216
1217
2010-04-30 06:22 +0000 [r260281-260303] Tilghman Lesher <tlesher@digium.com>
1218
1219
* /, main/app.c: Merged revisions 260292 via svnmerge from
1220
https://origsvn.digium.com/svn/asterisk/trunk ........ r260292 |
1221
tilghman | 2010-04-30 01:19:35 -0500 (Fri, 30 Apr 2010) | 13
1222
lines Don't allow file descriptors to go above 64k, when we're
1223
closing them in a fork(2). This saves time, when, even though the
1224
system allows the process limit to be that high, the practical
1225
limit is much lower. (closes issue #17223) Reported by:
1226
dbackeberg Patches: 20100423__issue17223.diff.txt uploaded by
1227
tilghman (license 14) Tested by: dbackeberg ........
1228
1229
* configs/extensions.conf.sample, /: Merged revisions 260280 via
1230
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
1231
........ r260280 | tilghman | 2010-04-30 00:23:56 -0500 (Fri, 30
1232
Apr 2010) | 7 lines Logic fixups for a sample FREENUM dialplan
1233
context. (closes issue #17263) Reported by: pprindeville Patches:
1234
freenum-dialplan.patch#3 uploaded by pprindeville (license 347)
1235
........
1236
1237
2010-04-29 23:13 +0000 [r260234] Richard Mudgett <rmudgett@digium.com>
1238
1239
* channels/chan_dahdi.c, /: Merged revisions 260231 via svnmerge
1240
from https://origsvn.digium.com/svn/asterisk/trunk
1241
................ r260231 | rmudgett | 2010-04-29 17:44:14 -0500
1242
(Thu, 29 Apr 2010) | 33 lines Merged revisions 260195 via
1243
svnmerge from
1244
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1245
r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010)
1246
| 26 lines DTMF CallerID detection problems. The code handling
1247
DTMF CallerID drops digits on long CallerID numbers and may
1248
timeout waiting for the first ring with shorter numbers. The DTMF
1249
emulation mode was not turned off when processing DTMF CallerID.
1250
When the emulation code gets behind in processing the DTMF digits
1251
it can skip a digit. For shorter numbers, the timeout may have
1252
been too short. I increased it from 2 seconds to 4 seconds. Four
1253
seconds is a typical time between rings for many countries.
1254
(closes issue #16460) Reported by: sum Patches: issue16460.patch
1255
uploaded by rmudgett (license 664) issue16460_v1.6.2.patch
1256
uploaded by rmudgett (license 664) Tested by: sum, rmudgett
1257
Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA
1258
AST-334 JIRA SWP-901 ........ ................
1259
1260
2010-04-29 18:18 +0000 [r260156] Tilghman Lesher <tlesher@digium.com>
1261
1262
* configs/extensions.conf.sample, /: Merged revisions 260148 via
1263
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
1264
........ r260148 | tilghman | 2010-04-29 13:15:57 -0500 (Thu, 29
1265
Apr 2010) | 2 lines Pattern match fail. ........
1266
1267
2010-04-29 15:35 +0000 [r260051] David Vossel <dvossel@digium.com>
1268
1269
* main/audiohook.c, /, include/asterisk/audiohook.h: Merged
1270
revisions 260050 via svnmerge from
1271
https://origsvn.digium.com/svn/asterisk/trunk ................
1272
r260050 | dvossel | 2010-04-29 10:33:27 -0500 (Thu, 29 Apr 2010)
1273
| 21 lines Merged revisions 260049 via svnmerge from
1274
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1275
r260049 | dvossel | 2010-04-29 10:31:02 -0500 (Thu, 29 Apr 2010)
1276
| 14 lines Fixes crash in audiohook_write_list The middle_frame
1277
in the audiohook_write_list function was being freed if a
1278
audiohook manipulator returned a failure. This is incorrect
1279
logic. This patch resolves this and adds detailed descriptions of
1280
how this function should work and why manipulator failures must
1281
be ignored. (closes issue #17052) Reported by: dvossel Tested by:
1282
dvossel (closes issue #16196) Reported by: atis Review:
1283
https://reviewboard.asterisk.org/r/623/ ........ ................
1284
1285
2010-04-28 22:36 +0000 [r259959] Mark Michelson <mmichelson@digium.com>
1286
1287
* /, channels/chan_sip.c: Merged revisions 259957 via svnmerge from
1288
https://origsvn.digium.com/svn/asterisk/trunk ........ r259957 |
1289
mmichelson | 2010-04-28 17:34:15 -0500 (Wed, 28 Apr 2010) | 11
1290
lines Don't override peer context with domain context. (closes
1291
issue #17040) Reported by: pprindeville Patches:
1292
asterisk-1.6-bugid17040.patch uploaded by pprindeville (license
1293
347) Tested by: pprindeville Review:
1294
https://reviewboard.asterisk.org/r/565/ ........
1295
1296
2010-04-28 21:26 +0000 [r259899] David Vossel <dvossel@digium.com>
1297
1298
* main/channel.c, channels/chan_local.c, /: Merged revisions 259870
1299
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
1300
................ r259870 | dvossel | 2010-04-28 16:20:03 -0500
1301
(Wed, 28 Apr 2010) | 39 lines Merged revisions 259858 via
1302
svnmerge from
1303
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1304
r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010)
1305
| 33 lines resolves deadlocks in chan_local Issue_1. In the
1306
local_hangup() 3 locks must be held at the same time... pvt,
1307
pvt->chan, and pvt->owner. Proper deadlock avoidance is done when
1308
the channel to hangup is the outbound chan_local channel, but
1309
when it is not the outbound channel we have an issue... We
1310
attempt to do deadlock avoidance only on the tech pvt, when both
1311
the tech pvt and the pvt->owner are locked coming into that loop.
1312
By never giving up the pvt->owner channel deadlock avoidance is
1313
not entirely possible. This patch resolves that by doing deadlock
1314
avoidance on both the pvt->owner and the pvt when trying to get
1315
the pvt->chan lock. Issue_2. ast_prod() is used in
1316
ast_activate_generator() to queue a frame on the channel and make
1317
the channel's read function get called. This function is used in
1318
ast_activate_generator() while the channel is locked, which
1319
mean's the channel will have a lock both from the generator code
1320
and the frame_queue code by the time it gets to chan_local.c's
1321
local_queue_frame code... local_queue_frame contains some of the
1322
same crazy deadlock avoidance that local_hangup requires, and
1323
this recursive lock prevents that deadlock avoidance from
1324
happening correctly. This patch removes ast_prod() from the
1325
channel lock so only one lock is held during the
1326
local_queue_frame function. (closes issue #17185) Reported by:
1327
schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel
1328
(license 671) issue_17185_v2.diff uploaded by dvossel (license
1329
671) Tested by: schmoozecom, GameGamer43 Review:
1330
https://reviewboard.asterisk.org/r/631/ ........ ................
1331
1332
2010-04-28 21:09 +0000 [r259854] Leif Madsen <lmadsen@digium.com>
1333
1334
* config.guess: Merged revisions 259853 via svnmerge from
1335
https://origsvn.digium.com/svn/asterisk/trunk ................
1336
r259853 | lmadsen | 2010-04-28 16:08:34 -0500 (Wed, 28 Apr 2010)
1337
| 14 lines Merged revisions 259852 via svnmerge from
1338
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1339
r259852 | lmadsen | 2010-04-28 16:07:48 -0500 (Wed, 28 Apr 2010)
1340
| 6 lines Update config.guess. Updating config.guess because
1341
after installing Ubuntu Server 9.10 and running all the update
1342
scripts, running ./configure would not continue because it was
1343
unable to determine what kind of system I had. After updating
1344
config.guess things started working again. ........
1345
................
1346
1347
2010-04-28 20:34 +0000 [r259781-259851] Jason Parker <jparker@digium.com>
1348
1349
* /, configure, configure.ac: Merged revisions 259848 via svnmerge
1350
from https://origsvn.digium.com/svn/asterisk/trunk
1351
................ r259848 | qwell | 2010-04-28 15:32:14 -0500
1352
(Wed, 28 Apr 2010) | 9 lines Merged revisions 259847 via svnmerge
1353
from https://origsvn.digium.com/svn/asterisk/branches/1.4
1354
........ r259847 | qwell | 2010-04-28 15:30:21 -0500 (Wed, 28 Apr
1355
2010) | 1 line Add AC_CONFIG_AUX_DIR to configure script, so
1356
systems without install can use install-sh from our source dir.
1357
........ ................
1358
1359
* makeopts.in, /: Merged revisions 259837 via svnmerge from
1360
https://origsvn.digium.com/svn/asterisk/trunk ................
1361
r259837 | qwell | 2010-04-28 15:26:35 -0500 (Wed, 28 Apr 2010) |
1362
9 lines Merged revisions 259833 via svnmerge from
1363
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1364
r259833 | qwell | 2010-04-28 15:25:36 -0500 (Wed, 28 Apr 2010) |
1365
1 line Missed this when removing $ID ........ ................
1366
1367
* Makefile, /, configure, configure.ac: Merged revisions 259760 via
1368
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
1369
................ r259760 | qwell | 2010-04-28 14:19:54 -0500
1370
(Wed, 28 Apr 2010) | 14 lines Merged revisions 259748 via
1371
svnmerge from
1372
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1373
r259748 | qwell | 2010-04-28 14:17:38 -0500 (Wed, 28 Apr 2010) |
1374
7 lines Remove usage of `id` since it isn't useful and was
1375
causing breakge. Solaris `id` doesn't support the -u argument.
1376
Instead of figuring out how to fix this to work on Solaris, I
1377
decided to check why it was necessary and where else it was used.
1378
It was only used in one place, and it hasn't been needed for a
1379
very long time (I question whether it was ever needed). ........
1380
................
1381
1382
2010-04-28 17:19 +0000 [r259681] Jeff Peeler <jpeeler@digium.com>
1383
1384
* apps/app_voicemail.c, /: Merged revisions 259672 via svnmerge
1385
from https://origsvn.digium.com/svn/asterisk/trunk
1386
................ r259672 | jpeeler | 2010-04-28 12:18:43 -0500
1387
(Wed, 28 Apr 2010) | 11 lines Merged revisions 259664 via
1388
svnmerge from
1389
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1390
r259664 | jpeeler | 2010-04-28 12:13:29 -0500 (Wed, 28 Apr 2010)
1391
| 4 lines Do not play goodbye prompt after timeout of message
1392
review. ABE-2124 ........ ................
1393
1394
2010-04-27 22:46 +0000 [r259616] Richard Mudgett <rmudgett@digium.com>
1395
1396
* channels/chan_dahdi.c, /: Merged revisions 259538 via svnmerge
1397
from https://origsvn.digium.com/svn/asterisk/trunk
1398
................ r259538 | rmudgett | 2010-04-27 17:18:09 -0500
1399
(Tue, 27 Apr 2010) | 18 lines Merged revisions 259531 via
1400
svnmerge from
1401
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1402
r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27 Apr 2010)
1403
| 11 lines DAHDI "WARNING" message is confusing and vague
1404
"WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed
1405
failed: Success" Changed the warning to "Failed to decode
1406
CallerID on channel 'name'". The message before it is likely more
1407
specific about why the CallerID decode failed. SWP-501 AST-283
1408
........ ................
1409
1410
2010-04-27 21:50 +0000 [r259528] Leif Madsen <lmadsen@digium.com>
1411
1412
* sounds/Makefile: Merged revisions 259527 via svnmerge from
1413
https://origsvn.digium.com/svn/asterisk/trunk ................
1414
r259527 | lmadsen | 2010-04-27 16:49:36 -0500 (Tue, 27 Apr 2010)
1415
| 23 lines Merged revisions 259526 via svnmerge from
1416
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1417
r259526 | lmadsen | 2010-04-27 16:48:47 -0500 (Tue, 27 Apr 2010)
1418
| 15 lines Update sounds files. * Add additional sounds prompts
1419
for say_enumeration * Update the English conference sounds
1420
prompts so they are better quality and all sound more consistent
1421
* Clean up the core-sounds-XX.txt and extra-sounds-XX.txt files
1422
to include all present sound files Both core (en, fr, es) and
1423
extra (en, fr) sounds files have been updated. (closes issue
1424
#16200) Reported by: murf (closes issue #17137) Reported by:
1425
lmadsen ........ ................
1426
1427
2010-04-27 21:25 +0000 [r259356-259486] Jason Parker <jparker@digium.com>
1428
1429
* main/editline/configure.in, /, main/editline/configure,
1430
main/editline/Makefile.in: Merged revisions 259439 via svnmerge
1431
from https://origsvn.digium.com/svn/asterisk/trunk ........
1432
r259439 | qwell | 2010-04-27 16:13:01 -0500 (Tue, 27 Apr 2010) |
1433
5 lines Add gar to the check for AR for those silly OSes
1434
(Solaris) that don't have ar. autoconf2.13 couldn't handle
1435
AC_PROG_GREP, so I removed it. This is fine, since we don't need
1436
to use anything that the configure script doesn't. ........
1437
1438
* /: Unblock revision 259439.
1439
1440
* /, configure, configure.ac: Merged revisions 259353 via svnmerge
1441
from https://origsvn.digium.com/svn/asterisk/trunk
1442
................ r259353 | qwell | 2010-04-27 14:31:55 -0500
1443
(Tue, 27 Apr 2010) | 12 lines Merged revisions 259352 via
1444
svnmerge from
1445
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1446
r259352 | qwell | 2010-04-27 14:29:26 -0500 (Tue, 27 Apr 2010) |
1447
5 lines Support the silly OSes that don't have ar and strip.
1448
Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when path isn't
1449
specified, and AC_PATH_TOOLS doesn't exist, we'll just switch to
1450
AC_CHECK_TOOLS. ........ ................
1451
1452
2010-04-27 19:03 +0000 [r259310] Richard Mudgett <rmudgett@digium.com>
1453
1454
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
1455
revisions 259307 via svnmerge from
1456
https://origsvn.digium.com/svn/asterisk/trunk ................
1457
r259307 | rmudgett | 2010-04-27 13:29:33 -0500 (Tue, 27 Apr 2010)
1458
| 21 lines Merged revisions 259270 via svnmerge from
1459
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1460
r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010)
1461
| 14 lines hidecalleridname parameter in chan_dahdi.conf Issue
1462
#7321 implements a new chan_dahdi configuration option. However,
1463
a change mentioned in the issue was never implemented. This is
1464
the change that will allow the feature to work. I added a note to
1465
chan_dahdi.conf.sample about the feature. (closes issue #17143)
1466
Reported by: djensen99 Patches: diff.txt uploaded by djensen99
1467
(license NA) (One line change) Tested by: djensen99 ........
1468
................
1469
1470
2010-04-26 21:48 +0000 [r259103-259109] Mark Michelson <mmichelson@digium.com>
1471
1472
* main/channel.c, /: Merged revisions 259105 via svnmerge from
1473
https://origsvn.digium.com/svn/asterisk/trunk ................
1474
r259105 | mmichelson | 2010-04-26 16:45:13 -0500 (Mon, 26 Apr
1475
2010) | 9 lines Merged revisions 259104 via svnmerge from
1476
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1477
r259104 | mmichelson | 2010-04-26 16:44:43 -0500 (Mon, 26 Apr
1478
2010) | 3 lines Let compilation succeed warning-free when
1479
DONT_OPTIMIZE is turned off. ........ ................
1480
1481
* main/channel.c, /: Merged revisions 259023 via svnmerge from
1482
https://origsvn.digium.com/svn/asterisk/trunk ................
1483
r259023 | mmichelson | 2010-04-26 16:13:35 -0500 (Mon, 26 Apr
1484
2010) | 19 lines Merged revisions 259018 via svnmerge from
1485
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1486
r259018 | mmichelson | 2010-04-26 16:03:08 -0500 (Mon, 26 Apr
1487
2010) | 13 lines Prevent Newchannel manager events for dummy
1488
channels. No Newchannel manager event will be fired for channels
1489
that are allocated to not match a registered technology type.
1490
Thus bogus channels allocated solely for variable substitution or
1491
CDR operations do not result in a Newchannel event. (closes issue
1492
#16957) Reported by: atis Review:
1493
https://reviewboard.asterisk.org/r/601 ........ ................
1494
1495
2010-04-26 16:00 +0000 [r258935] Leif Madsen <lmadsen@digium.com>
1496
1497
* /, channels/chan_sip.c: Merged revisions 258934 via svnmerge from
1498
https://origsvn.digium.com/svn/asterisk/trunk ........ r258934 |
1499
lmadsen | 2010-04-26 10:59:34 -0500 (Mon, 26 Apr 2010) | 7 lines
1500
Small error in the T.140 RTP port verbose log. (closes issue
1501
#16988) Reported by: frawd Patches: chan_sip_sdp_verbose_fix.diff
1502
uploaded by frawd (license 610) Tested by: russell ........
1503
1504
2010-04-25 18:14 +0000 [r258779] Tilghman Lesher <tlesher@digium.com>
1505
1506
* res/res_monitor.c, /: Merged revisions 258776 via svnmerge from
1507
https://origsvn.digium.com/svn/asterisk/trunk ................
1508
r258776 | tilghman | 2010-04-25 13:12:14 -0500 (Sun, 25 Apr 2010)
1509
| 13 lines Merged revisions 258775 via svnmerge from
1510
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1511
r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010)
1512
| 6 lines When StopMonitor is called, ensure that it will not be
1513
restarted by a channel event. (closes issue #16590) Reported by:
1514
kkm Patches: resmonitor-16590-trunk.239289.diff uploaded by kkm
1515
(license 888) ........ ................
1516
1517
2010-04-22 22:15 +0000 [r258676] Matthew Nicholson <mnicholson@digium.com>
1518
1519
* main/cdr.c, main/channel.c, /, main/features.c: Merged revisions
1520
258671,258675 via svnmerge from
1521
https://origsvn.digium.com/svn/asterisk/trunk ................
1522
r258671 | mnicholson | 2010-04-22 16:57:59 -0500 (Thu, 22 Apr
1523
2010) | 32 lines Merged revisions 193391,258670 via svnmerge from
1524
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1525
r193391 | mnicholson | 2009-05-08 16:01:25 -0500 (Fri, 08 May
1526
2009) | 8 lines Set the proper disposition on originated calls.
1527
(closes issue #14167) Reported by: jpt Patches:
1528
call-file-missing-cdr2.diff uploaded by mnicholson (license 96)
1529
Tested by: dlotina, rmartinez, mnicholson ........ r258670 |
1530
mnicholson | 2010-04-22 16:49:07 -0500 (Thu, 22 Apr 2010) | 11
1531
lines Fix broken CDR behavior. This change allows a CDR record
1532
previously marked with disposition ANSWERED to be set as BUSY or
1533
NO ANSWER. Additionally this change partially reverts r235635 and
1534
does not set the AST_CDR_FLAG_ORIGINATED flag on CDRs generated
1535
from ast_call(). To preserve proper CDR behavior, the
1536
AST_CDR_FLAG_DIALED flag is now cleared from all brige CDRs in
1537
ast_bridge_call(). (closes issue #16797) Reported by:
1538
VarnishedOtter Tested by: mnicholson ........ (closes issue
1539
#16222) Reported by: telles Tested by: mnicholson
1540
................ r258675 | mnicholson | 2010-04-22 17:11:23 -0500
1541
(Thu, 22 Apr 2010) | 2 lines Fix previous commit.
1542
................
1543
1544
2010-04-22 21:58 +0000 [r258516-258672] Russell Bryant <russell@digium.com>
1545
1546
* /, main/event.c: Merged revisions 258632 via svnmerge from
1547
https://origsvn.digium.com/svn/asterisk/trunk For 1.6.2, only
1548
merge the bug fixes, not the unit test. ........ r258632 |
1549
russell | 2010-04-22 16:06:53 -0500 (Thu, 22 Apr 2010) | 22 lines
1550
Add ast_event subscription unit test and fix some ast_event API
1551
bugs. This patch introduces another test in test_event.c that
1552
exercises most of the subscription related ast_event API calls. I
1553
made some minor additions to the existing event allocation test
1554
to increase API coverage by the test code. Finally, I made a list
1555
in a comment of API calls not yet touched by the test module as a
1556
to-do list for future test development. During the development of
1557
this test code, I discovered a number of bugs in the event API.
1558
1) subscriptions to AST_EVENT_ALL were not handled appropriately
1559
in a couple of different places. The API allows a subscription to
1560
all event types, but with IE parameters, just as if it was a
1561
subscription to a specific event type. However, the parameters
1562
were being ignored. This affected ast_event_check_subscriber()
1563
and event distribution to subscribers. 2) Some of the logic in
1564
ast_event_check_subscriber() for checking subscriptions against
1565
query parameters was wrong. Review:
1566
https://reviewboard.asterisk.org/r/617/ ........
1567
1568
* /, doc/tex/channelvariables.tex: Merged revisions 258515 via
1569
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
1570
........ r258515 | russell | 2010-04-22 12:36:34 -0500 (Thu, 22
1571
Apr 2010) | 2 lines Add MEETMEBOOKID from r256019. ........
1572
1573
2010-04-21 22:11 +0000 [r258436] Jeff Peeler <jpeeler@digium.com>
1574
1575
* apps/app_voicemail.c, /: Merged revisions 258433 via svnmerge
1576
from https://origsvn.digium.com/svn/asterisk/trunk
1577
................ r258433 | jpeeler | 2010-04-21 16:56:09 -0500
1578
(Wed, 21 Apr 2010) | 15 lines Merged revisions 258432 via
1579
svnmerge from
1580
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1581
r258432 | jpeeler | 2010-04-21 16:45:36 -0500 (Wed, 21 Apr 2010)
1582
| 8 lines Fix looping forever when no input received in certain
1583
voicemail menu scenarios. Specifically, prompting for an
1584
extension (when leaving or forwarding a message) or when
1585
prompting for a digit (when saving a message or changing
1586
folders). ABE-2122 SWP-1268 ........ ................
1587
1588
2010-04-21 19:44 +0000 [r258384-258386] Leif Madsen <lmadsen@digium.com>
1589
1590
* doc/tex/asterisk.tex: Remove missed line in previous merge.
1591
(issue #17220)
1592
1593
* configure: Forgot to merge the updated configure script. (issue
1594
#17220)
1595
1596
* doc/tex/localchannel.tex, doc/tex/enum.tex, makeopts.in,
1597
doc/tex/asterisk.tex, Makefile, /, doc/tex/Makefile,
1598
configure.ac, doc/tex/phoneprov.tex, doc/tex, doc/tex/ael.tex,
1599
build_tools/prep_tarball: Merged revisions 258351 via svnmerge
1600
from https://origsvn.digium.com/svn/asterisk/trunk ........
1601
r258351 | lmadsen | 2010-04-21 14:18:35 -0500 (Wed, 21 Apr 2010)
1602
| 20 lines Add ability to generate ASCII documentation from the
1603
TeX files. These changes add the ability to run 'make
1604
asterisk.txt' just like the existing 'make asterisk.pdf' commands
1605
to generate a text document from the TeX files we have in the
1606
doc/tex/ directory. I've also updated a few of the .tex files
1607
because they weren't properly escaping certain characters so they
1608
would show up as Unicode characters (like [U+021C]). Made changes
1609
to the configure scripts so it would detect the catdvi program
1610
which is required to convert the .dvi file generated by latex.
1611
I've also added a few lines to the build_tools/prep_tarball
1612
script so that the text documentation gets generated and added to
1613
future tarballs of Asterisk releases. (closes issue #17220)
1614
Reported by: lmadsen Patches: asterisk.txt.patch uploaded by
1615
lmadsen (license 10) asterisk.txt.patch-v4 uploaded by pabelanger
1616
(license 224) Tested by: lmadsen, pabelanger ........
1617
1618
2010-04-21 18:19 +0000 [r258314] David Vossel <dvossel@digium.com>
1619
1620
* /, channels/chan_sip.c: Merged revisions 258305 via svnmerge from
1621
https://origsvn.digium.com/svn/asterisk/trunk ........ r258305 |
1622
dvossel | 2010-04-21 13:13:36 -0500 (Wed, 21 Apr 2010) | 12 lines
1623
fixes issue with double "sip:" in header field This is a clear
1624
mistake in logic. Future discussions about how to avoid having to
1625
handle uri's like this should take place in the future, but this
1626
fix needs to go in for now. (closes issue #15847) Reported by:
1627
ebroad Patches: doublesip.patch uploaded by ebroad (license 878)
1628
........
1629
1630
2010-04-20 19:03 +0000 [r258148-258150] Leif Madsen <lmadsen@digium.com>
1631
1632
* /, configs/cli_aliases.conf.sample: Merged revisions 258149 via
1633
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
1634
........ r258149 | lmadsen | 2010-04-20 14:02:49 -0500 (Tue, 20
1635
Apr 2010) | 1 line Add 'soft hangup' alias per Steve Johnson on
1636
asterisk-users. ........
1637
1638
* configs/extensions.conf.sample, /: Merged revisions 258147 via
1639
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
1640
........ r258147 | lmadsen | 2010-04-20 13:38:39 -0500 (Tue, 20
1641
Apr 2010) | 8 lines Add example dialplan for dialing ISN numbers
1642
(http://www.freenum.org). Minor tweaks and documentation added by
1643
me. (closes issue #17058) Reported by: pprindeville Patches:
1644
freenum.patch#5 uploaded by pprindeville (license 347) Tested by:
1645
lmadsen ........
1646
1647
2010-04-20 18:04 +0000 [r258108] Jeff Peeler <jpeeler@digium.com>
1648
1649
* apps/app_voicemail.c, /: Merged revisions 258065 via svnmerge
1650
from https://origsvn.digium.com/svn/asterisk/trunk
1651
................ r258065 | jpeeler | 2010-04-20 12:06:19 -0500
1652
(Tue, 20 Apr 2010) | 17 lines Merged revisions 258029 via
1653
svnmerge from
1654
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1655
r258029 | jpeeler | 2010-04-20 11:16:33 -0500 (Tue, 20 Apr 2010)
1656
| 11 lines Play correct prompt when voicemail store failure
1657
occurs after attempted forward. If a user's mailbox was full and
1658
a message was attempted to be forwarded to said box, warnings on
1659
the console would indicate failure. However, the played prompt
1660
was that of success (vm-msgsaved). Now storage failure is taken
1661
into account and the correct prompt (vm-mailboxfull) is played
1662
when appropriate. ABE-2123 SWP-1262 ........ ................
1663
1664
2010-04-20 18:02 +0000 [r258107] Leif Madsen <lmadsen@digium.com>
1665
1666
* contrib/scripts/sip-friends.sql, /: Merged revisions 258106 via
1667
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
1668
........ r258106 | lmadsen | 2010-04-20 13:01:28 -0500 (Tue, 20
1669
Apr 2010) | 7 lines Add missing 'useragent' field to
1670
sip-friends.sql file. (closes issue #17171) Reported by: thehar
1671
Patches: sip-friends.patch uploaded by thehar (license 831)
1672
Tested by: pabelanger, thehar ........
1673
1674
2010-04-19 21:58 +0000 [r257948-257950] Jason Parker <jparker@digium.com>
1675
1676
* main/indications.c, /: Merged revisions 257949 via svnmerge from
1677
https://origsvn.digium.com/svn/asterisk/trunk ........ r257949 |
1678
qwell | 2010-04-19 16:57:56 -0500 (Mon, 19 Apr 2010) | 1 line
1679
Change log message to match severity. ........
1680
1681
* main/indications.c, /: Merged revisions 257947 via svnmerge from
1682
https://origsvn.digium.com/svn/asterisk/trunk ........ r257947 |
1683
qwell | 2010-04-19 16:49:30 -0500 (Mon, 19 Apr 2010) | 6 lines
1684
Don't consider a missing indications.conf to be a critical error.
1685
There were many changes in revision 176627 which would avoid the
1686
error that a missing config would have caused. Other than this,
1687
there are no other config files (including asterisk.conf,
1688
surprisingly) that are required. ........
1689
1690
2010-04-19 18:30 +0000 [r257850] Terry Wilson <twilson@digium.com>
1691
1692
* /, main/features.c: Merged revisions 257810 via svnmerge from
1693
https://origsvn.digium.com/svn/asterisk/trunk ........ r257810 |
1694
twilson | 2010-04-19 12:57:41 -0500 (Mon, 19 Apr 2010) | 5 lines
1695
Fix incomplete CDR merge from r195881 Because res/res_features.c
1696
was removed and main/cdr.c added, these changes didn't make it to
1697
trunk and the 1.6.x branches ........
1698
1699
2010-04-18 17:28 +0000 [r257771] Tilghman Lesher <tlesher@digium.com>
1700
1701
* configs/cdr_odbc.conf.sample, /: Merged revisions 257768 via
1702
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
1703
........ r257768 | tilghman | 2010-04-18 12:25:53 -0500 (Sun, 18
1704
Apr 2010) | 2 lines Removing unused configuration parameters
1705
........
1706
1707
2010-04-16 21:47 +0000 [r257740] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
1708
1709
* apps/app_mixmonitor.c, /: Merged revisions 257713 via svnmerge
1710
from https://origsvn.digium.com/svn/asterisk/trunk
1711
................ r257713 | dhubbard | 2010-04-16 16:22:30 -0500
1712
(Fri, 16 Apr 2010) | 28 lines Merged revisions 257686 via
1713
svnmerge from
1714
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1715
r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16 Apr 2010)
1716
| 21 lines Make the mixmonitor thread process audio frames faster
1717
Mantis issue 17078 reports MixMonitor recordings have shorter
1718
durations than the call duration. This was because the mixmonitor
1719
thread was not processing frames from the audiohook fast enough.
1720
The mixmonitor thread would slowly fall behind the most recent
1721
audio frame and when the channel hangs up, the mixmonitor thread
1722
would exit without processing the same number of frames as the
1723
channel; leaving the mixmonitor recording shorter than actual
1724
call duration. This revision fixes this issue by moving the
1725
ast_audiohook_trigger_wait() and the subsequent audiohook.status
1726
check into the block where the ast_audiohook_read_frame()
1727
function returns NULL. (closes issue #17078) Reported by:
1728
geoff2010 Patches: dw-M17078.patch uploaded by dhubbard (license
1729
733) Tested by: dhubbard, geoff2010 Review:
1730
https://reviewboard.asterisk.org/r/611/ ........ ................
1731
1732
2010-04-15 21:34 +0000 [r257510-257597] Tilghman Lesher <tlesher@digium.com>
1733
1734
* include/asterisk/app.h, /, main/app.c: Merged revisions 257560
1735
via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
1736
................ r257560 | tilghman | 2010-04-15 16:26:19 -0500
1737
(Thu, 15 Apr 2010) | 13 lines Merged revisions 257544 via
1738
svnmerge from
1739
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1740
r257544 | tilghman | 2010-04-15 16:23:24 -0500 (Thu, 15 Apr 2010)
1741
| 6 lines Allow application options with arguments to contain
1742
parentheses, through a variety of escaping techniques. Fixes
1743
SWP-1194 (ABE-2143). Review:
1744
https://reviewboard.asterisk.org/r/604/ ........ ................
1745
1746
* /, channels/chan_sip.c: Merged revisions 257493 via svnmerge from
1747
https://origsvn.digium.com/svn/asterisk/trunk ................
1748
r257493 | tilghman | 2010-04-15 15:30:15 -0500 (Thu, 15 Apr 2010)
1749
| 20 lines Merged revisions 257467 via svnmerge from
1750
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1751
r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010)
1752
| 13 lines Don't recreate peer, when responding to a repeated
1753
deregistration attempt. When a reply to a deregistration is lost
1754
in transmit, the client retries the deregistration. Previously,
1755
this would cause a realtime/autocreate peer to be loaded back
1756
into memory, after it had already been correctly purged. Instead,
1757
we just want to resend the reply without loading the peer.
1758
(closes issue #16908) Reported by: kkm Patches:
1759
20100412__issue16908.diff.txt uploaded by tilghman (license 14)
1760
Tested by: kkm ........ ................
1761
1762
2010-04-15 19:42 +0000 [r257344-257428] Leif Madsen <lmadsen@digium.com>
1763
1764
* doc/backtrace.txt: Merged revisions 257427 via svnmerge from
1765
https://origsvn.digium.com/svn/asterisk/trunk ................
1766
r257427 | lmadsen | 2010-04-15 14:41:05 -0500 (Thu, 15 Apr 2010)
1767
| 21 lines Merged revisions 257426 via svnmerge from
1768
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1769
r257426 | lmadsen | 2010-04-15 14:40:33 -0500 (Thu, 15 Apr 2010)
1770
| 13 lines Update backtrace.txt documentation. Update the
1771
backtrace.txt documentation so it conforms to the same layout as
1772
other documents we've been working on recently. Additionally, add
1773
a bunch of new information about gathering backtraces for crashes
1774
and deadlocks, along with ways of verifying your file before
1775
uploading it. Create a couple of one line commands for people to
1776
generate the files we need. (closes issue #17190) Reported by:
1777
lmadsen Patches: backtrace.txt.patch-2 uploaded by lmadsen
1778
(license 10) Tested by: lmadsen, pabelanger ........
1779
................
1780
1781
* doc/backtrace.txt: Merged revisions 257343 via svnmerge from
1782
https://origsvn.digium.com/svn/asterisk/trunk ................
1783
r257343 | lmadsen | 2010-04-15 08:44:38 -0500 (Thu, 15 Apr 2010)
1784
| 9 lines Merged revisions 257342 via svnmerge from
1785
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1786
r257342 | lmadsen | 2010-04-15 08:41:45 -0500 (Thu, 15 Apr 2010)
1787
| 1 line Update address of the bug tracker. ........
1788
................
1789
1790
2010-04-14 23:00 +0000 [r257265] Tilghman Lesher <tlesher@digium.com>
1791
1792
* configs/features.conf.sample, /, main/features.c: Merged
1793
revisions 257262 via svnmerge from
1794
https://origsvn.digium.com/svn/asterisk/trunk ........ r257262 |
1795
tilghman | 2010-04-14 17:57:35 -0500 (Wed, 14 Apr 2010) | 15
1796
lines Yet another issue where the conversion of the application
1797
delimiter to comma caused an issue. Application arguments within
1798
the feature map could possibly contain a comma, which conflicts
1799
with the syntax of the features.conf configuration file. This
1800
patch allows the argument to be wrapped in parentheses or quoted,
1801
to allow the application arguments to be interpreted as a single
1802
configuration parameter. (closes issue #16646) Reported by:
1803
pinga-fogo Patches: 20100414__issue16646.diff.txt uploaded by
1804
tilghman (license 14) Tested by: tilghman Review:
1805
https://reviewboard.asterisk.org/r/547/ ........
1806
1807
2010-04-13 19:20 +0000 [r257210] Tilghman Lesher <tlesher@digium.com>
1808
1809
* /, channels/chan_sip.c: Merged revisions 257191 via svnmerge from
1810
https://origsvn.digium.com/svn/asterisk/trunk ........ r257191 |
1811
tilghman | 2010-04-13 14:17:48 -0500 (Tue, 13 Apr 2010) | 10
1812
lines Also unref the pvt when we delete the provisional keepalive
1813
job. (closes issue #16774) Reported by: kowalma Patches:
1814
20100315__issue16774.diff.txt uploaded by tilghman (license 14)
1815
Tested by: falves11, jamicque Review:
1816
https://reviewboard.asterisk.org/r/591/ ........
1817
1818
2010-04-13 18:43 +0000 [r257184] Matthew Nicholson <mnicholson@digium.com>
1819
1820
* main/manager.c, /, configs/manager.conf.sample: Merged revisions
1821
257146 via svnmerge from
1822
https://origsvn.digium.com/svn/asterisk/trunk ................
1823
r257146 | mnicholson | 2010-04-13 13:10:30 -0500 (Tue, 13 Apr
1824
2010) | 16 lines Merged revisions 257070 via svnmerge from
1825
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1826
r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr
1827
2010) | 9 lines Add an option to restore past broken behavor of
1828
the Events manager action Before r238915, certain values for the
1829
EventMask parameter of the Events action would result in no
1830
response being returned. This patch adds an option to restore
1831
that broken behavior. Also while fixing this bug I discovered
1832
that passing an empty EventMasks parameter would also result in
1833
no response being returned, this has been fixed as well while
1834
being preserved when the broken behavior is requested. (closes
1835
issue #17023) Reported by: nblasgen Review:
1836
https://reviewboard.asterisk.org/r/602/ ........ ................
1837
1838
2010-04-13 16:38 +0000 [r257068] Tilghman Lesher <tlesher@digium.com>
1839
1840
* cdr/cdr_sqlite3_custom.c, /: Merged revisions 257065 via svnmerge
1841
from https://origsvn.digium.com/svn/asterisk/trunk ........
1842
r257065 | tilghman | 2010-04-13 11:33:21 -0500 (Tue, 13 Apr 2010)
1843
| 8 lines Ensure that we can have commas within cdr values.
1844
(closes issue #17001) Reported by: snuffy Patches:
1845
20100412__issue17001.diff.txt uploaded by tilghman (license 14)
1846
Tested by: snuffy ........
1847
1848
2010-04-12 17:30 +0000 [r256822-256902] Leif Madsen <lmadsen@digium.com>
1849
1850
* doc/HOWTO_collect_debug_information.txt (added): Merged revisions
1851
256901 via svnmerge from
1852
https://origsvn.digium.com/svn/asterisk/trunk ................
1853
r256901 | lmadsen | 2010-04-12 12:29:53 -0500 (Mon, 12 Apr 2010)
1854
| 23 lines Merged revisions 256900 via svnmerge from
1855
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
1856
r256900 | lmadsen | 2010-04-12 12:29:26 -0500 (Mon, 12 Apr 2010)
1857
| 15 lines Add How-To document on collecting debugging info for
1858
issues.asterisk.org Paul Belanger has been helping a lot with bug
1859
tracking recently and created this document that we can now point
1860
to when additional debugging information is required. This
1861
document will help those filing issues to know how to get the
1862
information required when filing their issues. This will make
1863
things easier on the developers. Initial text and changes by
1864
pabelanger. Tweaks and editing by myself. (closes issue #17159)
1865
Reported by: pabelanger Patches:
1866
HOWTO_collect_debug_information.txt.patch uploaded by lmadsen
1867
(license 10) Tested by: tzafrir, pabelanger, lmadsen ........
1868
................
1869
1870
* apps/app_voicemail.c, /: Merged revisions 256860 via svnmerge
1871
from https://origsvn.digium.com/svn/asterisk/trunk ........
1872
r256860 | lmadsen | 2010-04-12 11:16:43 -0500 (Mon, 12 Apr 2010)
1873
| 3 lines Remove silly debug message that is not useful. (issue
1874
#17159) ........
1875
1876
* /, main/logger.c: Merged revisions 256821 via svnmerge from
1877
https://origsvn.digium.com/svn/asterisk/trunk ........ r256821 |
1878
lmadsen | 2010-04-12 09:39:37 -0500 (Mon, 12 Apr 2010) | 8 lines
1879
CLI command logger set level auto complete. A simple patch to
1880
enable auto tab complete. (closes issue #17152) Reported by:
1881
pabelanger Patches: 0017152.patch uploaded by pabelanger (license
1882
224) ........
1883
1884
2010-04-08 22:03 +0000 [r256483] Tilghman Lesher <tlesher@digium.com>
1885
1886
* main/app.c: Backport /proc/%d/fd method of closing file
1887
descriptors to 1.6.2.
1888
1889
2010-04-06 19:40 +0000 [r256373] Tilghman Lesher <tlesher@digium.com>
1890
1891
* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
1892
include/asterisk/lock.h: Merged revisions 256370 via svnmerge
1893
from https://origsvn.digium.com/svn/asterisk/trunk ........
1894
r256370 | tilghman | 2010-04-06 14:28:42 -0500 (Tue, 06 Apr 2010)
1895
| 2 lines Mac OS X does not support comparing a mutex to its
1896
initializer. Create a test for this. ........
1897
1898
2010-04-06 18:53 +0000 [r256268-256368] Richard Mudgett <rmudgett@digium.com>
1899
1900
* channels/chan_dahdi.c: CallerID channel DAHDI port FXS are empty
1901
after the first call. The bug is exposed if MFC/R2 support is
1902
built into asterisk (i.e., openr2.h is present in the include
1903
path). Code that unconditionally clears the CallerID name and
1904
number is included. Also fixed a malformed if test in mkintf()
1905
added by issue 15883. Converted the if statement to a switch
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statement for clarity. Regression of the issue 15883 fix. (closes
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issue #16968) Reported by: grecco Patches: issue16968.patch
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uploaded by rmudgett (license 664) (closes issue #16747) Reported
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by: viniciusfontes
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* channels/chan_dahdi.c, /: Merged revisions 256265 via svnmerge
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from https://origsvn.digium.com/svn/asterisk/trunk
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................ r256265 | rmudgett | 2010-04-05 19:39:44 -0500
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(Mon, 05 Apr 2010) | 12 lines Merged revisions 256225 via
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svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
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r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05 Apr 2010)
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| 5 lines DAHDI/PRI call to pri_channel_bridge() not protected by
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PRI lock. SWP-1231 ABE-2163 ........ ................
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2010-05-03 Leif Madsen <lmadsen@digium.com>
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* Asterisk 1.6.2.7 Released
Loggerhead is a web-based interface for Breezy
Version: 2.0.1