[asterisk-commits] simon.perreault: branch group/v6 r89180 - in /team/group/v6/trunk: ./ apps/ b...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Nov 12 08:43:52 CST 2007
Author: simon.perreault
Date: Mon Nov 12 08:43:51 2007
New Revision: 89180
URL: http://svn.digium.com/view/asterisk?view=rev&rev=89180
Log:
Merged revisions 87651,87687,87724,87740,87776,87811,87815,87817,87833,87850-87851,87854,87888-87889,87907,87909,87953,87971,88007-88008,88010,88027,88062,88077,88079,88117,88164-88166,88182-88184,88197,88209,88211-88212,88250,88267,88284,88287,88327,88329,88367-88368,88370-88371,88376,88408-88409,88422,88437,88454,88490,88510,88525,88540,88553,88584,88586-88587,88615,88651-88653,88673,88675,88710,88740,88764,88766-88767,88769-88770,88827,88863,88898,88913,88932,88934-88935,88937,88973-88974,88995,89031,89034,89038,89041,89043-89044,89047-89052,89054-89055,89057,89062,89067-89078,89080-89084,89086,89089,89091-89092,89094,89096,89098,89100,89102,89104,89106-89110,89112-89114,89116-89118,89120-89124,89126,89129-89130,89132,89174-89179 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r87651 | qwell | 2007-10-30 16:30:35 -0400 (Tue, 30 Oct 2007) | 9 lines
Merged revisions 87650 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r87650 | qwell | 2007-10-30 15:29:41 -0500 (Tue, 30 Oct 2007) | 1 line
Only try to clean out h323/ if the h323/Makefile exists.
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r87687 | russell | 2007-10-30 17:22:48 -0400 (Tue, 30 Oct 2007) | 19 lines
Merged revisions 87686 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r87686 | russell | 2007-10-30 16:19:09 -0500 (Tue, 30 Oct 2007) | 11 lines
Merge the changes from team/russell/iax2_poke_fix and iax2-poke-fix-trunk
There was a race condition related to the handling of POKEing peers. Essentially,
a reference to a peer is held by the scheduler when there are pending callbacks,
but the reference count didn't reflect it. So, it was possible for a peer to hit
a reference count of zero and have its destructor begin to be called at the same
time that the scheduler thread ran a POKE related callback. If that happened,
a crash would likely occur.
(closes issue #11082, closes issue #11094)
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r87724 | tilghman | 2007-10-30 18:15:28 -0400 (Tue, 30 Oct 2007) | 5 lines
If no '?' is found in the arguments, don't attempt to continue.
Reported by: blitzrage
Fixed by: tilghman
Closes issue #11111
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r87740 | tilghman | 2007-10-30 19:08:59 -0400 (Tue, 30 Oct 2007) | 13 lines
Merged revisions 87739 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r87739 | tilghman | 2007-10-30 18:02:22 -0500 (Tue, 30 Oct 2007) | 5 lines
Fix for uninitialized mutexes on *BSD
Reported by: ys
Fixed by: ys
Closes issue #11116
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r87776 | murf | 2007-10-31 00:20:40 -0400 (Wed, 31 Oct 2007) | 9 lines
Merged revisions 87775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r87775 | murf | 2007-10-30 21:51:52 -0600 (Tue, 30 Oct 2007) | 1 line
Included some verbage in the check_includes func, to inform the user that included contexts that have no match in the AEL, might be OK, as AEL cannot check in the extensions.conf or the in-memory contexts, as they may not be there at the time of the check.
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r87811 | tilghman | 2007-10-31 11:46:41 -0400 (Wed, 31 Oct 2007) | 2 lines
Optimize pbx_substitute_variables
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r87815 | file | 2007-10-31 12:07:50 -0400 (Wed, 31 Oct 2007) | 2 lines
Remove old whisper remnants from channel.h
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r87817 | tilghman | 2007-10-31 12:13:40 -0400 (Wed, 31 Oct 2007) | 3 lines
Formatting cleanups, remove obsolete contributions (modules no longer in
Asterisk), and obfuscate email addresses enough to stop most spam harvesters.
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r87833 | file | 2007-10-31 13:25:16 -0400 (Wed, 31 Oct 2007) | 2 lines
Add volume adjustment to spy audiohook in app_chanspy.
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r87850 | file | 2007-10-31 13:56:21 -0400 (Wed, 31 Oct 2007) | 2 lines
Restore operation of the option that only writes when the channel is bridged.
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r87851 | file | 2007-10-31 14:03:17 -0400 (Wed, 31 Oct 2007) | 2 lines
Add volume adjustment in.
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r87854 | tilghman | 2007-10-31 14:09:52 -0400 (Wed, 31 Oct 2007) | 10 lines
Merged revisions 87852 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r87852 | tilghman | 2007-10-31 13:03:53 -0500 (Wed, 31 Oct 2007) | 2 lines
Create samples for ALL of the available options in asterisk.conf
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r87888 | murf | 2007-10-31 14:53:13 -0400 (Wed, 31 Oct 2007) | 9 lines
Merged revisions 87849 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r87849 | murf | 2007-10-31 11:49:39 -0600 (Wed, 31 Oct 2007) | 1 line
closes issue #11108 -- where the 'dialplan save' cli command saves a file where the semicolon is not escaped. Fixed this; User also wanted comments to be preserved across dialplan save, but this is impossible at this point in time, because comments are not stored in the dialplan. They are 'compiled' out of extensions.conf. The only way to preserve those comments is to use the config file reader/writer that the GUI uses to allow online user edits. extensions.conf is first and foremost, a config file, and is read in by the normal config-file reading routines. Then, it is processed into a dialplan (context/exten structs). (in the case of trunk, tho, no mods needed to be made -- works OK there -- just make sure you use ',' to sep app args!)
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r87889 | qwell | 2007-10-31 15:24:29 -0400 (Wed, 31 Oct 2007) | 38 lines
More changes to change return values from load_module functions.
(issue #11096)
Patches:
codec_adpcm.c.patch uploaded by moy (license 222)
codec_alaw.c.patch uploaded by moy (license 222)
codec_a_mu.c.patch uploaded by moy (license 222)
codec_g722.c.patch uploaded by moy (license 222)
codec_g726.c.diff uploaded by moy (license 222)
codec_gsm.c.patch uploaded by moy (license 222)
codec_ilbc.c.patch uploaded by moy (license 222)
codec_lpc10.c.patch uploaded by moy (license 222)
codec_speex.c.patch uploaded by moy (license 222)
codec_ulaw.c.patch uploaded by moy (license 222)
codec_zap.c.patch uploaded by moy (license 222)
format_g723.c.patch uploaded by moy (license 222)
format_g726.c.patch uploaded by moy (license 222)
format_g729.c.patch uploaded by moy (license 222)
format_gsm.c.patch uploaded by moy (license 222)
format_h263.c.patch uploaded by moy (license 222)
format_h264.c.patch uploaded by moy (license 222)
format_ilbc.c.patch uploaded by moy (license 222)
format_jpeg.c.patch uploaded by moy (license 222)
format_ogg_vorbis.c.patch uploaded by moy (license 222)
format_pcm.c.patch uploaded by moy (license 222)
format_sln.c.patch uploaded by moy (license 222)
format_vox.c.patch uploaded by moy (license 222)
format_wav.c.patch uploaded by moy (license 222)
format_wav_gsm.c.patch uploaded by moy (license 222)
res_adsi.c.patch uploaded by eliel (license 64)
res_ael_share.c.patch uploaded by eliel (license 64)
res_clioriginate.c.patch uploaded by eliel (license 64)
res_convert.c.patch uploaded by eliel (license 64)
res_indications.c.patch uploaded by eliel (license 64)
res_musiconhold.c.patch uploaded by eliel (license 64)
res_smdi.c.patch uploaded by eliel (license 64)
res_speech.c.patch uploaded by eliel (license 64)
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r87907 | qwell | 2007-10-31 17:18:52 -0400 (Wed, 31 Oct 2007) | 15 lines
Merged revisions 87906 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #11130)
(closes issue #11132)
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r87906 | qwell | 2007-10-31 16:16:20 -0500 (Wed, 31 Oct 2007) | 4 lines
Don't try to allocate memory that we're just going to re-allocate later anyways.
Issues 11130 and 11132, patch by eliel.
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r87909 | qwell | 2007-10-31 17:23:42 -0400 (Wed, 31 Oct 2007) | 13 lines
Merged revisions 87908 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #11131)
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r87908 | qwell | 2007-10-31 16:23:11 -0500 (Wed, 31 Oct 2007) | 4 lines
Make sure we free some allocated memory before returning.
Issue 11131, patch by eliel.
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r87953 | file | 2007-10-31 20:04:43 -0400 (Wed, 31 Oct 2007) | 6 lines
Drop any more references to type in the Exception dialplan function.
(closes issue #11134)
Reported by: blitzrage
Patches:
exception_patch.txt uploaded by blitzrage (license 10)
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r87971 | file | 2007-10-31 21:55:38 -0400 (Wed, 31 Oct 2007) | 12 lines
Merged revisions 87970 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r87970 | file | 2007-10-31 22:53:55 -0300 (Wed, 31 Oct 2007) | 4 lines
If a Zap channel contains a spy or a spy is added take it out of the conference in kernel space and make it go through Asterisk so the spy gets audio from both sides.
(closes issue #10060)
Reported by: mparker
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r88007 | tilghman | 2007-11-01 01:44:40 -0400 (Thu, 01 Nov 2007) | 5 lines
Fix memory leak
Reported by: eliel
Fixed by: tilghman
Closes issue #11136
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r88008 | tilghman | 2007-11-01 02:07:18 -0400 (Thu, 01 Nov 2007) | 5 lines
Janitor: use ast_free to pair calls of ast_malloc and ast_calloc
Reported by: eliel
Patch by: eliel
Closes issue #11135
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r88010 | tilghman | 2007-11-01 02:12:08 -0400 (Thu, 01 Nov 2007) | 5 lines
Conditionally free lock_info->thread_name to avoid a useless warning
Reported by: snuffy
Patch by: snuffy
Closes issue #11125
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r88027 | file | 2007-11-01 09:30:28 -0400 (Thu, 01 Nov 2007) | 10 lines
Merged revisions 88026 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88026 | file | 2007-11-01 10:27:37 -0300 (Thu, 01 Nov 2007) | 2 lines
Fix up commit for my Zap channel with spies in Meetme fix. (thanks Tony Mountifield!)
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r88062 | russell | 2007-11-01 11:18:06 -0400 (Thu, 01 Nov 2007) | 7 lines
Remove another copied source file on "make clean".
(closes issue #11137)
Reported by: IgorG
Patches:
addonclean-87971-1.patch uploaded by IgorG (license 20)
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r88077 | russell | 2007-11-01 11:56:25 -0400 (Thu, 01 Nov 2007) | 8 lines
Change some uses of free() to ast_free(). (No functional differences.)
(closes issue #11138)
Reported by: eliel
Patches:
pbx_dundi.c.patch uploaded by eliel (license 64)
chan_sip.c.patch uploaded by eliel (license 64)
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r88079 | qwell | 2007-11-01 12:22:09 -0400 (Thu, 01 Nov 2007) | 12 lines
Merged revisions 88078 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88078 | qwell | 2007-11-01 11:21:22 -0500 (Thu, 01 Nov 2007) | 4 lines
Make sure we set the poll fds to NULL after free()ing it.
Part of issue 11017, patch by tzafrir.
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r88117 | tilghman | 2007-11-01 13:25:58 -0400 (Thu, 01 Nov 2007) | 10 lines
Merged revisions 88116 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88116 | tilghman | 2007-11-01 12:17:56 -0500 (Thu, 01 Nov 2007) | 2 lines
Add some notes on using valgrind
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r88164 | qwell | 2007-11-01 18:10:33 -0400 (Thu, 01 Nov 2007) | 4 lines
Switch res_jabber to use openssl rather than gnutls.
Closes issue #9972, patch by phsultan. Copied from branch at http://svn.digium.com/svn/asterisk/team/phsultan/res_jabber-openssl/
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r88165 | qwell | 2007-11-01 18:19:56 -0400 (Thu, 01 Nov 2007) | 1 line
Crap, accidentally copied the props. Thanks for pointing this out mvanbaak. The odds are quite high that this will break automerge on every team branch.
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r88166 | murf | 2007-11-01 18:26:51 -0400 (Thu, 01 Nov 2007) | 1 line
This commits the performance mods that give the priority processing engine in the pbx, a 25-30% speed boost. The two updates used, are, first, to merge the ast_exists_extension() and the ast_spawn_extension() where they are called sequentially in a loop in the code, into a slightly upgraded version of ast_spawn_extension(), with a few extra args; and, second, I modified the substitute_variables_helper_full, so it zeroes out the byte after the evaluated string instead of demanding you pre-zero the buffer; I also went thru the code and removed the code that zeroed this buffer before every call to the substitute_variables_helper_full. The first fix provides about a 9% speedup, and the second the rest. These figures come from the 'PIPS' benchmark I describe in blogs, conf. reports, etc.
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r88182 | tilghman | 2007-11-01 18:43:46 -0400 (Thu, 01 Nov 2007) | 3 lines
Convert cdr_odbc to use res_odbc managed connections
Closes issue #10614
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r88183 | tilghman | 2007-11-01 19:26:35 -0400 (Thu, 01 Nov 2007) | 3 lines
Modify WaitExten to include an optional dialtone
Closes issue #10783
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r88184 | qwell | 2007-11-01 19:26:51 -0400 (Thu, 01 Nov 2007) | 1 line
Remove traces of gnutls, since we no longer use/need it.
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r88197 | file | 2007-11-01 23:09:02 -0400 (Thu, 01 Nov 2007) | 2 lines
Restore building under 64-bit platforms.
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r88209 | tilghman | 2007-11-02 08:54:31 -0400 (Fri, 02 Nov 2007) | 5 lines
'h' extension doesn't execute past first priority
Reported by: dimas
Patch by: dimas
Closes bug #11146
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r88211 | tilghman | 2007-11-02 09:10:29 -0400 (Fri, 02 Nov 2007) | 13 lines
Merged revisions 88210 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88210 | tilghman | 2007-11-02 08:03:03 -0500 (Fri, 02 Nov 2007) | 5 lines
Fix build on Solaris
Reported by: snuffy
Patch by: ys
Closes issue #11143
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r88212 | tilghman | 2007-11-02 09:17:48 -0400 (Fri, 02 Nov 2007) | 5 lines
Don't re-cache the filename, but check to see if it already exists
Reported by: jamesgolovich
Patch by: jamesgolovich
Closes issue #11144
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r88250 | tilghman | 2007-11-02 11:36:34 -0400 (Fri, 02 Nov 2007) | 5 lines
Add pbx_lua as a method of doing extensions
Reported by: mnicholson
Patch by: mnicholson
Closes issue #11140
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r88267 | tilghman | 2007-11-02 12:26:31 -0400 (Fri, 02 Nov 2007) | 2 lines
Add a few bytes on LUA
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r88284 | qwell | 2007-11-02 12:54:11 -0400 (Fri, 02 Nov 2007) | 13 lines
Merged revisions 88283 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #11147)
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r88283 | qwell | 2007-11-02 11:51:08 -0500 (Fri, 02 Nov 2007) | 4 lines
We need to make sure to specify a language to ast_fileexists, otherwise it may fail for anything besides en
Issue 11147, fix discovered by both citats and myself (independently), with input from Corydon76
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r88287 | tilghman | 2007-11-02 13:34:33 -0400 (Fri, 02 Nov 2007) | 2 lines
Oops, some dev-mode changes for ISO C90
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r88327 | russell | 2007-11-02 16:13:18 -0400 (Fri, 02 Nov 2007) | 3 lines
Fix replacing the version number when it has a '/' in it, like
SVN-group-chan_unistim-r88326M-/trunk
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r88329 | file | 2007-11-02 16:22:40 -0400 (Fri, 02 Nov 2007) | 14 lines
Merged revisions 88328 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88328 | file | 2007-11-02 17:20:21 -0300 (Fri, 02 Nov 2007) | 6 lines
If an INFO request within a dialog is received with a content length of 0 simply send back a 200 OK. It is valid to do this and the remote side is probably using it to make sure the signalling is still alive.
(closes issue #5747)
Reported by: chandi
Patches:
infofix-81430-1.patch uploaded by IgorG (license 20)
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r88367 | file | 2007-11-02 16:51:53 -0400 (Fri, 02 Nov 2007) | 12 lines
Merged revisions 88366 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88366 | file | 2007-11-02 17:49:45 -0300 (Fri, 02 Nov 2007) | 4 lines
Make subscribecontext behave as advertised. It will now look for the presence of a hint in the given context (be it subscribecontext or context).
(closes issue #10702)
Reported by: slavon
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r88368 | russell | 2007-11-02 16:56:12 -0400 (Fri, 02 Nov 2007) | 13 lines
Merge the code from asterisk/team/group/chan_unistim:
This introduces a new channel driver, chan_unistim, that supports the Unistim
VoIP protocol for Nortel phones. The following models have been confirmed
to work: i2002, i2004 and i2050.
(closes issue #8864)
Reported by: c_hans
Patches:
chan_unistim.patch uploaded by c (license 304)
ustm_no_conf.diff uploaded by junky (license 177)
Tested by: c_hans, dbowerman, math, junky, loloski
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r88370 | russell | 2007-11-02 17:25:55 -0400 (Fri, 02 Nov 2007) | 3 lines
Show the channel unique ID in the "show channel concise" output
(closes issue #11148, requested by falves11, patched by me)
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r88371 | russell | 2007-11-02 17:36:37 -0400 (Fri, 02 Nov 2007) | 2 lines
Don't kill asterisk if extensions.lua is not present.
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r88376 | russell | 2007-11-02 17:39:36 -0400 (Fri, 02 Nov 2007) | 2 lines
propagate the DECLINE return value back to the loader
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r88408 | russell | 2007-11-02 18:08:15 -0400 (Fri, 02 Nov 2007) | 2 lines
Convert the CLI commands to the new format
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r88409 | russell | 2007-11-02 18:36:30 -0400 (Fri, 02 Nov 2007) | 1 line
fix some issues with crashing on unload, when it didn't completely load cleanly
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r88422 | jamesgolovich | 2007-11-03 00:55:06 -0400 (Sat, 03 Nov 2007) | 4 lines
Set CLI command to the correct name. Rev 85460 introduced two 'database show' commands when this one
should have been 'database showkey'
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r88437 | tilghman | 2007-11-03 10:19:47 -0400 (Sat, 03 Nov 2007) | 2 lines
Revert commit #86119. Some users intentionally do not want colorized terminals, so this was a misfeature.
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r88454 | rizzo | 2007-11-04 14:44:31 -0500 (Sun, 04 Nov 2007) | 7 lines
Simplify the implementation and the API for stringfields;
details and examples are in include/asterisk/stringfields.h.
Not applicable to older branches except for 1.4 which will
receive a fix for the routines that free memory pools.
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r88490 | rizzo | 2007-11-04 17:42:54 -0500 (Sun, 04 Nov 2007) | 1 line
block merging of not-applicable patch
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r88510 | file | 2007-11-05 08:11:37 -0500 (Mon, 05 Nov 2007) | 6 lines
Fix memory leaks and deadlocks in chan_unistim.
(closes issue #11158)
Reported by: eliel
Patches:
chan_unistim.c.patch uploaded by eliel (license 64)
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r88525 | rizzo | 2007-11-05 10:15:22 -0500 (Mon, 05 Nov 2007) | 3 lines
remove a cygwin-specific function remap that does not work.
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r88540 | tilghman | 2007-11-05 11:30:51 -0500 (Mon, 05 Nov 2007) | 12 lines
Merged revisions 88539 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88539 | tilghman | 2007-11-05 10:20:13 -0600 (Mon, 05 Nov 2007) | 4 lines
Don't check used pooled connections for connection status, as it will cause issues for prepared queries.
Reported by: Nick Gorham (via -dev list)
Patch by: tilghman
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r88553 | rizzo | 2007-11-05 11:41:39 -0500 (Mon, 05 Nov 2007) | 11 lines
Put extra compiler flags into a variable so they are not repeated
too many times.
On passing, add some comments and fix indentation a bit.
On passing, i suspect that the following pattern is wrong
%.eoo: %.o
but in case it will be fixed in a later commit.
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r88584 | kpfleming | 2007-11-05 12:00:07 -0500 (Mon, 05 Nov 2007) | 2 lines
use a variable name that actually indicates what it is for
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r88586 | qwell | 2007-11-05 12:21:05 -0500 (Mon, 05 Nov 2007) | 13 lines
Merged revisions 88585 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #11163)
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r88585 | qwell | 2007-11-05 11:19:41 -0600 (Mon, 05 Nov 2007) | 4 lines
Make sure we destroy the config structure on configuration failure.
Issue 11163, patch by eliel.
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r88587 | rizzo | 2007-11-05 12:27:02 -0500 (Mon, 05 Nov 2007) | 4 lines
merge two rules with the same right hand;
document a bit what is done here.
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r88615 | rizzo | 2007-11-05 12:44:10 -0500 (Mon, 05 Nov 2007) | 4 lines
print messages when entering/leaving a directory so we know where we are
(sometimes it is obvious, sometimes it is not).
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r88651 | russell | 2007-11-05 12:52:08 -0500 (Mon, 05 Nov 2007) | 13 lines
Merged revisions 88624 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88624 | russell | 2007-11-05 11:46:02 -0600 (Mon, 05 Nov 2007) | 5 lines
Fix up datastore handling in ast_do_masquerade(). The code is intended to move
any channel datastores from the old channel to the new one. However, it did
not use the linked list macros properly to accomplish the task. The existing
code would only work if there was only a single datastore on the old channel.
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r88652 | rizzo | 2007-11-05 13:00:24 -0500 (Mon, 05 Nov 2007) | 3 lines
simplify (hopefully) the printing of $(MAKE) in aligned output.
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r88653 | tilghman | 2007-11-05 13:22:20 -0500 (Mon, 05 Nov 2007) | 2 lines
Change wording to that suggested by MasterYoda
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r88673 | file | 2007-11-05 13:52:12 -0500 (Mon, 05 Nov 2007) | 15 lines
Merged revisions 88671 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88671 | file | 2007-11-05 14:47:13 -0400 (Mon, 05 Nov 2007) | 7 lines
If a SIP channel is put on hold multiple times do not keep incrementing the onHold value.
(closes issue #11085)
Reported by: francesco_r
Tested by: blitzrage
(closes issue #10474)
Reported by: acennami
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r88675 | rizzo | 2007-11-05 14:22:56 -0500 (Mon, 05 Nov 2007) | 7 lines
Cleanup the installation of samples, avoiding repetitions.
I am preserving the behaviour on *.adsi files,
i.e. overwrite anything there without making a backup.
However I am not sure that this is the intended behaviour.
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r88710 | russell | 2007-11-05 15:14:48 -0500 (Mon, 05 Nov 2007) | 28 lines
Merged revisions 88709 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88709 | russell | 2007-11-05 14:11:04 -0600 (Mon, 05 Nov 2007) | 20 lines
Merge the last bit of changes from asterisk/team/russell/readq-1.4
The issue here is that the channel frame readq handling got broken when the
code was converted to use the linked list macros. It caused corruption of the
list head and tail pointers. So, I fixed up the usage of the linked list
macros and in passing, simplified the code. I also documented what the code
is doing, as it was a bit difficult to figure out at first.
This bug showed itself with crashes showing messed up head/tail pointers for
the readq. However, there are a couple of crashes that aren't quite as obvious,
but I think may be related. So, if your bug gets closed by this commit, but
you still have a problem, please reopen or create a new bug report.
(closes issue #10936)
(closes issue #10595)
(closes issue #10368)
(closes issue #11084)
(closes issue #10040)
(closes issue #10840)
........
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r88740 | russell | 2007-11-05 15:44:05 -0500 (Mon, 05 Nov 2007) | 15 lines
Merged revisions 88719 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r88719 | russell | 2007-11-05 14:40:01 -0600 (Mon, 05 Nov 2007) | 7 lines
Merge changes from asterisk/team/kpfleming/SRV-priority-handling
Previously, the SRV record support in Asterisk was broken. There was no
guarantee on what record Asterisk would choose to actually use. This set of
changes improves the situation by ensuring that Asterisk will choose the
highest priority record.
........
................
r88764 | rizzo | 2007-11-05 15:50:26 -0500 (Mon, 05 Nov 2007) | 4 lines
comment out an unused variable. Remove it in a few days
if no problems arise.
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r88766 | russell | 2007-11-05 16:23:32 -0500 (Mon, 05 Nov 2007) | 10 lines
Merged revisions 88765 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88765 | russell | 2007-11-05 15:21:39 -0600 (Mon, 05 Nov 2007) | 2 lines
Fix up some indentation.
........
................
r88767 | rizzo | 2007-11-05 16:27:04 -0500 (Mon, 05 Nov 2007) | 11 lines
Move the last instance of AST_LIBS to the only place it is used,
namely main/Makefile .
I am unclear where decisions on the build environment (CFLAGS,
LDFLAGS, LIBS and so on) should be made - right now they are
split here and there.
As a first step in cleaning up this situation, i am trying to at
least collect all instances of each variable in one place.
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r88769 | russell | 2007-11-05 16:35:51 -0500 (Mon, 05 Nov 2007) | 16 lines
Merged revisions 88768 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88768 | russell | 2007-11-05 15:33:56 -0600 (Mon, 05 Nov 2007) | 8 lines
When traversing the list of channel variables here in transmit_invite(), the
asterisk channel must be locked, as this data may change at any time.
(I have seen numerous reports of crashes related to the handling of channel
variables. There are a couple of issues on the bug tracker related to it,
but it has also been noted on IRC and mailing lists. So, I am finding and
fixing some places where channel variables are handled improperly.)
........
................
r88770 | rizzo | 2007-11-05 16:36:11 -0500 (Mon, 05 Nov 2007) | 4 lines
Move AUDIO_LIBS outside the top level Makefile. This too is used only
in one place.
................
r88827 | mmichelson | 2007-11-05 18:31:00 -0500 (Mon, 05 Nov 2007) | 14 lines
Merged revisions 88826 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88826 | mmichelson | 2007-11-05 17:29:29 -0600 (Mon, 05 Nov 2007) | 6 lines
Reworked deadlock avoidance in __ast_read. Restored audio to
callback agents.
(closes issue #11071, reported by callguy, patched by me, tested by callguy and Ted Brown)
........
................
r88863 | kpfleming | 2007-11-05 21:53:13 -0500 (Mon, 05 Nov 2007) | 10 lines
Merged revisions 88862 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88862 | kpfleming | 2007-11-05 20:52:05 -0600 (Mon, 05 Nov 2007) | 2 lines
update comment to match the state of the code
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r88898 | rizzo | 2007-11-06 02:26:25 -0500 (Tue, 06 Nov 2007) | 2 lines
don't export variables that can be retrieved from makeopts in child subdirs
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r88913 | rizzo | 2007-11-06 03:17:42 -0500 (Tue, 06 Nov 2007) | 4 lines
explain that the host environment must be used to build gentone;
Remove unset variables, they would be misleading.
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r88932 | russell | 2007-11-06 08:57:16 -0500 (Tue, 06 Nov 2007) | 16 lines
Merged revisions 88931 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r88931 | russell | 2007-11-06 07:50:15 -0600 (Tue, 06 Nov 2007) | 8 lines
Remove some checks to see if locks are initialized from the non-DEBUG_THREADS
versions of the lock routines. These are incorrect for a number of reasons:
- It breaks the build on mac.
- If there is a problem with locks not getting initialized, then the proper
fix is to find that place and fix the code so that it does get initialized.
- If additional debug code is needed to help find the problem areas, then this
type of things should _only_ be put in the DEBUG_THREADS wrappers.
........
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r88934 | russell | 2007-11-06 09:08:54 -0500 (Tue, 06 Nov 2007) | 20 lines
Merged revisions 88805 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r88805 | russell | 2007-11-05 16:07:54 -0600 (Mon, 05 Nov 2007) | 12 lines
After seeing crashes related to channel variables, I went looking around at the
ways that channel variables are handled. In general, they were not handled in
a thread-safe way. The channel _must_ be locked when reading or writing from/to
the channel variable list.
What I have done to improve this situation is to make pbx_builtin_setvar_helper()
and friends lock the channel when doing their thing. Asterisk API calls almost
all lock the channel for you as necessary, but this family of functions did not.
(closes issue #10923, reported by atis)
(closes issue #11159, reported by 850t)
........
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r88935 | russell | 2007-11-06 09:11:34 -0500 (Tue, 06 Nov 2007) | 7 lines
Add jitterbuffer support to chan_unistim.
(closes issue #11168)
Reported by: IgorG
Patches:
unistimjb-88863-1.patch uploaded by IgorG (license 20)
................
r88937 | russell | 2007-11-06 09:17:33 -0500 (Tue, 06 Nov 2007) | 2 lines
convert uses of LOG_DEBUG to use ast_debug()
................
r88973 | tilghman | 2007-11-06 10:01:56 -0500 (Tue, 06 Nov 2007) | 2 lines
Set up detection of IP_PKTINFO in autoconf for chan_unistim
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r88974 | file | 2007-11-06 10:37:17 -0500 (Tue, 06 Nov 2007) | 6 lines
Update chan_agent documentation. Change a | to , as that is now the required way.
(closes issue #11167)
Reported by: eliel
Patches:
chan_agent.c.patch uploaded by eliel (license 64)
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r88995 | file | 2007-11-06 11:29:16 -0500 (Tue, 06 Nov 2007) | 14 lines
Merged revisions 88994 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r88994 | file | 2007-11-06 12:24:56 -0400 (Tue, 06 Nov 2007) | 6 lines
Fix improbable but possible memory leaks in chan_zap.
(closes issue #11166)
Reported by: eliel
Patches:
chan_zap.c.patch uploaded by eliel (license 64)
........
................
r89031 | rizzo | 2007-11-06 12:05:13 -0500 (Tue, 06 Nov 2007) | 17 lines
Fix embedding of modules on FreeBSD:
the constructor for the list of modules was run
after the constructors for the embedded modules
(which appended entries to the list).
As a result, the list appeared empty when it was
time to use it.
On linux the order of execution of constructor
was evidently different (it may depend on the
ordering of modules in the ELF file).
This is only a workaround - there may be other
situations where the execution of constructors
causes problems, so if we manage to find a more
general solution this workaround can go away.
................
r89034 | file | 2007-11-06 12:10:03 -0500 (Tue, 06 Nov 2007) | 12 lines
Merged revisions 89032 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89032 | file | 2007-11-06 13:08:05 -0400 (Tue, 06 Nov 2007) | 4 lines
Make it so that if a peer is determined to be unreachable using qualify their devicestate will report back unavailable.
(closes issue #11006)
Reported by: pj
........
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r89038 | russell | 2007-11-06 13:23:36 -0500 (Tue, 06 Nov 2007) | 19 lines
Merged revisions 89037 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89037 | russell | 2007-11-06 12:20:07 -0600 (Tue, 06 Nov 2007) | 11 lines
If someone were to delete the files used by an existing MOH class, and then
issue a reload, further use of that class could result in a crash due to
dividing by zero. This set of changes fixes up some places to prevent this
from happening.
(closes issue #10948)
Reported by: jcomellas
Patches:
res_musiconhold_division_by_zero.patch uploaded by jcomellas (license 282)
Additional changes added by me.
........
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r89041 | qwell | 2007-11-06 13:44:19 -0500 (Tue, 06 Nov 2007) | 4 lines
Allow gtalk and jingle to use TLS connections again.
Closes issue #9972
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r89043 | oej | 2007-11-06 14:04:29 -0500 (Tue, 06 Nov 2007) | 12 lines
Merged revisions 89042 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89042 | oej | 2007-11-06 19:53:37 +0100 (Tis, 06 Nov 2007) | 2 lines
Bug fixes to tdd support in zaptel.
........
(Small changes for trunk)
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r89044 | mmichelson | 2007-11-06 14:04:45 -0500 (Tue, 06 Nov 2007) | 7 lines
"show application <foo>" changes for clarity.
(closes issue #11171, reported and patched by blitzrage)
Many thanks!
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r89047 | qwell | 2007-11-06 14:10:18 -0500 (Tue, 06 Nov 2007) | 12 lines
Merged revisions 89046 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89046 | qwell | 2007-11-06 13:09:30 -0600 (Tue, 06 Nov 2007) | 4 lines
Correctly set the total number of channels from a zaptel transcoder board.
SPD-49, patch by Matthew Nicholson.
........
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r89048 | oej | 2007-11-06 14:10:26 -0500 (Tue, 06 Nov 2007) | 2 lines
Additional TDD changes (preparing for SIP changes - adding TDD support to SIP)
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r89049 | tilghman | 2007-11-06 14:16:02 -0500 (Tue, 06 Nov 2007) | 10 lines
Merged revisions 89045 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89045 | tilghman | 2007-11-06 13:09:06 -0600 (Tue, 06 Nov 2007) | 2 lines
We went to the trouble of creating a method of tracking failed trylocks, then never turned it on (oops).
........
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r89050 | oej | 2007-11-06 14:23:10 -0500 (Tue, 06 Nov 2007) | 2 lines
Formatting. Illegaly using some spare spaces from Russell's space-bucket.
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r89051 | murf | 2007-11-06 14:40:33 -0500 (Tue, 06 Nov 2007) | 1 line
Hoping to avoid a crash in OSX for a problem blitzrage found
................
r89052 | russell | 2007-11-06 14:51:37 -0500 (Tue, 06 Nov 2007) | 4 lines
Fix the memory show allocations CLI command so that it doesn't spew out all
of the current memory allocations when you start Asterisk, when the command's
handler gets called for initialization.
................
r89054 | russell | 2007-11-06 15:22:50 -0500 (Tue, 06 Nov 2007) | 11 lines
Merged revisions 89053 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89053 | russell | 2007-11-06 14:18:49 -0600 (Tue, 06 Nov 2007) | 3 lines
Fix init_classes() so that classes that actually do have files loaded aren't
treated as empty, and immediately destroyed ...
........
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r89055 | mmichelson | 2007-11-06 15:32:49 -0500 (Tue, 06 Nov 2007) | 9 lines
Instead of trying to callback a local channel on a failed attended transfer, call
the device that made the transfer instead. This makes for much smoother calling back
when queues are involved.
(closes issue #11155, reported by IPetrov)
Tremendous thanks to Russell for pulling me out of my block I was having on this one
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r89057 | file | 2007-11-06 15:55:58 -0500 (Tue, 06 Nov 2007) | 4 lines
Remove native bridging check for DTMF based transfers. Thanks to the last batch of RTP changes it is no longer required for the media stream to go through Asterisk if DTMF is going over signalling. It will simply reinvite back as needed.
(closes issue #11172)
Reported by: ibc
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r89062 | murf | 2007-11-06 16:08:38 -0500 (Tue, 06 Nov 2007) | 9 lines
Merged revisions 89036 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89036 | murf | 2007-11-06 10:52:50 -0700 (Tue, 06 Nov 2007) | 1 line
closes issue #8786 - where the [catname](!) and [catname](othercat1,othercat2,...) notation gets dropped across a ConfigUpdate (or any other thing that would cause a config file to be written). While I was at it, I also cleaned up some of the destroy routines to free up comments, which was not being done. Made sure the new struct I introduced is also cleaned up properly at destruction time. My code handles multiple template inclusions. Many thanks to ssokol for his patch, which, while not literally used in the final merge, served as a foundation for the fix.
........
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r89067 | mattf | 2007-11-06 17:01:10 -0500 (Tue, 06 Nov 2007) | 1 line
Add some more locking as well as API update for libss7 for new transport types
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r89068 | mmichelson | 2007-11-06 17:05:56 -0500 (Tue, 06 Nov 2007) | 5 lines
Added CLI and manager commands for changing a queue member's penalty
(closes issue #9374, reported and initially patched by wuwu, intermediate patch by eliel, and final patch by me)
................
r89069 | russell | 2007-11-06 17:15:32 -0500 (Tue, 06 Nov 2007) | 10 lines
Added the S() and L() options to the MeetMe application. These are pretty
much identical to the S() and L() options to Dial(). They let you set
timeouts for the conference, as well as have warning sounds played to
let the caller know how much time is left, and when it is running out.
(closes issue #8030)
Reported by: areski
Patches:
meetme_timeout_timelimit_v2.patch uploaded by areski (license 29)
................
r89070 | mmichelson | 2007-11-06 17:36:55 -0500 (Tue, 06 Nov 2007) | 5 lines
Adding the queue strategy wrandom
(closes issue #10942, reported and patched by julianjm, documentation changes by me)
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r89071 | tilghman | 2007-11-06 17:51:48 -0500 (Tue, 06 Nov 2007) | 5 lines
Commit some cleanups to the format type code.
- Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits.
- Add a native slin16 type, so that 16kHz codecs can translate without losing resolution.
(This doesn't affect anything immediately, until another codec has wb support.)
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r89072 | file | 2007-11-06 18:08:36 -0500 (Tue, 06 Nov 2007) | 4 lines
Fix up some PBX logic that became broken. The code would exit prematurely when it should have been collecting more digits.
(closes issue #11175)
Reported by: pj
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r89073 | russell | 2007-11-06 18:44:39 -0500 (Tue, 06 Nov 2007) | 10 lines
Added the ability to do "meetme concise" with the "meetme" CLI command.
This extends the concise capabilities of this CLI command to include
listing all conferences, instead of an addition to the other sub commands
for the "meetme" command.
(closes issue #11078)
Reported by: jthomas
Patches:
meetme-concise.patch uploaded by jthomas (license 293)
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r89074 | russell | 2007-11-06 19:00:38 -0500 (Tue, 06 Nov 2007) | 9 lines
Print out the channel name as a prefix to the "agi debug" output. This makes
AGI debugging on busy systems much easier.
(closes issue #10730)
Reported by: junky
Patches:
agi_debug_chan.diff uploaded by junky (license 177)
20070923_10730.diff uploaded by mvanbaak (license 7)
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r89075 | mmichelson | 2007-11-06 19:04:30 -0500 (Tue, 06 Nov 2007) | 6 lines
Adding documentation regarding imapfolder, imapgreetings, and greetingsfolder options
in voicemail.conf
(closes issue #11133, reported by selsky, patched by blitzrage)
................
r89076 | russell | 2007-11-06 19:16:03 -0500 (Tue, 06 Nov 2007) | 2 lines
Fix another CLI command so it doesn't run the real code when called for initialization.
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r89077 | tilghman | 2007-11-06 20:08:31 -0500 (Tue, 06 Nov 2007) | 6 lines
Add contributed EAGI proxy, which provides FastAGI functionality for EAGI, while also
buffering the audio stream.
Reported by: devil_slayer
Patch by: devil_slayer
Closes issue #8921
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r89078 | tilghman | 2007-11-06 21:14:40 -0500 (Tue, 06 Nov 2007) | 5 lines
Provide the ability to directly manipulate the TON/NPI bits in the dialstring.
Reported by: thetatag
Patch by: thetatag/stevens/tilghman
Closes issue #5331
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