Kevin P. Fleming [Thu, 27 Jul 2006 22:24:19 +0000 (22:24 +0000)]
block this patch from 1.2, it's already fixed here as part of a larger commit
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38371
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Kevin P. Fleming [Thu, 27 Jul 2006 22:21:46 +0000 (22:21 +0000)]
restore infinite timeout for ast_waitfor() call
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38369
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Kevin P. Fleming [Thu, 27 Jul 2006 22:20:52 +0000 (22:20 +0000)]
various cleanups:
use API call for finding channel by name prefix
code formatting to match guidelines (lost about half the of the indenting)
remove useless automatic variable initializations
don't set the spying channel's read format to SLINEAR when we don't do anything with the voice frames we read from it anyway
use proper option argument checking for volume argument
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38368
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Kevin P. Fleming [Thu, 27 Jul 2006 15:42:49 +0000 (15:42 +0000)]
Merged revisions 38347 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r38347 | kpfleming | 2006-07-27 10:40:03 -0500 (Thu, 27 Jul 2006) | 2 lines
do a better job avoiding translation path teardown/setup when not needed
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38348
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Russell Bryant [Thu, 27 Jul 2006 04:44:49 +0000 (04:44 +0000)]
fix seg fault when the parked call that timed out was the last one in the list
of parked calls (fixes issue #7565)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38330
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Russell Bryant [Thu, 27 Jul 2006 04:28:09 +0000 (04:28 +0000)]
Merged revisions 38328 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r38328 | russell | 2006-07-27 00:25:41 -0400 (Thu, 27 Jul 2006) | 4 lines
Fix crash when using the "regexten" option with MALLOC_DEBUG enabled. This was
not reported in the bug tracker but the same bug has been demonstrated in other
places in the code.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38329
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Kevin P. Fleming [Thu, 27 Jul 2006 02:46:22 +0000 (02:46 +0000)]
Merged revisions 38310 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r38310 | kpfleming | 2006-07-26 21:43:49 -0500 (Wed, 26 Jul 2006) | 2 lines
don't do useless translation destroy/build when the channel is already in the correct format
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38311
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Russell Bryant [Thu, 27 Jul 2006 02:00:20 +0000 (02:00 +0000)]
Merged revisions 38288 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r38288 | russell | 2006-07-26 21:58:41 -0400 (Wed, 26 Jul 2006) | 3 lines
fix a crash when MALLOC_DEBUG is enabled and the regexten is enabled. The crash
would occur when the extension got removed. (fixes issue #7484)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38291
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Russell Bryant [Thu, 27 Jul 2006 01:49:57 +0000 (01:49 +0000)]
fix crashes with MALLOC_DEBUG enabled that were a result of my recent thread
storage changes (fixes issue #7595)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38286
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Russell Bryant [Thu, 27 Jul 2006 01:29:57 +0000 (01:29 +0000)]
fix some breakage of ast_cli() that resulted in seg faults on Josh's machine.
I'm not sure why this never caused problems for me ...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38283
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Russell Bryant [Wed, 26 Jul 2006 23:12:47 +0000 (23:12 +0000)]
fix setting the CFLAGS for building codec libs so that they are built with
astmm support and astmm doesn't get really upset and complain that it is being
asked to free memory that was never allocated
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38271
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Tilghman Lesher [Wed, 26 Jul 2006 19:59:25 +0000 (19:59 +0000)]
Typo found on the -dev list
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38258
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Tilghman Lesher [Wed, 26 Jul 2006 17:43:17 +0000 (17:43 +0000)]
Only unlock these if they were locked on entry
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38250
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Joshua Colp [Wed, 26 Jul 2006 15:27:45 +0000 (15:27 +0000)]
Merged revisions 38234 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r38234 | file | 2006-07-26 11:26:06 -0400 (Wed, 26 Jul 2006) | 2 lines
Put default callerid into contact when the one specified is either NULL or has a zero string length. (issue #7590 reported by key2)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38235
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Russell Bryant [Wed, 26 Jul 2006 07:48:48 +0000 (07:48 +0000)]
merge the changes from my ast_cli_tls branch ...
Instead of having a static buffer size of 16k, start out with 256 bytes and
increase the size of the buffer as needed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38223
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Russell Bryant [Tue, 25 Jul 2006 19:51:31 +0000 (19:51 +0000)]
This exact deadlock situation that I observed can't happen in trunk due to the
recent hold changes so that MOH is not started on the bridged channel directly.
However, the change is still not a bad idea.
Merged revisions 38200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r38200 | russell | 2006-07-25 15:43:38 -0400 (Tue, 25 Jul 2006) | 6 lines
This resolves a deadlock that a tech support customer was getting frequently
when his users would answer call waiting. If another thread is currently
holding the zt_pvt lock for the first channel, unlock both channels and let
asterisk retry the native bridge, just like what is done for the second channel
directly below these changes.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38201
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Steve Murphy [Mon, 24 Jul 2006 20:58:27 +0000 (20:58 +0000)]
Added a small fix to exclude the AMD optimized assembly file from being included in the compile.
I don't particular like the approach, which basically tries to exclude the file based on
NOT being in a list of arches. This, IMHO, is backwards, it should be a list of AMD arches
instead. "Better to light a candle, than to curse the darkness", is apropo here. But since I don't
have an exhaustive list of what those arches could be, I leave as-is for now.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38187
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Steve Murphy [Mon, 24 Jul 2006 17:24:47 +0000 (17:24 +0000)]
blocked rev 38167 on the 1.2 branch from trunk, because it's
coded differently on trunk, would conflict and have to be hand-merged
anyway.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38168
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Joshua Colp [Mon, 24 Jul 2006 15:47:59 +0000 (15:47 +0000)]
Start out with cleared RTP payload structures instead of defaults. This should prevent issues where if a stream (audio/stream) is not present and it's RTP payload structure is combined with the overall capability then the capability would be every codec that Asterisk supports.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38158
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Luigi Rizzo [Mon, 24 Jul 2006 11:46:27 +0000 (11:46 +0000)]
move middle-of-the-block declaration at the top.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38149
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Luigi Rizzo [Mon, 24 Jul 2006 11:45:06 +0000 (11:45 +0000)]
fix uninitialized variable
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38148
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Joshua Colp [Mon, 24 Jul 2006 03:42:27 +0000 (03:42 +0000)]
Only deal with getting the supported payloads on audio if an audio RTP stream exists
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38139
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Russell Bryant [Sun, 23 Jul 2006 15:21:44 +0000 (15:21 +0000)]
ast_malloc is sufficient here, since snprintf is called on the buffer
immediately after it gets allocated, which will automatically NULL terminate
the string
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38129
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Russell Bryant [Sun, 23 Jul 2006 15:19:16 +0000 (15:19 +0000)]
Merge team/russell/ast_cli_tls into the trunk.
This improves the performance of ast_cli() by not doing a heap memory
allocation and deallocation every single time the function is called. Instead,
a thread-specific buffer is allocatted the first time the function is called
and automatically free'd when the thread exits. Also note that this buffer will
only be allocatted in threads that actually call this function, which is
probably only the threads spawned to service connected asterisk consoles.
This does introduce a new limitation on the maximum length of the resulting
string from the arguments passed to ast_cli. Previously there was no limit
since it was just allocating a buffer big enough every time the function was
called. The current buffer size is 16kB. If there is ever a case where we want
to print more than 16k characters in a single call to ast_cli(), this will have
to be increased.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38127
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Russell Bryant [Sun, 23 Jul 2006 05:20:59 +0000 (05:20 +0000)]
fix a seg fault in this application if no context paramater is given
(issue #7571)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38118
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Russell Bryant [Sun, 23 Jul 2006 05:06:47 +0000 (05:06 +0000)]
resolve another XXX comment by implementing proper handling of control frames
in ast_write(), which is to call the channel's indicate function if it exists
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38117
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Russell Bryant [Sun, 23 Jul 2006 04:58:56 +0000 (04:58 +0000)]
take the advice of an XXX comment and use an atomic operation to decrement a
variable. Also, change a use of strdup to ast_strdup and remove a duplicated
error message.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38108
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Russell Bryant [Sun, 23 Jul 2006 03:29:19 +0000 (03:29 +0000)]
remove a stray newline
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38099
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Russell Bryant [Sun, 23 Jul 2006 02:50:25 +0000 (02:50 +0000)]
update to mention svn instead of cvs
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38089
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Russell Bryant [Sun, 23 Jul 2006 02:41:02 +0000 (02:41 +0000)]
various cleanups regarding coding guidelines issues
- malloc to ast_malloc
- malloc + memset to ast_calloc
- sizeof(struct foo) to sizeof(*bar)
- remove indentation of the entire body of a function by returning immediately
on an allocation failure
(issue #7581, tempest1)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38088
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Russell Bryant [Sat, 22 Jul 2006 00:38:30 +0000 (00:38 +0000)]
ast_pbx_outgoing_cdr_failed() is not used outside of pbx.c so make it static
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38078
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Russell Bryant [Sat, 22 Jul 2006 00:33:56 +0000 (00:33 +0000)]
use ast_atomic_fetchadd_int in chan_zap, sip, and iax2 for usecount handling
and fix a couple little things in passing
- usecnt was not initialized in chan_iax2
- ast_update_use_count() was not called after incrementing the count in chan_sip
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38077
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Russell Bryant [Sat, 22 Jul 2006 00:08:21 +0000 (00:08 +0000)]
remove an XXX comment and document that ast_autoservice_start() will return -1
if the channel is already in the autoservice list.
Why is this a valid case to return -1, you ask? Well, there should never be
any code where it is not clear if the channel is in autoservice or not because
trying to read frames from a channel that is in the autoservice list will lead
to bad results because more than one thread will be waiting on frames to arrive
on the channel and then trying to read them.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38076
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Russell Bryant [Fri, 21 Jul 2006 23:58:14 +0000 (23:58 +0000)]
slightly simplify a loop and replace an open coded version of ast_skip_blanks
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38068
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Russell Bryant [Fri, 21 Jul 2006 22:36:25 +0000 (22:36 +0000)]
make ast_state2str thread safe by using thread local storage instead of a
static buffer for storing the result when the state value is unknown
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38060
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BJ Weschke [Fri, 21 Jul 2006 19:10:58 +0000 (19:10 +0000)]
This corrects the crash condition present in #7575, but I'm not really sure if it's the "right" fix. Please review and make any adjustments you see necessary.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38051
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Russell Bryant [Fri, 21 Jul 2006 17:31:28 +0000 (17:31 +0000)]
Merge a new implementation of ast_inet_ntoa, our thread safe replacement for
inet_ntoa, which uses thread specific data (aka thread local storage) instead
of stack allocatted buffers to store the result.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38042
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Joshua Colp [Thu, 20 Jul 2006 18:42:22 +0000 (18:42 +0000)]
Only bitmaskify the RTP payload structure for video if an RTP structure exists for it... otherwise the default values will cause codec combination madness
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38030
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Russell Bryant [Thu, 20 Jul 2006 16:18:07 +0000 (16:18 +0000)]
add a verbose message to the AGI command, STREAM FILE, similar to the
verbose messages when using Playback, Background, or the GET DATA command
(issue #7297, softins)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38021
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North Antara [Wed, 19 Jul 2006 22:48:00 +0000 (22:48 +0000)]
Remove "initialization from incompatible pointer type" warnings.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38010
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North Antara [Wed, 19 Jul 2006 22:40:36 +0000 (22:40 +0000)]
Fix a few doxygen warnings.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38009
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Russell Bryant [Wed, 19 Jul 2006 21:52:59 +0000 (21:52 +0000)]
add menuselect_stub.c to deps list
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38000
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Russell Bryant [Wed, 19 Jul 2006 21:07:14 +0000 (21:07 +0000)]
add notes on the changes to music on hold handling
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37999
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Kevin P. Fleming [Wed, 19 Jul 2006 20:44:39 +0000 (20:44 +0000)]
merge Russell's 'hold_handling' branch, finally implementing music-on-hold handling the way it was decided at AstriDevCon Europe 2006 (and the way people really want it to be)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37988
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Kevin P. Fleming [Wed, 19 Jul 2006 18:50:55 +0000 (18:50 +0000)]
make installation of French and Spanish sounds work properly
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37974
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Kevin P. Fleming [Wed, 19 Jul 2006 18:33:14 +0000 (18:33 +0000)]
use new sound file version
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37973
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Mark Spencer [Wed, 19 Jul 2006 17:26:47 +0000 (17:26 +0000)]
Fix some reinitialization of prev!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37953
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Kevin P. Fleming [Wed, 19 Jul 2006 17:21:33 +0000 (17:21 +0000)]
restore buildability
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37952
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Kevin P. Fleming [Wed, 19 Jul 2006 17:16:25 +0000 (17:16 +0000)]
Merged revisions 37949 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r37949 | kpfleming | 2006-07-19 12:10:10 -0500 (Wed, 19 Jul 2006) | 2 lines
ensure that global 'maxauthreq' is reset to zero during 'reload'
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37951
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Kevin P. Fleming [Wed, 19 Jul 2006 17:16:08 +0000 (17:16 +0000)]
restore buildability (bad marko!)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37950
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Kevin P. Fleming [Wed, 19 Jul 2006 13:50:19 +0000 (13:50 +0000)]
revert this morning's incorrect indentation change
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37937
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Mark Spencer [Wed, 19 Jul 2006 13:28:38 +0000 (13:28 +0000)]
First pass at in-place file manipulation via manager
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37936
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Steve Murphy [Wed, 19 Jul 2006 02:55:24 +0000 (02:55 +0000)]
committing a fix for colons in goto arguments and suppressing operator messages in globals assignments.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37925
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Luigi Rizzo [Tue, 18 Jul 2006 21:19:59 +0000 (21:19 +0000)]
move variable declarations to the beginning of a block.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37907
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Mark Spencer [Tue, 18 Jul 2006 20:26:38 +0000 (20:26 +0000)]
Eliminate file descriptor leak in built in HTTP server
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37896
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Joshua Colp [Tue, 18 Jul 2006 16:22:26 +0000 (16:22 +0000)]
Expand speech API so that the developer can interact with the engine more directly and use specific functions of the connector even if a generic API call is not available
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37881
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Russell Bryant [Tue, 18 Jul 2006 00:43:57 +0000 (00:43 +0000)]
Merged revisions 37856 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r37856 | russell | 2006-07-17 20:41:47 -0400 (Mon, 17 Jul 2006) | 2 lines
don't crash if the frame has no data, but has a src
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37857
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Russell Bryant [Mon, 17 Jul 2006 23:31:24 +0000 (23:31 +0000)]
if asked to duplicate a frame that has no data, don't set the frame's data
pointer past the end of the buffer allocated for the new frame
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37830
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Russell Bryant [Mon, 17 Jul 2006 23:26:22 +0000 (23:26 +0000)]
Blocked revisions 37828 via svnmerge
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r37828 | russell | 2006-07-17 19:25:33 -0400 (Mon, 17 Jul 2006) | 3 lines
if asked to duplicate a frame that has no data, don't set the frame's data
pointer past the end of the allocatted buffer for the new frame
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37829
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Tilghman Lesher [Mon, 17 Jul 2006 22:39:02 +0000 (22:39 +0000)]
Blocked revisions 37808 via svnmerge
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r37808 | tilghman | 2006-07-17 17:36:56 -0500 (Mon, 17 Jul 2006) | 2 lines
Backport buffer increase to 1.2
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37809
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Tilghman Lesher [Mon, 17 Jul 2006 16:31:43 +0000 (16:31 +0000)]
H.263 frames can apparently be larger than was originally coded.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37785
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Tilghman Lesher [Mon, 17 Jul 2006 15:56:16 +0000 (15:56 +0000)]
Merged revisions 37765 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r37765 | tilghman | 2006-07-17 10:52:15 -0500 (Mon, 17 Jul 2006) | 2 lines
Overflow bad
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37766
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Mark Spencer [Mon, 17 Jul 2006 02:17:38 +0000 (02:17 +0000)]
Make app_meetme's do_invite more generic.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37755
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Luigi Rizzo [Sun, 16 Jul 2006 22:24:39 +0000 (22:24 +0000)]
use a couple of variables to remove duplicate definitions
of the various rules for the NOISY/!NOISY cases.
Approved by: kevin
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37744
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Russell Bryant [Sun, 16 Jul 2006 19:36:29 +0000 (19:36 +0000)]
add a QueueLog application that allows for custom entries in the queue_log
(issue #7368, by stevedavies with some modifications)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37734
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Russell Bryant [Sun, 16 Jul 2006 19:35:09 +0000 (19:35 +0000)]
malloc + memset to ast_calloc
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Luigi Rizzo [Sun, 16 Jul 2006 13:01:48 +0000 (13:01 +0000)]
typo LDFAGS -> LDFLLAGS
detected by: noisy build on FreeBSD
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Tilghman Lesher [Sat, 15 Jul 2006 23:35:21 +0000 (23:35 +0000)]
Merged revisions 37691 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r37691 | tilghman | 2006-07-15 18:29:28 -0500 (Sat, 15 Jul 2006) | 4 lines
Bug 7513 - ensure that each time we do a query, the results are returned in the
same logical order, so that when we iterate over the list, we get all results,
not some results repeated, due to insufficient sorting.
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North Antara [Sat, 15 Jul 2006 21:56:58 +0000 (21:56 +0000)]
case consistency in DISA (or is it "disa"?)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37682
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Kevin P. Fleming [Sat, 15 Jul 2006 21:41:40 +0000 (21:41 +0000)]
update configure script with new curl version test
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37680
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Tilghman Lesher [Sat, 15 Jul 2006 21:39:26 +0000 (21:39 +0000)]
Move curl version test to autoconf script
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Kevin P. Fleming [Sat, 15 Jul 2006 20:12:28 +0000 (20:12 +0000)]
add comments on what special variables are for
put library path into proper variable for *BSD
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37670
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North Antara [Sat, 15 Jul 2006 20:02:20 +0000 (20:02 +0000)]
oops?
This should've been 5 chars..
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Kevin P. Fleming [Sat, 15 Jul 2006 19:59:35 +0000 (19:59 +0000)]
more Makefile cleanup and consistency stuff
don't reuse LIBS variable from top-level Makefile (oops)
build Asterisk binary after subdirs (preparing for embedded modules)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37661
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Kevin P. Fleming [Sat, 15 Jul 2006 16:40:05 +0000 (16:40 +0000)]
use LDFLAGS and LIBS properly, and allow dependencies to provide LDFLAGS if needed (although none do today)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37653
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Kevin P. Fleming [Sat, 15 Jul 2006 16:39:17 +0000 (16:39 +0000)]
update to autoconf 2.60 version
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37652
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North Antara [Fri, 14 Jul 2006 21:49:40 +0000 (21:49 +0000)]
config sample for the previous, regarding ADSI
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37644
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North Antara [Fri, 14 Jul 2006 21:49:20 +0000 (21:49 +0000)]
There was an issue with ADSI and iax2, where on an iaxy with a normal phone connected,
in vm_authenticate, it was try to start ADSI on the channel, and it WOULD because it
was "supported", according to the iaxy. There is now a config option (adsi=yes) for
this, which defaults to no. (config sample coming shortly)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37643
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Russell Bryant [Fri, 14 Jul 2006 21:02:55 +0000 (21:02 +0000)]
add missing unregistration of a manager action on module unload
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37638
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North Antara [Fri, 14 Jul 2006 20:14:38 +0000 (20:14 +0000)]
Rename a couple variables, to be consistent with the rest of the functions
Abstract out some common code into a single function
With the recent scheduler "issues", it pointed out a few things I might have been missing,
so I added some rudimentary vrtp and rtcp stuff
General cleanup...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37629
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Tilghman Lesher [Fri, 14 Jul 2006 13:34:25 +0000 (13:34 +0000)]
Merged revisions 37612 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r37612 | tilghman | 2006-07-14 08:31:11 -0500 (Fri, 14 Jul 2006) | 2 lines
Bug 7526 - previous commit broke app_sms
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Tilghman Lesher [Fri, 14 Jul 2006 06:03:41 +0000 (06:03 +0000)]
Whitespace fixes
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Tilghman Lesher [Fri, 14 Jul 2006 05:42:06 +0000 (05:42 +0000)]
Whitespace fix
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Russell Bryant [Thu, 13 Jul 2006 22:03:47 +0000 (22:03 +0000)]
fix some typos in the conversion from the old SOONER macro to ast_tvcmp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37586
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Kevin P. Fleming [Thu, 13 Jul 2006 21:23:11 +0000 (21:23 +0000)]
Merged revisions 37571 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r37571 | kpfleming | 2006-07-13 16:22:11 -0500 (Thu, 13 Jul 2006) | 2 lines
don't fail/abort if the message category sound file cannot be played, just generate a warning message and continue message playback
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37572
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Kevin P. Fleming [Thu, 13 Jul 2006 20:39:34 +0000 (20:39 +0000)]
swap the G726-32 format numbers, so that IAX2 connections with prior versions of Asterisk will still work properly
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37565
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Kevin P. Fleming [Thu, 13 Jul 2006 20:35:41 +0000 (20:35 +0000)]
actually make the non-standard G726-32 behavior available for SIP clients
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Tilghman Lesher [Thu, 13 Jul 2006 20:14:27 +0000 (20:14 +0000)]
Whitespace fix
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Russell Bryant [Thu, 13 Jul 2006 18:45:21 +0000 (18:45 +0000)]
Blocked revisions 37546 via svnmerge
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r37546 | russell | 2006-07-13 14:44:17 -0400 (Thu, 13 Jul 2006) | 3 lines
yeah, ummm... This frame pointer should not be static. This situation only
exists in 1.2 (pointed out by Constantine Filin on the asterisk-dev mailing list)
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Kevin P. Fleming [Thu, 13 Jul 2006 16:47:20 +0000 (16:47 +0000)]
Merged revisions 37531 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r37531 | kpfleming | 2006-07-13 11:44:23 -0500 (Thu, 13 Jul 2006) | 2 lines
report address of peer trying to subscribe to unknown hint
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Tilghman Lesher [Thu, 13 Jul 2006 15:47:30 +0000 (15:47 +0000)]
Merged revisions 37516 via svnmerge from
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r37516 | tilghman | 2006-07-13 10:45:33 -0500 (Thu, 13 Jul 2006) | 2 lines
Bug 7532 - Typo in enum example
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Kevin P. Fleming [Thu, 13 Jul 2006 15:37:56 +0000 (15:37 +0000)]
only compare the queue entry's max penalty to the member's penalty when the queue entry actually has a non-zero penalty
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37515
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Kevin P. Fleming [Thu, 13 Jul 2006 15:20:02 +0000 (15:20 +0000)]
Merged revisions 37441-37442 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r37441 | kpfleming | 2006-07-12 10:46:56 -0500 (Wed, 12 Jul 2006) | 3 lines
fix a case where ast_lock_path() could leave a randomly-named lock file hanging around
make ast_unlock_path actually report when unlocking fails
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r37442 | kpfleming | 2006-07-12 10:53:53 -0500 (Wed, 12 Jul 2006) | 2 lines
fix a weird case where a lock file could be left (but would happen almost never)
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Christian Richter [Thu, 13 Jul 2006 14:13:24 +0000 (14:13 +0000)]
added even more statefulness for sending out disconnect/release/release_complete messages. added support for incoming presentation/screening. fixed a bug that we generate TONE_EVENTS on hanguptone_indicatem, which caused asterisk to write blocking thread messages. added nodialtone option to prevent dialtone for always_immediate
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37508
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Kevin P. Fleming [Thu, 13 Jul 2006 01:38:47 +0000 (01:38 +0000)]
allow users of RTP to use G726-32 AAL2 packing even when RFC3551 packing has been requested (Sipura/Grandstream ATAs and others will need this)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37501
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Kevin P. Fleming [Wed, 12 Jul 2006 22:42:13 +0000 (22:42 +0000)]
G726-32 changes:
split support for G726-32 into RFC3551 and AAL2 packing orders, since both are in use
change "G726-32" to be RFC3551 packing order, in spite of devices that use AAL2 order with this MIME type
add ability to directly transcode between packing orders
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37494
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Russell Bryant [Wed, 12 Jul 2006 22:10:59 +0000 (22:10 +0000)]
fix some more app_voicemail breakage
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Olle Johansson [Wed, 12 Jul 2006 20:56:49 +0000 (20:56 +0000)]
- Change filename to current file name
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Russell Bryant [Wed, 12 Jul 2006 20:35:14 +0000 (20:35 +0000)]
fix the build options for app_voicemail
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37484
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Russell Bryant [Wed, 12 Jul 2006 20:27:46 +0000 (20:27 +0000)]
revert this change for now since it isn't working correctly and there are more
important issues to work on at the moment ...
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