[svn-commits] tilghman: branch phsultan/gmail-voice-video r213969 - in /team/phsultan/gmail...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Aug 25 01:20:00 CDT 2009
Author: tilghman
Date: Tue Aug 25 01:19:29 2009
New Revision: 213969
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=213969
Log:
Merged revisions 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via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r186058 | tilghman | 2009-04-02 12:04:40 -0500 (Thu, 02 Apr 2009) | 8 lines
Blocked revisions 186057 via svnmerge
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r186057 | tilghman | 2009-04-02 12:03:59 -0500 (Thu, 02 Apr 2009) | 2 lines
Avoid multiple warning messages in SIP, due to this column not existing
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r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009) | 16 lines
Merged revisions 186059 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186059 | tilghman | 2009-04-02 12:09:13 -0500 (Thu, 02 Apr 2009) | 9 lines
Merged revisions 186056 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) | 2 lines
Fix for AST-2009-003
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r186078 | file | 2009-04-02 12:20:52 -0500 (Thu, 02 Apr 2009) | 10 lines
Merge in the RTP engine API.
This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.
Review: http://reviewboard.digium.com/r/209/
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r186101 | kpfleming | 2009-04-02 12:26:07 -0500 (Thu, 02 Apr 2009) | 9 lines
Merged revisions 186081 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02 Apr 2009) | 3 lines
ensure that the buffer passed to DAHDI_SET_BUFINFO is fully initialized
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r186175 | mmichelson | 2009-04-02 16:56:21 -0500 (Thu, 02 Apr 2009) | 11 lines
Merged revisions 186174 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr 2009) | 5 lines
Fix instructions in one-step parking comment to make more sense.
Changed a capital K to a lowercase k.
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r186230 | russell | 2009-04-02 21:03:48 -0500 (Thu, 02 Apr 2009) | 29 lines
Merged revisions 186229 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009) | 21 lines
Fix a memory leak in cdr_radius.
I came across this while doing some testing of my ast_channel_ao2 branch.
After running a test overnight that generated over 5 million calls, Asterisk
had taken up about 1 GB of my system memory. So, I re-ran the test with
MALLOC_DEBUG turned on. However, it showed no leaks in Asterisk during the
test, even though Asterisk was still consuming it somehow.
Instead, I turned to valgrind, which when run with --leak-check=full, told
me exactly where the leak came from, which was from allocations inside the
radiusclient-ng library. This explains why MALLOC_DEBUG did not report it.
After a bit of analysis, I found that we were leaking a little bit of memory
every time a CDR record was passed to cdr_radius.
I don't actually have a radius server set up to receive CDR records. However,
I always have my development systems compile and install all modules. In
addition to making sure there are not build errors across modules, always
loading modules helps find bugs like this, too, so it is strongly recommend for
all developers.
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r186286 | mmichelson | 2009-04-03 09:32:05 -0500 (Fri, 03 Apr 2009) | 20 lines
Fix the ability to retrieve voicemail messages from IMAP.
A recent change made interactive vm_states no longer get
added to the list of vm_states and instead get stored in
thread-local storage.
In trunk and all the 1.6.X branches, the problem is that
when we search for messages in a voicemail box, we would
attempt to update the appropriate vm_state struct by directly
searching in the list of vm_states instead of using the
get_vm_state_by_imap_user function. This meant we could not
find the interactive vm_state that we wanted.
(closes issue #14685)
Reported by: BlargMaN
Patches:
14685.patch uploaded by mmichelson (license 60)
Tested by: BlargMaN, qualleyiv, mmichelson
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r186297 | tilghman | 2009-04-03 10:18:28 -0500 (Fri, 03 Apr 2009) | 4 lines
Compatibility fix for glibc 2.4
(Closes issue #14820)
Reported by: phsultan
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r186321 | file | 2009-04-03 10:52:50 -0500 (Fri, 03 Apr 2009) | 12 lines
Merged revisions 186320 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5 lines
Fix a problem with the crypto variable definitions not actually being defined properly.
(closes issue #14804)
Reported by: jvandal
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r186379 | dvossel | 2009-04-03 11:29:47 -0500 (Fri, 03 Apr 2009) | 6 lines
audio_audiohook_write_list() did not correctly update sample size after ast_translate.
audio_audiohook_write_list() did not take into account that the sample size may change after translation depending on if the original frame is is 8khz or 16khz. the sample size is now updated after translating to reflect this possibility. This caused the audio on the receiving end to sound terrible. Thanks to jcolp and mmichelson for helping me work this out.
(issue AST-197)
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r186382 | file | 2009-04-03 11:47:27 -0500 (Fri, 03 Apr 2009) | 11 lines
Add better support for relaying success or failure of the ast_transfer() API call.
This API call now waits for a special frame from the underlying channel driver to
indicate success or failure. This allows the return value to truly convey whether
the transfer worked or not. In the case of the Transfer() dialplan application this
means the value of the TRANSFERSTATUS dialplan variable is actually true.
(closes issue #12713)
Reported by: davidw
Tested by: file
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r186444 | tilghman | 2009-04-03 14:30:34 -0500 (Fri, 03 Apr 2009) | 14 lines
Merged revisions 186415 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009) | 7 lines
Distinguish in a sent email between simple sends and forwards.
(closes issue #11678)
Reported by: jamessan
Patches:
20090330__bug11678.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman, lmadsen
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r186447 | tilghman | 2009-04-03 14:59:55 -0500 (Fri, 03 Apr 2009) | 9 lines
Merged revisions 186445 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186445 | tilghman | 2009-04-03 14:56:48 -0500 (Fri, 03 Apr 2009) | 2 lines
Found a conflict in the last commit, due to multiple targets
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r186461 | kpfleming | 2009-04-03 15:20:01 -0500 (Fri, 03 Apr 2009) | 11 lines
Merged revisions 186458 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03 Apr 2009) | 5 lines
Fix a bug where DAHDI/Zaptel channels would not properly switch formats when requested
Don't offer AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could provide a slight performance benefit, the translation core in Asterisk has some flaws when a channel driver offers multiple raw formats. this fix is much simpler than fixing the translation core to solve that issue (although that will be done later).
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r186525 | mmichelson | 2009-04-03 17:41:46 -0500 (Fri, 03 Apr 2009) | 22 lines
This commit introduces COLP/CONP and Redirecting party information into Asterisk.
The channel drivers which have been most heavily tested with these enhancements are
chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be
introduced in a later commit. chan_skinny has code added to it here, but according
to user pj, the support on chan_skinny is not working as of now. This will be fixed in
a later commit.
A special thanks goes out to bugtracker user gareth for getting the ball rolling and
providing the initial support for this work. Without his initial work on this, this would
not have been nearly as painless as it was.
This functionality has been tested by Digium's product quality department, as well as a
customer site running thousands of calls every day. In addition, many many many many bugtracker
users have tested this, too.
(closes issue #8824)
Reported by: gareth
Review: http://reviewboard.digium.com/r/201
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r186537 | rmudgett | 2009-04-03 19:13:50 -0500 (Fri, 03 Apr 2009) | 1 line
Remove merged branch properties accidentally merged to trunk.
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r186563 | file | 2009-04-06 08:23:12 -0500 (Mon, 06 Apr 2009) | 8 lines
Pass the correct value to sizeof when copying address information.
(issue #14827)
Reported by: pj
Patches:
14827.diff uploaded by file (license 11)
Tested by: pj
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r186566 | mmichelson | 2009-04-06 08:57:39 -0500 (Mon, 06 Apr 2009) | 8 lines
Blocked revisions 186565 via svnmerge
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r186565 | mmichelson | 2009-04-06 08:54:41 -0500 (Mon, 06 Apr 2009) | 3 lines
Revert commit 186445 because it causes the build to fail when IMAP_STORAGE is used.
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r186620 | mmichelson | 2009-04-06 11:06:25 -0500 (Mon, 06 Apr 2009) | 3 lines
Silly svn. These files didn't get merged over in the merge of the issue8824 branch.
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r186624 | file | 2009-04-06 11:15:30 -0500 (Mon, 06 Apr 2009) | 13 lines
Add support for changing the outbound codec on a SIP call using
a dialplan variable.
This adds a dialplan variable (SIP_CODEC_OUTBOUND) which controls
the codec offered for an outgoing SIP call. This is much like the
SIP_CODEC dialplan variable and has the same restrictions. The codec
set must be one that is configured for the call.
(closes issue #13243)
Reported by: samdell3
Patches:
13243.diff uploaded by file (license 11)
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r186653 | file | 2009-04-06 12:03:07 -0500 (Mon, 06 Apr 2009) | 2 lines
Fix problem when authenticating a non-RTP dialog.
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r186687 | file | 2009-04-06 18:11:13 -0500 (Mon, 06 Apr 2009) | 2 lines
Fix a log message getting output when it should not have been.
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r186720 | mmichelson | 2009-04-07 15:46:18 -0500 (Tue, 07 Apr 2009) | 12 lines
Merged revisions 186719 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186719 | mmichelson | 2009-04-07 15:43:49 -0500 (Tue, 07 Apr 2009) | 6 lines
Ensure that \r\n is printed after the ActionID in an OriginateResponse.
(closes issue #14847)
Reported by: kobaz
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r186799 | tilghman | 2009-04-07 17:23:46 -0500 (Tue, 07 Apr 2009) | 10 lines
Merged revisions 186775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186775 | tilghman | 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009) | 3 lines
Fix Macro documentation to match current (and intended) behavior.
(See -dev mailing list)
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r186833 | mmichelson | 2009-04-07 18:50:56 -0500 (Tue, 07 Apr 2009) | 15 lines
Merged revisions 186832 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr 2009) | 8 lines
Set the AST_FEATURE_WARNING_ACTIVE flag when a p2p bridge returns AST_BRIDGE_RETRY.
Without this flag set, warning sounds will not be properly played to either party
of the bridge.
(closes issue #14845)
Reported by: adomjan
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r186837 | mmichelson | 2009-04-07 19:01:49 -0500 (Tue, 07 Apr 2009) | 7 lines
Fix bad merge from fix for issue 13867.
(closes issue #14686)
Reported by: davidw
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r186842 | mmichelson | 2009-04-07 19:09:28 -0500 (Tue, 07 Apr 2009) | 14 lines
Merged revisions 186841 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr 2009) | 8 lines
Fix a few typos of the word "frequency."
(closes issue #14842)
Reported by: jvandal
Patches:
frequency-typo.diff uploaded by jvandal (license 413)
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r186899 | tilghman | 2009-04-08 00:06:22 -0500 (Wed, 08 Apr 2009) | 2 lines
Add lastms to the require API call.
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r186928 | russell | 2009-04-08 07:35:57 -0500 (Wed, 08 Apr 2009) | 13 lines
Update some comments and resolve potential memory corruption in chan_sip.
While browsing chan_sip the other day, I noticed this dangerous code in
dialog_needdestroy(). This function is an ao2_callback. It is absolutely
_not_ okay to unlock the container from within this function. It's also not
clear why it was useful. Given that it could cause memory corruption, I have
removed it.
There was also a TODO comment left describing a potential implementation of
an improvement to the needdestroy handling. I'm not convinced that what was
described is the best choice here, so I have briefly described the way that
this function is used today that could be improved.
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r186953 | russell | 2009-04-08 08:24:48 -0500 (Wed, 08 Apr 2009) | 7 lines
Start splitting up miscellaneous doxygen documentation into separate files.
doxyref.h was created to hold miscellaneous documentation that was not specific
to a part of the code. This file has grown quite a bit so I decided to start
splitting parts of it out into new files. Now, you can drop a new file into
include/asterisk/doxygen/ and it will be processed by doxygen.
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r186957 | russell | 2009-04-08 08:38:27 -0500 (Wed, 08 Apr 2009) | 2 lines
Add some additional notes on release numbering.
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r186985 | mmichelson | 2009-04-08 10:27:41 -0500 (Wed, 08 Apr 2009) | 30 lines
Merged revisions 186984 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr 2009) | 24 lines
Make a couple of changes with regards to a new message printed in ast_read().
"ast_read() called with no recorded file descriptor" is a new message added
after a bug was discovered. Unfortunately, it seems there are a bunch of places
that potentially make such calls to ast_read() and trigger this error message
to be displayed. This commit does two things to help to make this message appear
less.
First, the message has been downgraded to a debug level message if dev mode is
not enabled. The message means a lot more to developers than it does to end users,
and so developers should take an effort to be sure to call ast_read only when
a channel is ready to be read from. However, since this doesn't actually cause an
error in operation and is not something a user can easily fix, we should not spam
their console with these messages.
Second, the message has been moved to after the check for any pending masquerades.
ast_read() being called with no recorded file descriptor should not interfere with
a masquerade taking place.
This could be seen as a simple way of resolving issue #14723. However, I still want
to try to clear out the existing ways of triggering this message, since I feel that
would be a better resolution for the issue.
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r187036 | file | 2009-04-08 11:27:36 -0500 (Wed, 08 Apr 2009) | 2 lines
Turn a warning message into a debug message and do not treat two situations as errors when they are not.
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r187046 | mmichelson | 2009-04-08 11:52:20 -0500 (Wed, 08 Apr 2009) | 16 lines
Merged revisions 187045 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed, 08 Apr 2009) | 10 lines
Fix a small logical error when loading moh classes.
We were unconditionally incrementing the number of mohclasses
registered. However, we should actually only increment if the
call to moh_register was successful.
While this probably has never caused problems, I noticed it
and decided to fix it anyway.
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r187050 | tilghman | 2009-04-08 12:08:43 -0500 (Wed, 08 Apr 2009) | 7 lines
If the first column is empty, output a delimiter anyway.
(closes issue #14848)
Reported by: john8675309
Patches:
20090408__bug14848.diff.txt uploaded by tilghman (license 14)
Tested by: john8675309
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r187105 | russell | 2009-04-08 12:51:35 -0500 (Wed, 08 Apr 2009) | 2 lines
Remove duplicate prototype for temp_peer().
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r187108 | file | 2009-04-08 13:12:28 -0500 (Wed, 08 Apr 2009) | 2 lines
Fix a bug where we would native bridge when we did not want to.
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r187138 | mmichelson | 2009-04-08 14:18:10 -0500 (Wed, 08 Apr 2009) | 13 lines
Blocked revisions 187135 via svnmerge
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r187135 | mmichelson | 2009-04-08 14:16:49 -0500 (Wed, 08 Apr 2009) | 8 lines
Fix a crash due to too few arguments to RetryDial.
(closes issue #14852)
Reported by: junky
Patches:
retry_fix.diff uploaded by junky (license 177)
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r187179 | russell | 2009-04-08 14:59:21 -0500 (Wed, 08 Apr 2009) | 2 lines
Add documentation for reviewboard usage and guidelines.
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r187210 | tilghman | 2009-04-08 15:39:55 -0500 (Wed, 08 Apr 2009) | 11 lines
Recorded merge of revisions 187209 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r187209 | tilghman | 2009-04-08 15:39:13 -0500 (Wed, 08 Apr 2009) | 4 lines
Backport resolution for file descriptor leak in 1.6.0 to 1.4.
This fixes short reads in http manager sessions, such as those done by the
ast-gui branch. (Fixes AST-198)
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r187211 | jpeeler | 2009-04-08 16:00:39 -0500 (Wed, 08 Apr 2009) | 20 lines
Add timer for features so that backup bridge config can go away
The biggest change done here was elimination of the backup_config for use with
features. Previously, the bridging code upon detecting a feature would set the
start time of the bridge to the start time of the feature. Then after the
feature had either expired or timed out the start time would be reset to the
true bridge start time from the backup_config. Now, the time differences are
calculated with respect to the newly added feature_start_time timeval instead.
There should be no behavior changes from the previous functionality aside from
the bridge timing being unaffected by either valid or partial feature matches.
Previously the timing would be increased by the length of time configured for
featuredigittimeout, which was probably never noticed.
(closes issue #14503)
Reported by: KNK
Tested by: jpeeler
Review: http://reviewboard.digium.com/r/179/
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r187269 | kpfleming | 2009-04-08 21:44:27 -0500 (Wed, 08 Apr 2009) | 5 lines
add a dedicated log channel for modules to be able report security-related events, so that they can be fed into external processes for analysis and possible mitigation efforts
(inspired by this evening's Toronto Asterisk Users Group meeting and previous dicussions amongst various community members)
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r187302 | tilghman | 2009-04-08 23:59:05 -0500 (Wed, 08 Apr 2009) | 14 lines
Merged revisions 187300-187301 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009) | 3 lines
Add debugging mode for diagnosing file descriptor leaks.
(Related to issue #14625)
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r187301 | tilghman | 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines
Oops, missed this file in the last commit.
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r187360 | file | 2009-04-09 11:19:35 -0500 (Thu, 09 Apr 2009) | 6 lines
Add support for allowing the channel driver to handle transcoding.
This was accomplished using a set of options and the setoption channel callback.
The core calls into the channel driver using these options and the channel driver
either returns success or failure.
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r187361 | file | 2009-04-09 11:27:53 -0500 (Thu, 09 Apr 2009) | 2 lines
Do not try to send the format read/format write/make compatible options over IAX2.
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r187363 | tilghman | 2009-04-09 11:39:43 -0500 (Thu, 09 Apr 2009) | 10 lines
Merged revisions 187362 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r187362 | tilghman | 2009-04-09 11:38:37 -0500 (Thu, 09 Apr 2009) | 3 lines
Permit zero-length text messages in SIP.
(Related to an issue posted to the -users list, subject "AEL2, BASE64_DECODE and hexadecimal")
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r187381 | tilghman | 2009-04-09 12:20:49 -0500 (Thu, 09 Apr 2009) | 4 lines
Allow '/' in username portion of register; this is a regression.
(closes issue #14668)
Reported by: Netview
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r187421 | mmichelson | 2009-04-09 12:30:39 -0500 (Thu, 09 Apr 2009) | 21 lines
Fix a crash in res_musiconhold when using cached realtime moh.
The moh_register function links an mohclass and then immediately
unrefs the class since the container now has a reference. The problem
with using realtime music on hold is that the class is allocated,
registered, and started in one fell swoop. The refcounting logic
resulted in the count being off by one. The same problem did not
happen when using a static config because the allocation and registration
of an mohclass is a separate operation from starting moh. This also did
not affect non-cached realtime moh because the classes are not registered
at all.
I also have modified res_musiconhold to use the _t_ variants of the ao2_
functions so that more info can be gleaned when attempting to trace the
refcounts. I found this to be incredibly helpful for debugging this issue
and there's no good reason to remove it.
(closes issue #14661)
Reported by: sum
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r187424 | mmichelson | 2009-04-09 12:34:39 -0500 (Thu, 09 Apr 2009) | 3 lines
Use safe macro practices even though they really aren't necessary.
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r187426 | dvossel | 2009-04-09 12:39:10 -0500 (Thu, 09 Apr 2009) | 5 lines
Fixes deadlock caused by calling get_cid_name with chan locked.
get_cid_name should not be called with a channel lock. get_cid_name calls ast_get_hint which eventually calls pbx_find_extension. pbx_find_extension starts and stops autoservice which should not be done with a channel lock, so get_cid_name should not be called with one.
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r187483 | tilghman | 2009-04-09 13:40:01 -0500 (Thu, 09 Apr 2009) | 15 lines
Merged revisions 187428 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09 Apr 2009) | 8 lines
Race condition between ast_cli_command() and 'module unload' could cause a deadlock.
Add lock timeouts to avoid this potential deadlock.
(closes issue #14705)
Reported by: jamessan
Patches:
20090320__bug14705.diff.txt uploaded by tilghman (license 14)
Tested by: jamessan
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r187488 | mmichelson | 2009-04-09 13:58:41 -0500 (Thu, 09 Apr 2009) | 24 lines
Merged revisions 187484 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r187484 | mmichelson | 2009-04-09 13:51:20 -0500 (Thu, 09 Apr 2009) | 18 lines
Handle a SIP race condition (reinvite before an ACK) properly.
RFC 5047 explains the proper course of action to take if a
reINVITE is received before the ACK from a previous invite
transaction. What we are to do is to treat the reINVITE as
if it were both an ACK and a reINVITE and process it normally.
Later, when we receive the ACK we had been expecting, we will
ignore it since its CSeq is less than the current iseqno of
the sip_pvt representing this dialog.
(closes issue #13849)
Reported by: klaus3000
Patches:
13849_v2.patch uploaded by mmichelson (license 60)
Tested by: mmichelson, klaus3000
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r187491 | jpeeler | 2009-04-09 14:10:02 -0500 (Thu, 09 Apr 2009) | 15 lines
Add ability for dialplan execution to continue when caller hangs up.
The F option to app_dial has been modified to accept no parameters and perform
the above functionality. I don't see anywhere else that is doing function
overloading, but this really is the best place for this operation because:
- It makes it close to the 'g' option in the argument list which provides
similar functionality.
- The existing code to support the current F option provides a very
convienient location to add this new feature.
(closes issue #12381)
Reported by: michael-fig
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r187556 | dvossel | 2009-04-09 15:40:34 -0500 (Thu, 09 Apr 2009) | 3 lines
More changes concerning r187426. Revised where locks are placed.
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r187560 | mmichelson | 2009-04-09 16:06:26 -0500 (Thu, 09 Apr 2009) | 11 lines
Add a new option, mwi_from, to sip.conf.
This allows for you to change the From header for outgoing MWI
NOTIFY requests. Prior to this, the best you could do was to
set a callerid in the general section of sip.conf. The problem
was that this was used for all outbound requests, not just
MWI NOTIFY requests.
AST-201
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r187599 | tilghman | 2009-04-09 22:55:27 -0500 (Thu, 09 Apr 2009) | 2 lines
Modify headers and macros, according to Russell's suggestions on the -dev list
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r187634 | rmudgett | 2009-04-10 09:50:42 -0500 (Fri, 10 Apr 2009) | 1 line
Make chan_misdn_log() avoid generating the log message if logging is disabled.
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r187635 | rmudgett | 2009-04-10 09:53:59 -0500 (Fri, 10 Apr 2009) | 7 lines
Miscellaneous minor changes to chan_misdn.
* Miscellaneous spacing and comment changes.
* Minor code rearangements.
* Miscellaneous doxygen comments.
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r187636 | kpfleming | 2009-04-10 10:11:16 -0500 (Fri, 10 Apr 2009) | 3 lines
revert addition of LOG_SECURITY log channel; after further discussion, a much better solution will be used
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r187673 | dvossel | 2009-04-10 10:49:16 -0500 (Fri, 10 Apr 2009) | 2 lines
Even more changes concerning r187426. Revised where locks are placed yet once again. ast_call() should not be called with a channel locked. could cause deadlock issues with local channels.
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r187674 | tilghman | 2009-04-10 10:59:40 -0500 (Fri, 10 Apr 2009) | 4 lines
Ensure pvt is not NULL before dereferencing it.
(closes issue #14784)
Reported by: pj
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r187675 | russell | 2009-04-10 11:00:29 -0500 (Fri, 10 Apr 2009) | 2 lines
Disable test modules by default.
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r187680 | mmichelson | 2009-04-10 11:06:22 -0500 (Fri, 10 Apr 2009) | 5 lines
Don't let ast_channel_alloc fail if explicitly passed NULL cid_name or cid_number.
This also fixes a small memory leak.
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r187714 | mmichelson | 2009-04-10 11:26:48 -0500 (Fri, 10 Apr 2009) | 3 lines
ast_strdup failures aren't really failures if the original value was NULL.
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r187721 | kpfleming | 2009-04-10 11:51:44 -0500 (Fri, 10 Apr 2009) | 5 lines
clean up some patterns for files to remove
add embedding support for bridge and test modules
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r187764 | tilghman | 2009-04-10 12:29:34 -0500 (Fri, 10 Apr 2009) | 9 lines
Merged revisions 187763 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r187763 | tilghman | 2009-04-10 12:28:46 -0500 (Fri, 10 Apr 2009) | 2 lines
Add lastms column to the contributed table designs
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r187770 | mmichelson | 2009-04-10 12:32:25 -0500 (Fri, 10 Apr 2009) | 4 lines
Make sure tc is unlocked before calling ast_call since calling a Local
channel could result in a deadlock.
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r187772 | file | 2009-04-10 13:02:44 -0500 (Fri, 10 Apr 2009) | 2 lines
Fix some uninitialized memory notices that appeared under valgrind.
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r187773 | file | 2009-04-10 13:14:47 -0500 (Fri, 10 Apr 2009) | 4 lines
Change how we set the local and remote address.
The code will now only change the address and port. It will not overwrite any other values.
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r187830 | mmichelson | 2009-04-10 13:56:52 -0500 (Fri, 10 Apr 2009) | 3 lines
Indicating connected line or redirecting updates were missing a call to lock the local_pvt.
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r187866 | russell | 2009-04-10 14:27:23 -0500 (Fri, 10 Apr 2009) | 11 lines
Blocked revisions 187865 via svnmerge
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r187865 | russell | 2009-04-10 14:26:40 -0500 (Fri, 10 Apr 2009) | 4 lines
Support "signaling" in addition to "signalling".
The sample configuration file has references to both spellings.
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r187906 | jpeeler | 2009-04-10 15:26:46 -0500 (Fri, 10 Apr 2009) | 12 lines
Fix module embedding for chan_h323.
Include libchanh323.a in the modules.link file so that all the symbols can be
resolved at link time.
(closes issue #11966)
Reported by: dome
Patches:
issue_11966.patch uploaded by kpfleming (license 421)
Tested by: jpeeler
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r187963 | jpeeler | 2009-04-10 17:16:38 -0500 (Fri, 10 Apr 2009) | 14 lines
Blocked revisions 187962 via svnmerge
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r187962 | jpeeler | 2009-04-10 17:16:13 -0500 (Fri, 10 Apr 2009) | 9 lines
Fix module embedding for chan_h323.
Include libchanh323.a in the modules.link file so that all the symbols can be
resolved at link time.
(closes issue #11966)
Reported by: dome
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r188032 | mmichelson | 2009-04-13 09:17:56 -0500 (Mon, 13 Apr 2009) | 6 lines
Set all queue variables on both the caller and member channels.
This allows for the variables to be accessed if a member macro is run.
Thanks to Grigoriy Puzankin for bringing this up on the -dev list.
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r188067 | file | 2009-04-13 11:28:06 -0500 (Mon, 13 Apr 2009) | 10 lines
Fix a bug where using an outbound proxy would cause the local address to be 127.0.0.1.
Copy the outbound proxy IP address into the SIP dialog structure as the IP address we will
be sending to. This has to be done because the logic that determines what local IP address to use
in the SIP messages is not aware of an outbound proxy being in place. It only knows what IP address
we are sending to.
(closes issue #12006)
Reported by: mnicholson
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r188102 | mmichelson | 2009-04-13 14:31:48 -0500 (Mon, 13 Apr 2009) | 5 lines
Fix another crash related to cached realtime music on hold.
This was another off-by-one problem caused by moh_register.
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r188150 | tilghman | 2009-04-13 18:05:53 -0500 (Mon, 13 Apr 2009) | 10 lines
Blocked revisions 188149 via svnmerge
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r188149 | tilghman | 2009-04-13 18:04:15 -0500 (Mon, 13 Apr 2009) | 4 lines
If fileconfig limit exceeds our maximum, then set the limit to the maximum.
(Closes issue #14888)
Reported by: falves11
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r188206 | tilghman | 2009-04-14 00:27:53 -0500 (Tue, 14 Apr 2009) | 6 lines
Application delimiter is ',', not '|'.
(closes issue #14881)
Reported by: stegro
Patches:
smsq.patch uploaded by stegro (license 752)
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r188210 | tilghman | 2009-04-14 00:45:13 -0500 (Tue, 14 Apr 2009) | 2 lines
As suggested by Russell, warn users when their dialplan arguments contain pipes, but not commas.
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r188247 | file | 2009-04-14 08:14:21 -0500 (Tue, 14 Apr 2009) | 6 lines
Fix a bug with the change I made yesterday to outbound proxy support.
Per discussion with oej on IRC we need the actual IP address, not the
outbound proxy IP address, in the sa field. This change matches the already
existing code for all other uses of the outbound proxy setting.
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r188283 | oej | 2009-04-14 09:20:10 -0500 (Tue, 14 Apr 2009) | 4 lines
Making sure we have references to external libraries.
Note: Update h.323 with the recent changes too
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r188284 | oej | 2009-04-14 09:22:39 -0500 (Tue, 14 Apr 2009) | 2 lines
New actions should go under "New Actions", not "new events"
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r188342 | jpeeler | 2009-04-14 10:54:16 -0500 (Tue, 14 Apr 2009) | 33 lines
Add service maintenance message support
This is the companion commit to libpri r732. Service messages are now supported
for switch types 4ess/5ess. A new option service_message_support has been added
to chan_dahdi.conf and is noted in the sample config file. The service message
[... 122639 lines stripped ...]
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