[asterisk-commits] snuffy: branch snuffy/ao2_jabber_take2 r187597 - in /team/snuffy/ao2_jabber_t...

[SVN commits to the Asterisk project]

Author: snuffy
Date: Thu Apr  9 17:41:24 2009
New Revision: 187597

URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=187597
Log:
Merged revisions 186058,186060,186078,186101,186175,186230,186286,186297,186321,186379,186382,186444,186447,186461,186525,186537,186563,186566,186620,186624,186653,186687,186720,186799,186833,186837,186842,186899,186928,186953,186957,186985,187036,187046,187050,187105,187108,187138,187179,187210-187211,187269,187302,187360-187361,187363,187381,187421,187424,187426,187483,187488,187491,187556,187560 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
  r186058 | tilghman | 2009-04-03 04:04:40 +1100 (Fri, 03 Apr 2009) | 8 lines
  
  Blocked revisions 186057 via svnmerge
  
  ........
    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 | 2009-04-03 04:10:28 +1100 (Fri, 03 Apr 2009) | 16 lines
  
  Merged revisions 186059 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ................
    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
    
    ........
      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 | 2009-04-03 04:20:52 +1100 (Fri, 03 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 | 2009-04-03 04:26:07 +1100 (Fri, 03 Apr 2009) | 9 lines
  
  Merged revisions 186081 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    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 | 2009-04-03 08:56:21 +1100 (Fri, 03 Apr 2009) | 11 lines
  
  Merged revisions 186174 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    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 | 2009-04-03 13:03:48 +1100 (Fri, 03 Apr 2009) | 29 lines
  
  Merged revisions 186229 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
  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 | 2009-04-04 01:32:05 +1100 (Sat, 04 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
................
  r186297 | tilghman | 2009-04-04 02:18:28 +1100 (Sat, 04 Apr 2009) | 4 lines
  
  Compatibility fix for glibc 2.4
  (Closes issue #14820)
  Reported by: phsultan
................
  r186321 | file | 2009-04-04 02:52:50 +1100 (Sat, 04 Apr 2009) | 12 lines
  
  Merged revisions 186320 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    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 | 2009-04-04 03:29:47 +1100 (Sat, 04 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 | 2009-04-04 03:47:27 +1100 (Sat, 04 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 | 2009-04-04 06:30:34 +1100 (Sat, 04 Apr 2009) | 14 lines
  
  Merged revisions 186415 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    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 | 2009-04-04 06:59:55 +1100 (Sat, 04 Apr 2009) | 9 lines
  
  Merged revisions 186445 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    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 | 2009-04-04 07:20:01 +1100 (Sat, 04 Apr 2009) | 11 lines
  
  Merged revisions 186458 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    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 | 2009-04-04 09:41:46 +1100 (Sat, 04 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 | 2009-04-04 11:13:50 +1100 (Sat, 04 Apr 2009) | 1 line
  
  Remove merged branch properties accidentally merged to trunk.
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  r186563 | file | 2009-04-06 23:23:12 +1000 (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 23:57:39 +1000 (Mon, 06 Apr 2009) | 8 lines
  
  Blocked revisions 186565 via svnmerge
  
  ........
    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 | 2009-04-07 02:06:25 +1000 (Tue, 07 Apr 2009) | 3 lines
  
  Silly svn. These files didn't get merged over in the merge of the issue8824 branch.
................
  r186624 | file | 2009-04-07 02:15:30 +1000 (Tue, 07 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 | 2009-04-07 03:03:07 +1000 (Tue, 07 Apr 2009) | 2 lines
  
  Fix problem when authenticating a non-RTP dialog.
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  r186687 | file | 2009-04-07 09:11:13 +1000 (Tue, 07 Apr 2009) | 2 lines
  
  Fix a log message getting output when it should not have been.
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  r186720 | mmichelson | 2009-04-08 06:46:18 +1000 (Wed, 08 Apr 2009) | 12 lines
  
  Merged revisions 186719 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    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 | 2009-04-08 08:23:46 +1000 (Wed, 08 Apr 2009) | 10 lines
  
  Merged revisions 186775 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    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 | 2009-04-08 09:50:56 +1000 (Wed, 08 Apr 2009) | 15 lines
  
  Merged revisions 186832 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    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 | 2009-04-08 10:01:49 +1000 (Wed, 08 Apr 2009) | 7 lines
  
  Fix bad merge from fix for issue 13867.
  
  (closes issue #14686)
  Reported by: davidw
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  r186842 | mmichelson | 2009-04-08 10:09:28 +1000 (Wed, 08 Apr 2009) | 14 lines
  
  Merged revisions 186841 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    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 15:06:22 +1000 (Wed, 08 Apr 2009) | 2 lines
  
  Add lastms to the require API call.
................
  r186928 | russell | 2009-04-08 22:35:57 +1000 (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.
................
  r186953 | russell | 2009-04-08 23:24:48 +1000 (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.
................
  r186957 | russell | 2009-04-08 23:38:27 +1000 (Wed, 08 Apr 2009) | 2 lines
  
  Add some additional notes on release numbering.
................
  r186985 | mmichelson | 2009-04-09 01:27:41 +1000 (Thu, 09 Apr 2009) | 30 lines
  
  Merged revisions 186984 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    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 | 2009-04-09 02:27:36 +1000 (Thu, 09 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 | 2009-04-09 02:52:20 +1000 (Thu, 09 Apr 2009) | 16 lines
  
  Merged revisions 187045 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    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 | 2009-04-09 03:08:43 +1000 (Thu, 09 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
................
  r187105 | russell | 2009-04-09 03:51:35 +1000 (Thu, 09 Apr 2009) | 2 lines
  
  Remove duplicate prototype for temp_peer().
................
  r187108 | file | 2009-04-09 04:12:28 +1000 (Thu, 09 Apr 2009) | 2 lines
  
  Fix a bug where we would native bridge when we did not want to.
................
  r187138 | mmichelson | 2009-04-09 05:18:10 +1000 (Thu, 09 Apr 2009) | 13 lines
  
  Blocked revisions 187135 via svnmerge
  
  ........
    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 | 2009-04-09 05:59:21 +1000 (Thu, 09 Apr 2009) | 2 lines
  
  Add documentation for reviewboard usage and guidelines.
................
  r187210 | tilghman | 2009-04-09 06:39:55 +1000 (Thu, 09 Apr 2009) | 11 lines
  
  Recorded merge of revisions 187209 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    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)
  ........
................
  r187211 | jpeeler | 2009-04-09 07:00:39 +1000 (Thu, 09 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/
................
  r187269 | kpfleming | 2009-04-09 12:44:27 +1000 (Thu, 09 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)
................
  r187302 | tilghman | 2009-04-09 14:59:05 +1000 (Thu, 09 Apr 2009) | 14 lines
  
  Merged revisions 187300-187301 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    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)
  ........
    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 | 2009-04-10 02:19:35 +1000 (Fri, 10 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.
................
  r187361 | file | 2009-04-10 02:27:53 +1000 (Fri, 10 Apr 2009) | 2 lines
  
  Do not try to send the format read/format write/make compatible options over IAX2.
................
  r187363 | tilghman | 2009-04-10 02:39:43 +1000 (Fri, 10 Apr 2009) | 10 lines
  
  Merged revisions 187362 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    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 | 2009-04-10 03:20:49 +1000 (Fri, 10 Apr 2009) | 4 lines
  
  Allow '/' in username portion of register; this is a regression.
  (closes issue #14668)
   Reported by: Netview
................
  r187421 | mmichelson | 2009-04-10 03:30:39 +1000 (Fri, 10 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
................
  r187424 | mmichelson | 2009-04-10 03:34:39 +1000 (Fri, 10 Apr 2009) | 3 lines
  
  Use safe macro practices even though they really aren't necessary.
................
  r187426 | dvossel | 2009-04-10 03:39:10 +1000 (Fri, 10 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.
................
  r187483 | tilghman | 2009-04-10 04:40:01 +1000 (Fri, 10 Apr 2009) | 15 lines
  
  Merged revisions 187428 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    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
  ........
................
  r187488 | mmichelson | 2009-04-10 04:58:41 +1000 (Fri, 10 Apr 2009) | 24 lines
  
  Merged revisions 187484 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    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
  ........
................
  r187491 | jpeeler | 2009-04-10 05:10:02 +1000 (Fri, 10 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
................
  r187556 | dvossel | 2009-04-10 06:40:34 +1000 (Fri, 10 Apr 2009) | 3 lines
  
  More changes concerning r187426. Revised where locks are placed.
................
  r187560 | mmichelson | 2009-04-10 07:06:26 +1000 (Fri, 10 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
................

Added:
    team/snuffy/ao2_jabber_take2/funcs/func_connectedline.c
      - copied unchanged from r187560, trunk/funcs/func_connectedline.c
    team/snuffy/ao2_jabber_take2/funcs/func_redirecting.c
      - copied unchanged from r187560, trunk/funcs/func_redirecting.c
    team/snuffy/ao2_jabber_take2/include/asterisk/doxygen/
      - copied from r187560, trunk/include/asterisk/doxygen/
    team/snuffy/ao2_jabber_take2/include/asterisk/doxygen/commits.h
      - copied unchanged from r187560, trunk/include/asterisk/doxygen/commits.h
    team/snuffy/ao2_jabber_take2/include/asterisk/doxygen/licensing.h
      - copied unchanged from r187560, trunk/include/asterisk/doxygen/licensing.h
    team/snuffy/ao2_jabber_take2/include/asterisk/doxygen/releases.h
      - copied unchanged from r187560, trunk/include/asterisk/doxygen/releases.h
    team/snuffy/ao2_jabber_take2/include/asterisk/doxygen/reviewboard.h
      - copied unchanged from r187560, trunk/include/asterisk/doxygen/reviewboard.h
    team/snuffy/ao2_jabber_take2/include/asterisk/rtp_engine.h
      - copied unchanged from r187560, trunk/include/asterisk/rtp_engine.h
    team/snuffy/ao2_jabber_take2/include/asterisk/stun.h
      - copied unchanged from r187560, trunk/include/asterisk/stun.h
    team/snuffy/ao2_jabber_take2/main/astfd.c
      - copied unchanged from r187560, trunk/main/astfd.c
    team/snuffy/ao2_jabber_take2/main/rtp_engine.c
      - copied unchanged from r187560, trunk/main/rtp_engine.c
    team/snuffy/ao2_jabber_take2/main/stun.c
      - copied unchanged from r187560, trunk/main/stun.c
    team/snuffy/ao2_jabber_take2/res/res_rtp_asterisk.c
      - copied unchanged from r187560, trunk/res/res_rtp_asterisk.c
Removed:
    team/snuffy/ao2_jabber_take2/include/asterisk/rtp.h
    team/snuffy/ao2_jabber_take2/main/rtp.c
Modified:
    team/snuffy/ao2_jabber_take2/   (props changed)
    team/snuffy/ao2_jabber_take2/CHANGES
    team/snuffy/ao2_jabber_take2/Makefile
    team/snuffy/ao2_jabber_take2/UPGRADE.txt
    team/snuffy/ao2_jabber_take2/agi/Makefile
    team/snuffy/ao2_jabber_take2/apps/app_dial.c
    team/snuffy/ao2_jabber_take2/apps/app_directed_pickup.c
    team/snuffy/ao2_jabber_take2/apps/app_macro.c
    team/snuffy/ao2_jabber_take2/apps/app_queue.c
    team/snuffy/ao2_jabber_take2/apps/app_sendtext.c
    team/snuffy/ao2_jabber_take2/apps/app_verbose.c
    team/snuffy/ao2_jabber_take2/apps/app_voicemail.c
    team/snuffy/ao2_jabber_take2/build_tools/cflags.xml
    team/snuffy/ao2_jabber_take2/cdr/cdr_radius.c
    team/snuffy/ao2_jabber_take2/channels/chan_agent.c
    team/snuffy/ao2_jabber_take2/channels/chan_bridge.c
    team/snuffy/ao2_jabber_take2/channels/chan_dahdi.c
    team/snuffy/ao2_jabber_take2/channels/chan_gtalk.c
    team/snuffy/ao2_jabber_take2/channels/chan_h323.c
    team/snuffy/ao2_jabber_take2/channels/chan_iax2.c
    team/snuffy/ao2_jabber_take2/channels/chan_jingle.c
    team/snuffy/ao2_jabber_take2/channels/chan_local.c
    team/snuffy/ao2_jabber_take2/channels/chan_mgcp.c
    team/snuffy/ao2_jabber_take2/channels/chan_misdn.c
    team/snuffy/ao2_jabber_take2/channels/chan_phone.c
    team/snuffy/ao2_jabber_take2/channels/chan_sip.c
    team/snuffy/ao2_jabber_take2/channels/chan_skinny.c
    team/snuffy/ao2_jabber_take2/channels/chan_unistim.c
    team/snuffy/ao2_jabber_take2/channels/misdn/chan_misdn_config.h
    team/snuffy/ao2_jabber_take2/channels/misdn/isdn_lib.c
    team/snuffy/ao2_jabber_take2/channels/misdn/isdn_lib.h
    team/snuffy/ao2_jabber_take2/channels/misdn/isdn_lib_intern.h
    team/snuffy/ao2_jabber_take2/channels/misdn/isdn_msg_parser.c
    team/snuffy/ao2_jabber_take2/channels/misdn_config.c
    team/snuffy/ao2_jabber_take2/configs/features.conf.sample
    team/snuffy/ao2_jabber_take2/configs/logger.conf.sample
    team/snuffy/ao2_jabber_take2/configs/misdn.conf.sample
    team/snuffy/ao2_jabber_take2/configs/sip.conf.sample
    team/snuffy/ao2_jabber_take2/configs/voicemail.conf.sample
    team/snuffy/ao2_jabber_take2/contrib/asterisk-ng-doxygen
    team/snuffy/ao2_jabber_take2/doc/tex/channelvariables.tex
    team/snuffy/ao2_jabber_take2/formats/format_wav.c
    team/snuffy/ao2_jabber_take2/formats/format_wav_gsm.c
    team/snuffy/ao2_jabber_take2/funcs/func_odbc.c
    team/snuffy/ao2_jabber_take2/include/asterisk.h
    team/snuffy/ao2_jabber_take2/include/asterisk/_private.h
    team/snuffy/ao2_jabber_take2/include/asterisk/callerid.h
    team/snuffy/ao2_jabber_take2/include/asterisk/channel.h
    team/snuffy/ao2_jabber_take2/include/asterisk/crypto.h
    team/snuffy/ao2_jabber_take2/include/asterisk/doxyref.h
    team/snuffy/ao2_jabber_take2/include/asterisk/frame.h
    team/snuffy/ao2_jabber_take2/include/asterisk/jabber.h
    team/snuffy/ao2_jabber_take2/include/asterisk/linkedlists.h
    team/snuffy/ao2_jabber_take2/include/asterisk/lock.h
    team/snuffy/ao2_jabber_take2/include/asterisk/logger.h
    team/snuffy/ao2_jabber_take2/include/asterisk/pbx.h
    team/snuffy/ao2_jabber_take2/main/Makefile
    team/snuffy/ao2_jabber_take2/main/asterisk.c
    team/snuffy/ao2_jabber_take2/main/audiohook.c
    team/snuffy/ao2_jabber_take2/main/callerid.c
    team/snuffy/ao2_jabber_take2/main/channel.c
    team/snuffy/ao2_jabber_take2/main/dial.c
    team/snuffy/ao2_jabber_take2/main/features.c
    team/snuffy/ao2_jabber_take2/main/file.c
    team/snuffy/ao2_jabber_take2/main/loader.c
    team/snuffy/ao2_jabber_take2/main/logger.c
    team/snuffy/ao2_jabber_take2/main/manager.c
    team/snuffy/ao2_jabber_take2/main/pbx.c
    team/snuffy/ao2_jabber_take2/main/stdtime/localtime.c
    team/snuffy/ao2_jabber_take2/res/res_jabber.c
    team/snuffy/ao2_jabber_take2/res/res_musiconhold.c
    team/snuffy/ao2_jabber_take2/utils/Makefile

Propchange: team/snuffy/ao2_jabber_take2/
------------------------------------------------------------------------------
Binary property 'branch-1.4-blocked' - no diff available.

Propchange: team/snuffy/ao2_jabber_take2/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Propchange: team/snuffy/ao2_jabber_take2/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Thu Apr  9 17:41:24 2009
@@ -1,1 +1,1 @@
-/trunk:1-186048
+/trunk:1-187593

Modified: team/snuffy/ao2_jabber_take2/CHANGES
URL: http://svn.digium.com/svn-view/asterisk/team/snuffy/ao2_jabber_take2/CHANGES?view=diff&rev=187597&r1=187596&r2=187597
==============================================================================
--- team/snuffy/ao2_jabber_take2/CHANGES (original)
+++ team/snuffy/ao2_jabber_take2/CHANGES Thu Apr  9 17:41:24 2009
@@ -7,7 +7,6 @@
 === and the other UPGRADE files for older releases.
 ===
 ======================================================================
-
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.6.3  -------------
 ------------------------------------------------------------------------------
@@ -16,16 +15,58 @@
 -----------
  * Added preferred_codec_only option in sip.conf. This feature limits the joint
    codecs sent in response to an INVITE to the single most preferred codec.
+ * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
+   to be used for the outgoing call. It must be one of the codecs configured
+   for the device.
 
 Applications
 ------------
  * Added progress option to the app_dial D() option.  When progress DTMF is
    present, those values are sent immediatly upon receiving a PROGRESS message
    regardless if the call has been answered or not.
-
-Functions
----------
+ * Added functionality to the app_dial F() option to continue with execution
+   at the current location when no parameters are provided.
+
+Dialplan Functions
+------------------
+ * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
+   setting various connected line and redirecting party information.
  * The CHANNEL() function now supports the "name" option.
+
+Queue changes
+-------------
+  * A new option, 'I' has been added to both app_queue and app_dial.
+    By setting this option, Asterisk will not update the caller with
+    connected line changes or redirecting party changes when they occur.
+
+mISDN channel driver (chan_misdn) changes
+----------------------------------------
+  * Added display_connected parameter to misdn.conf to put a display string
+    in the CONNECT message containing the connected name and/or number if
+    the presentation setting permits it.
+  * Added display_setup parameter to misdn.conf to put a display string
+    in the SETUP message containing the caller name and/or number if the
+    presentation setting permits it.
+  * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
+    indicate the dialplan settings are to be obtained from the asterisk
+    channel.
+  * Made misdn.conf parameter callerid accept the "name" <number> format
+    used by the rest of the system.
+  * Made use the nationalprefix and internationalprefix misdn.conf
+    parameters to prefix any received number from the ISDN link if that
+    number has the corresponding Type-Of-Number.
+  * Added the following new parameters: unknownprefix, netspecificprefix,
+    subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
+    received number from the ISDN link if that number has the corresponding
+    Type-Of-Number.
+
+
+SIP channel driver (chan_sip) changes
+-------------------------------------------
+  * The sendrpid parameter has been expanded to include the options
+    'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
+    header to be sent (equivalent to setting sendrpid=yes) and setting
+    sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
 
 Asterisk Manager Interface
 --------------------------

Modified: team/snuffy/ao2_jabber_take2/Makefile
URL: http://svn.digium.com/svn-view/asterisk/team/snuffy/ao2_jabber_take2/Makefile?view=diff&rev=187597&r1=187596&r2=187597
==============================================================================
--- team/snuffy/ao2_jabber_take2/Makefile (original)
+++ team/snuffy/ao2_jabber_take2/Makefile Thu Apr  9 17:41:24 2009
@@ -559,8 +559,10 @@
 		chmod 755 $(DESTDIR)$(ASTSBINDIR)/safe_asterisk;\
 	fi
 	$(INSTALL) -d $(DESTDIR)$(ASTHEADERDIR)
+	$(INSTALL) -d $(DESTDIR)$(ASTHEADERDIR)/doxygen
 	$(INSTALL) -m 644 include/asterisk.h $(DESTDIR)$(includedir)
 	$(INSTALL) -m 644 include/asterisk/*.h $(DESTDIR)$(ASTHEADERDIR)
+	$(INSTALL) -m 644 include/asterisk/doxygen/*.h $(DESTDIR)$(ASTHEADERDIR)/doxygen
 	if [ -n "$(OLDHEADERS)" ]; then \
 		rm -f $(addprefix $(DESTDIR)$(ASTHEADERDIR)/,$(OLDHEADERS)) ;\
 	fi

Modified: team/snuffy/ao2_jabber_take2/UPGRADE.txt
URL: http://svn.digium.com/svn-view/asterisk/team/snuffy/ao2_jabber_take2/UPGRADE.txt?view=diff&rev=187597&r1=187596&r2=187597
==============================================================================
--- team/snuffy/ao2_jabber_take2/UPGRADE.txt (original)
+++ team/snuffy/ao2_jabber_take2/UPGRADE.txt Thu Apr  9 17:41:24 2009
@@ -20,7 +20,11 @@
 
 From 1.6.2 to 1.6.3:
 
-* Nothing, yet!
+* The usage of RTP inside of Asterisk has now become modularized. This means
+  the Asterisk RTP stack now exists as a loadable module, res_rtp_asterisk.
+  If you are not using autoload=yes in modules.conf you will need to ensure
+  it is set to load. If not, then any module which uses RTP (such as chan_sip)
+  will not be able to send or receive calls.
 
 From 1.6.1 to 1.6.2:
 

Modified: team/snuffy/ao2_jabber_take2/agi/Makefile
URL: http://svn.digium.com/svn-view/asterisk/team/snuffy/ao2_jabber_take2/agi/Makefile?view=diff&rev=187597&r1=187596&r2=187597
==============================================================================
--- team/snuffy/ao2_jabber_take2/agi/Makefile (original)
+++ team/snuffy/ao2_jabber_take2/agi/Makefile Thu Apr  9 17:41:24 2009
@@ -24,6 +24,8 @@
 endif
 
 include $(ASTTOPDIR)/Makefile.rules
+
+ASTCFLAGS+=-DSTANDALONE
 
 all: $(AGIS)
 

Modified: team/snuffy/ao2_jabber_take2/apps/app_dial.c
URL: http://svn.digium.com/svn-view/asterisk/team/snuffy/ao2_jabber_take2/apps/app_dial.c?view=diff&rev=187597&r1=187596&r2=187597
==============================================================================
--- team/snuffy/ao2_jabber_take2/apps/app_dial.c (original)
+++ team/snuffy/ao2_jabber_take2/apps/app_dial.c Thu Apr  9 17:41:24 2009
@@ -54,7 +54,7 @@
 #include "asterisk/utils.h"
 #include "asterisk/app.h"
 #include "asterisk/causes.h"
-#include "asterisk/rtp.h"
+#include "asterisk/rtp_engine.h"
 #include "asterisk/cdr.h"
 #include "asterisk/manager.h"
 #include "asterisk/privacy.h"
@@ -133,6 +133,10 @@

[... 18588 lines stripped ...]