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  r295869 | seanbright | 2010-11-22 15:03:49 -0500 (Mon, 22 Nov 2010) | 9 lines
  
  Merged revisions 295868 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r295868 | seanbright | 2010-11-22 15:02:37 -0500 (Mon, 22 Nov 2010) | 2 lines
    
    Change some documentation to suggest dahdi_monitor instead of ztmonitor.
  ........
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  r295866 | rmudgett | 2010-11-22 13:36:10 -0600 (Mon, 22 Nov 2010) | 60 lines
  
  Merged revisions 295843 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r295843 | rmudgett | 2010-11-22 13:28:23 -0600 (Mon, 22 Nov 2010) | 53 lines
    
    Merged revisions 295790 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) | 46 lines
      
      The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.
      
      To recreate the problem:
      1) Party A calls Party B
      2) Invoke CLI "channel redirect" command to redirect channel call leg
      associated with A.
      3) All associated channels are hung up.
      
      Note that if the CLI command were done on the channel call leg associated
      with B it works.
      
      This regression was a result of the fix for issue #16946
      (https://reviewboard.asterisk.org/r/740/).
      
      The regression affects all features that use an async goto to execute the
      dialplan because of an external event: Channel redirect, AMI redirect, SIP
      REFER, and FAX detection.
      
      The struct ast_channel._softhangup code is a mess.  The variable is used
      for several purposes that do not necessarily result in the call being hung
      up.  I have added doxygen comments to describe how the various _softhangup
      bits are used.  I have corrected all the places where the variable was
      tested in a non-bit oriented manner.
      
      The primary fix is the new AST_CONTROL_END_OF_Q frame.  It acts as a weak
      hangup request so the soft hangup requests that do not normally result in
      a hangup do not hangup.
      
      JIRA SWP-2470
      JIRA SWP-2489
      
      (closes issue #18171)
      Reported by: SantaFox
      (closes issue #18185)
      Reported by: kwemheuer
      (closes issue #18211)
      Reported by: zahir_koradia
      (closes issue #18230)
      Reported by: vmarrone
      (closes issue #18299)
      Reported by: mbrevda
      (closes issue #18322)
      Reported by: nerbos
      
      Review:	https://reviewboard.asterisk.org/r/1013/
    ........
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behavior of the command with this patch is almost exactly like that of GET DATA.


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  r295747 | rmudgett | 2010-11-19 21:11:15 -0600 (Fri, 19 Nov 2010) | 13 lines
  
  One way audio before answering call waiting call on analog port.
  
  * Analog call waiting Caller ID spills could get stuck resulting in one
  way audio until the waiting call is answered.  This only happens on the
  second (and later) call waiting call if the active call is not the first
  call.
  
  * The CLI/AMI "dahdi show channel" command could report the wrong channel
  information.
  
  Must keep the struct analog_pvt.owner and struct dahdi_pvt.owner pointer
  in sync.
........


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  r295711 | russell | 2010-11-19 18:50:00 -0600 (Fri, 19 Nov 2010) | 36 lines
  
  Merged revisions 295710 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r295710 | russell | 2010-11-19 18:45:51 -0600 (Fri, 19 Nov 2010) | 29 lines
    
    Fix cache of device state changes for multiple servers.
    
    This patch addresses a regression where device states across multiple servers
    were not being processing completely correctly.  The code works to determine
    the overall state by looking at the last known state of a device on each
    server.  However, there was a regression due to some invasive rewrites of how
    the cache works that led to the cache only storing the last device state change
    for a device, regardless of which server it was on.
    
    The code is set up to cache device state change events by ensuring that each
    event in the cache has a unique device name + entity ID (server ID).  The code
    that was responsible for comparing raw information elements (which EID is)
    always returned a match due to a memcmp() with a length of 0.
    
    There isn't much code to fix the actual bug.  This patch also introduces a new
    CLI command that was very useful for debugging this problem.  The command
    allows you to dump the contents of the event cache.
    
    (closes issue #18284)
    Reported by: klaus3000
    Patches:
          issue18284.rev1.txt uploaded by russell (license 2)
    Tested by: russell, klaus3000
    
    (closes issue #18280)
    Reported by: klaus3000
    
    Review: https://reviewboard.asterisk.org/r/1012/
  ........
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  r295673 | twilson | 2010-11-19 14:06:10 -0800 (Fri, 19 Nov 2010) | 22 lines
  
  Merged revisions 295672 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r295672 | twilson | 2010-11-19 13:55:48 -0800 (Fri, 19 Nov 2010) | 15 lines
    
    Merged revisions 295628 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r295628 | twilson | 2010-11-19 12:53:36 -0800 (Fri, 19 Nov 2010) | 8 lines
      
      Discard responses with more than one Via
      
      This is not a perfect solution as headers that are joined via commas are not
      detected. This is a parsing issue that to fix "correctly" would necessitate 
      a new SIP parser.
      
      Review: https://reviewboard.asterisk.org/r/1019/
    ........
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  r295670 | bbryant | 2010-11-19 16:40:21 -0500 (Fri, 19 Nov 2010) | 8 lines
  
  Patch for deadlock from ordering issue between channel/queue locks in app_queue
  (set_queue_variables).
  
  (closes issue #18031)
  Reported by: rain
  
  Review: https://reviewboard.asterisk.org/r/1018/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@295671 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add the ability to play a sound file, listen for more than just one digit,
specify
escape characters. Backwards compatible (to work with only timeout specified).

(closes issue #15531)
Reported by: diLLec
Patches:
      asterisk-res_agi-203638-patched.patch uploaded by diLLec (license 839)
Tested by: diLLec, espiceland



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  r295516 | rmudgett | 2010-11-19 10:47:11 -0600 (Fri, 19 Nov 2010) | 13 lines
  
  Bring sig_analog extraction more into alignment with orig-trunk/v1.6.2 chan_dahdi.
  
  * Restore SMDI support.
  * Fixed initial value of struct analog_pvt.use_callerid.  It may get
  forced on depending upon other config options.
  * Call analog_dnd() instead of manual inlined code.
  * Removed unused struct analog_pvt.usedistinctiveringdetection.
  * Removed the struct analog_pvt.unknown_alarm flag.  It was really the
  struct analog_pvt.inalarm flag.
  * Use ast_debug() instead of ast_log(LOG_DEBUG).
  * Rename several function's index variable to idx.
  * Some formatting tweaks.
........


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  r295477 | lmadsen | 2010-11-18 14:30:35 -0600 (Thu, 18 Nov 2010) | 6 lines
  
  'sip notify clear-mwi' needs terminating CRLF.
  
  (closes issue #18275)
  Reported by: klaus3000
  Patches:
        fix_body_CRLF_patch.txt uploaded by klaus3000 (license 65)
........


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  r295441 | pabelanger | 2010-11-18 13:02:12 -0500 (Thu, 18 Nov 2010) | 11 lines
  
  Merged revisions 295440 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r295440 | pabelanger | 2010-11-18 12:51:34 -0500 (Thu, 18 Nov 2010) | 4 lines
    
    Fix compiler warnings when using openssl-dev 1.0.0+
    
    Review: https://reviewboard.asterisk.org/r/1016/
  ........
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  r295404 | pabelanger | 2010-11-18 00:12:05 -0500 (Thu, 18 Nov 2010) | 2 lines
  
  Add RedHat specific dependencies
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  r295361 | pabelanger | 2010-11-17 09:09:38 -0500 (Wed, 17 Nov 2010) | 2 lines
  
  Uncomment settings under [global], to surpress warning when loading Asterisk.
........


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  r295282 | rmudgett | 2010-11-16 17:02:36 -0600 (Tue, 16 Nov 2010) | 16 lines
  
  Merged revisions 295281 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r295281 | rmudgett | 2010-11-16 16:57:07 -0600 (Tue, 16 Nov 2010) | 9 lines
    
    Merged revisions 295280 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r295280 | rmudgett | 2010-11-16 16:52:06 -0600 (Tue, 16 Nov 2010) | 1 line
      
      Dead code elimination in channel.c:ast_channel_bridge() variable who.
    ........
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  r295278 | russell | 2010-11-16 16:41:11 -0600 (Tue, 16 Nov 2010) | 2 lines
  
  Check for pdftotext and give a useful error if not found.
........


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  r295201 | russell | 2010-11-16 15:46:18 -0600 (Tue, 16 Nov 2010) | 2 lines
  
  Remove intentional typo I had added when testing the check.  oops.
........


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  r295164 | russell | 2010-11-16 14:50:03 -0600 (Tue, 16 Nov 2010) | 2 lines
  
  Check for wikiexport.py in PATH and give a useful error message if not found.
........


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(testing something with bamboo ...)


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  r295078 | tilghman | 2010-11-15 12:30:13 -0600 (Mon, 15 Nov 2010) | 16 lines
  
  Merged revisions 295062 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r295062 | tilghman | 2010-11-15 12:24:02 -0600 (Mon, 15 Nov 2010) | 9 lines
    
    Merged revisions 295026 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r295026 | tilghman | 2010-11-15 11:58:37 -0600 (Mon, 15 Nov 2010) | 2 lines
      
      Create test verifying results of expression parser
    ........
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  r294989 | tilghman | 2010-11-15 01:44:38 -0600 (Mon, 15 Nov 2010) | 15 lines
  
  Merged revisions 294988 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r294988 | tilghman | 2010-11-15 01:42:39 -0600 (Mon, 15 Nov 2010) | 8 lines
    
    It is possible to crash Asterisk by feeding the curl engine invalid data.
    
    (closes issue #18161)
     Reported by: wdoekes
     Patches: 
           20101029__issue18161.diff.txt uploaded by tilghman (license 14)
     Tested by: tilghman
  ........
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  r294911 | jpeeler | 2010-11-12 15:14:43 -0600 (Fri, 12 Nov 2010) | 11 lines
  
  Merged revisions 294910 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r294910 | jpeeler | 2010-11-12 15:14:23 -0600 (Fri, 12 Nov 2010) | 4 lines
    
    Return correct error code if lock path fails. The recent changes to open_mailbox actually caused it to be fixed, but let's be consistent.
    
    Reported by alecdavis in asterisk-dev.
  ........
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  r294905 | jpeeler | 2010-11-12 14:52:06 -0600 (Fri, 12 Nov 2010) | 30 lines
  
  Merged revisions 294904 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r294904 | jpeeler | 2010-11-12 14:51:15 -0600 (Fri, 12 Nov 2010) | 23 lines
    
    Merged revisions 294903 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12 Nov 2010) | 16 lines
      
      Fix regression causing abort in voicemail after opening a mailbox with no mesgs.
      
      In order to be more safe, some error handling code was changed to respect more
      error conditions including the potential memory allocation failure for deleted
      and heard message tracking introduced in 293004. However, last_message_index
      returns -1 for zero messages (perhaps as expected) and was triggering the
      stricter error checking. Because last_message_index is only called directly
      in one place, just return 0 from open_mailbox (for file based storage) when no
      messages are detected unless a real error has occurred.
      
      (closes issue #18240)
      Reported by: leobrown
      Patches: 
            bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585)
      Tested by: pabelanger
    ........
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  r294823 | rmudgett | 2010-11-11 20:45:22 -0600 (Thu, 11 Nov 2010) | 25 lines
  
  Merged revisions 294822 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r294822 | rmudgett | 2010-11-11 20:44:12 -0600 (Thu, 11 Nov 2010) | 18 lines
    
    Merged revisions 294821 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11 Nov 2010) | 11 lines
      
      Asterisk is getting a "No D-channels available!" warning message every 4 seconds.
      
      Asterisk is just whining too much with this message: "No D-channels
      available!  Using Primary channel XXX as D-channel anyway!".
      
      Filtered the message so it only comes out once if there is no D channel
      available without an intervening D channel available period.
      
      (closes issue #17270)
      Reported by: jmls
    ........
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  r294745 | russell | 2010-11-11 16:17:57 -0600 (Thu, 11 Nov 2010) | 6 lines
  
  Remove CCSS architecture PDF.
  
  It has been moved to:
  
  https://wiki.asterisk.org/wiki/display/AST/CCSS+Architecture
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  r294740 | russell | 2010-11-11 16:13:38 -0600 (Thu, 11 Nov 2010) | 11 lines
  
  Remove most of the contents of the doc dir in favor of the wiki content.
  
  This merge does the following things:
  
   * Removes most of the contents from the doc/ directory in favor
     of the wiki - http://wiki.asterisk.org/
  
   * Updates the build_tools/prep_tarball script to know how to export
     the contents of the wiki in both PDF and plain text formats so that
     the documentation is still included in Asterisk release tarballs.
........


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  r294734 | jpeeler | 2010-11-11 15:58:25 -0600 (Thu, 11 Nov 2010) | 32 lines
  
  Merged revisions 294733 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r294733 | jpeeler | 2010-11-11 15:57:22 -0600 (Thu, 11 Nov 2010) | 25 lines
    
    Merged revisions 294688 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010) | 18 lines
      
      Fix problem with qualify option packets for realtime peers never stopping.
      
      The option packets not only never stopped, but if a realtime peer was not in
      the peer list multiple options dialogs could accumulate over time. This
      scenario has the potential to progress to the point of saturating a link just
      from options packets. The fix was to ensure that the poke scheduler checks to
      see if a peer is in the peer list before continuing to poke. The reason a peer
      must be in the peer list to be able to properly manage an options dialog is
      because otherwise the call pointer is lost when the peer is regenerated from
      the database, which is how existing qualify dialogs are detected.
      
      (closes issue #16382)
      (closes issue #17779)
      Reported by: lftsy
      Patches: 
            bug16382-3.patch uploaded by jpeeler (license 325)
      Tested by: zerohalo
    ........
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  r294605 | tilghman | 2010-11-10 17:26:39 -0600 (Wed, 10 Nov 2010) | 2 lines
  
  Fixing the Mac OS X build (bamboo warning)
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  r294569 | tilghman | 2010-11-10 17:13:37 -0600 (Wed, 10 Nov 2010) | 8 lines
  
  Properly queue files with inotify(7).
  
  (closes issue #18089)
   Reported by: abelbeck
   Patches: 
         20101021__issue18089.diff.txt uploaded by tilghman (license 14)
   Tested by: tilghman
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  r294535 | russell | 2010-11-10 08:14:51 -0600 (Wed, 10 Nov 2010) | 5 lines
  
  Tweak a couple of CLI commands back to their original form.
  
  The "module" in this case is two parts, so there are two words before
  the verb of the CLI command.
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  r294501 | russell | 2010-11-10 06:46:27 -0600 (Wed, 10 Nov 2010) | 14 lines
  
  Merged revisions 294500 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r294500 | russell | 2010-11-10 06:41:41 -0600 (Wed, 10 Nov 2010) | 7 lines
    
    Improve a debug message to be more readable and consistent.
    
    (closes issue #18282)
    Reported by: klaus3000
    Patches:
          ast_devstate2str-patch.txt uploaded by klaus3000 (license 65)
  ........
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  r294466 | rmudgett | 2010-11-09 16:46:45 -0600 (Tue, 09 Nov 2010) | 1 line
  
  Allow ast_do_masquerade() failure to be reported again.
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  r294430 | tilghman | 2010-11-09 14:33:05 -0600 (Tue, 09 Nov 2010) | 15 lines
  
  Merged revisions 294429 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
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    r294429 | tilghman | 2010-11-09 14:27:23 -0600 (Tue, 09 Nov 2010) | 8 lines
    
    Detect GMime properly on systems where gmime flags and libs are configured with pkg-config.
    
    (closes issue #16155)
     Reported by: jcollie
     Patches: 
           20100917__issue16155.diff.txt uploaded by tilghman (license 14)
     Tested by: tilghman
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  r294349 | rmudgett | 2010-11-09 10:55:32 -0600 (Tue, 09 Nov 2010) | 17 lines
  
  Analog lines do not transfer CONNECTED LINE or execute the interception macros.
  
  Add connected line update for sig_analog transfers and simplify the
  corresponding sig_pri and chan_misdn transfer code.
  
  Note that if you create a three-way call in sig_analog before transferring
  the call, the distinction of the caller/callee interception macros make
  little sense.  The interception macro writer needs to be prepared for
  either caller/callee macro to be executed.  The current implementation
  swaps which caller/callee interception macro is executed after a three-way
  call is created.
  
  Review:	https://reviewboard.asterisk.org/r/996/
  
  JIRA ABE-2589
  JIRA SWP-2372
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  r294313 | jpeeler | 2010-11-08 16:32:13 -0600 (Mon, 08 Nov 2010) | 9 lines
  
  Merged revisions 294312 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
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    r294312 | jpeeler | 2010-11-08 16:30:49 -0600 (Mon, 08 Nov 2010) | 1 line
    
    add missing unlock not present in 294277
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  r294278 | jpeeler | 2010-11-08 15:59:45 -0600 (Mon, 08 Nov 2010) | 23 lines
  
  Merged revisions 294277 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08 Nov 2010) | 16 lines
    
    Fix playback failure when using IAX with the timerfd module.
    
    To fix this issue the alert pipe will now be used when the timerfd module is
    in use. There appeared to be a race that was not solved by adding locking in the
    timerfd module, but needed to be there anyway. The race was between the timer
    being put in non-continuous mode in ast_read on the channel thread and the IAX 
    frame scheduler queuing a frame which would enable continuous mode before the
    non-continuous mode event was read. This race for now is simply avoided.
    
    (closes issue #18110)
    Reported by: tpanton
    Tested by: tpanton
    
    I put tested by tpanton because it was tested on his hardware. Thanks for the
    remote access to debug this issue!
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  r294243 | mnicholson | 2010-11-08 14:56:30 -0600 (Mon, 08 Nov 2010) | 15 lines
  
  Merged revisions 294242 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r294242 | mnicholson | 2010-11-08 14:50:21 -0600 (Mon, 08 Nov 2010) | 8 lines
    
    Go off hold when we get an empty reinvite telling us to.
    
    (closes issue 0014448)
    Reported by: frawd
    
    (closes issue #17878)
    Reported by: frawd
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  r294207 | twilson | 2010-11-08 13:56:10 -0600 (Mon, 08 Nov 2010) | 2 lines
  
  Set a default waittime, and make sure to convert it to milliseconds
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  r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines
  
  valgrind reported references to freed memory during a mISDN hangup collision.
  
  Bad things have been happening in chan_misdn because the chan_misdn
  channel private struct chan_list is not protected from reentrancy.  Hangup
  collisions have be causing read and write accesses to freed memory.
  
  Converted chan_misdn struct chan_list to an ao2 object for its reference
  counting feature.
  
  **********
  Removed an impediment to converting chan_list to an ao2 object.
  
  The use of the other_ch member in chan_list is shaky at best.  It is set
  if the incoming and outgoing call legs are mISDN.  The use of the other_ch
  member goes against the Asterisk architecture and can even cause problems.
  
  1) It is used to disable echo cancellation.  This could be bad if the call
  is forked and the winning call leg is not mISDN or the winning call leg is
  not the last mISDN channel called by the fork.  The other_ch would become
  a dangling pointer.
  
  2) It is used when the far end is alerting to hear the far end's inband
  audio instead of Asterisk's generated ringback tone.  This is bad if the
  call is forked.  You would only hear the last forked mISDN channel and it
  may not be ringing yet.
  
  The other_ch would become a dangling pointer if the call is later
  transferred.
  **********
  
  JIRA SWP-2423
  JIRA ABE-2614
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  r294084 | bbryant | 2010-11-05 18:03:11 -0400 (Fri, 05 Nov 2010) | 9 lines
  
  Fixed deadlock avoidance issues while locking channel when adding the
  Max-Forwards header to a request.
  
  (closes issue #17949)
  (closes issue #18200)
  Reported by: bwg
  
  Review: https://reviewboard.asterisk.org/r/997/
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RFC3261 section 12 about dialog creation says an INVITE transaction
results in an established dialog once it receives the 200 OK response.
It is possible to receive multiple differing 200 OK responses for a
single outbound INVITE Request, and this should result in establishing
multiple dialogs.

This patch allows for all differing 200 OK responses to an INVITE request
to establish a separate dialog, but only the first dialog is kept. All other
resulting dialogs from the initial request are immediately ACKed and then
immediately terminated with a BYE request.

Review: https://reviewboard.asterisk.org/r/946/



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