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20 authors
  1. Sep 28, 2011
  3. Sep 27, 2011
  5. Sep 26, 2011
    • Richard Mudgett's avatar
      Merged revisions 337974 via svnmerge from · 55b70ae6
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

................
  r337974 | rmudgett | 2011-09-26 14:35:23 -0500 (Mon, 26 Sep 2011) | 37 lines
  
  Merged revisions 337973 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r337973 | rmudgett | 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011) | 30 lines
    
    Fix deadlock when using dummy channels.
    
    Dummy channels created by ast_dummy_channel_alloc() should be destoyed by
    ast_channel_unref().  Using ast_channel_release() needlessly grabs the
    channel container lock and can cause a deadlock as a result.
    
    * Analyzed use of ast_dummy_channel_alloc() and made use
    ast_channel_unref() when done with the dummy channel.  (Primary reason for
    the reported deadlock.)
    
    * Made app_dial.c:dial_exec_full() not call ast_call() holding any channel
    locks.  Chan_local could not perform deadlock avoidance correctly.
    (Potential deadlock exposed by this issue.  Secondary reason for the
    reported deadlock since the held lock was part of the deadlock chain.)
    
    * Fixed some uses of ast_dummy_channel_alloc() not checking the returned
    channel pointer for failure.
    
    * Fixed some potential chan=NULL pointer usage in func_odbc.c.  Protected
    by testing the bogus_chan value.
    
    * Fixed needlessly clearing a 1024 char auto array when setting the first
    char to zero is enough in manager.c:action_getvar().
    
    (closes issue ASTERISK-18613)
    Reported by: Thomas Arimont
    Patches:
          jira_asterisk_18613_v1.8.patch (license #5621) patch uploaded by rmudgett
    Tested by: Thomas Arimont
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      55b70ae6
  7. Sep 23, 2011
  9. Sep 22, 2011
  11. Sep 21, 2011
  13. Sep 20, 2011
    • Matthew Jordan's avatar
      Merged revisions 337120 via svnmerge from · e218748a
      Matthew Jordan authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

................
  r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines
  
  Merged revisions 337118 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines
    
    Fix for incorrect voicemail duration in external notifications
    
    This patch fixes an issue where the voicemail duration was being reported
    with a duration significantly less than the actual sound file duration.
    Voicemails that contained mostly silence were reporting the duration of
    only the sound in the file, as opposed to the duration of the file with
    the silence.  This patch fixes this by having two durations reported in
    the __ast_play_and_record family of functions - the sound_duration and the
    actual duration of the file.  The sound_duration, which is optional, now
    reports the duration of the sound in the file, while the actual full duration
    of the file is reported in the duration parameter.  This allows the voicemail
    applications to use the sound_duration for minimum duration checking, while
    reporting the full duration to external parties if the voicemail is kept.
    
    (issue ASTERISK-2234)
    (closes issue ASTERISK-16981)
    Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH
    Tested by: Matt Jordan
    
    Review: https://reviewboard.asterisk.org/r/1443
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      e218748a
    • Richard Mudgett's avatar
      Merged revisions 337119 via svnmerge from · 1313c128
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

........
  r337119 | rmudgett | 2011-09-20 17:47:45 -0500 (Tue, 20 Sep 2011) | 16 lines
  
  Fix crash with STRREPLACE function.
  
  The ast_func_read() function calls the .read2 callback with the len
  parameter set to zero indicating no size restrictions on the supplied
  ast_str buffer.  The value was used to dimension a local starts[] array
  with the array subsequently used.
  
  * Reworked the strreplace() function to perform the string replacement in
  a straight forward manner.  Eliminated the need for the starts[] array.
  
  (closes issue ASTERISK-18545)
  Reported by: Federico Alves
  Patches:
        jira_asterisk_18545_v10.patch (license #5621) patch uploaded by rmudgett
  Tested by: rmudgett, Federico Alves
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337123 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      1313c128
    • Richard Mudgett's avatar
      Updated 10 merge property. · 38a7c688
      Richard Mudgett authored 
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      38a7c688
    • Richard Mudgett's avatar
      Restore branch-10 merge properties. · bbafe3bd
      Richard Mudgett authored 
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      bbafe3bd
    • Leif Madsen's avatar
      Merged revisions 337115 via svnmerge from · 6b715d8f
      Leif Madsen authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r337115 | lmadsen | 2011-09-20 17:18:25 -0500 (Tue, 20 Sep 2011) | 7 lines
  
  Update RedHat Init script to work with Heartbeat.
  
  The current RedHat init script was not LSB compatible. This change will make it LSB compatible so that
  it can work correctly with Heartbeat.
  
  (Closes issue ASTERISK-18253)
  Reported by: c0rnoTa
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337117 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      6b715d8f
    • Kinsey Moore's avatar
      Merged revisions 337062 via svnmerge from · 486b6042
      Kinsey Moore authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

................
  r337062 | kmoore | 2011-09-20 16:05:01 -0500 (Tue, 20 Sep 2011) | 18 lines
  
  Merged revisions 337061 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r337061 | kmoore | 2011-09-20 16:04:11 -0500 (Tue, 20 Sep 2011) | 11 lines
    
    Make CANMATCH with the new pattern match engine behave more like the old one
    
    When checking an extension for E_CANMATCH using the new extension matching
    algorithm, an exact match was not returned as a possible match resulting in the
    queue failing to allow a caller to exit on DTMF.  This removes the requirement
    that an extension be longer than acquired digits for an E_CANMATCH operation
    to succeed.
    
    (closes issue ASTERISK-18044)
    Review: https://reviewboard.asterisk.org/r/1367/
  ........
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      486b6042
    • Richard Mudgett's avatar
      Merged revisions 337008 via svnmerge from · 7fe331fd
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

................
  r337008 | rmudgett | 2011-09-20 14:12:24 -0500 (Tue, 20 Sep 2011) | 22 lines
  
  Merged revisions 337007 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r337007 | rmudgett | 2011-09-20 14:10:30 -0500 (Tue, 20 Sep 2011) | 15 lines
    
    Check if a channel was created before using the pointer in sig_ss7_new_ast_channel().
    
    Fixes the crash in ASTERISK-17955 gdb-11918.txt backtrace.
    
    * Added some missing libss7 access lock protection.
    
    * Prevent cancelling the ss7_linkset() thread at inoportune times just
    like the pri_dchannel() thread.
    
    (issue ASTERISK-17955)
    Reported by: Ian M Sherman
    Patches:
          jira_asterisk_17955_v1.8.patch (license #5621) patch uploaded by rmudgett
          (attached to related ASTERISK-17966)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337009 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      7fe331fd
    • Richard Mudgett's avatar
      Merged revisions 336978 via svnmerge from · b3768f04
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

................
  r336978 | rmudgett | 2011-09-20 13:14:40 -0500 (Tue, 20 Sep 2011) | 28 lines
  
  Merged revisions 336977 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r336977 | rmudgett | 2011-09-20 13:12:17 -0500 (Tue, 20 Sep 2011) | 21 lines
    
    Fix deadlock from not releasing SS7 linkset lock.
    
    sig_ss7_hangup() failed to release the SS7 linkset lock if the call had
    the alreadyhungup flag set.
    
    * Made unlock the SS7 linkset lock in sig_ss7_hangup() if the
    alreadyhungup flag is set.
    
    * Made ss7_start_call() not hold any locks while creating the channel for
    an incoming call to prevent deadlock.
    
    * Made ss7_grab() a void function, since it could never fail, to simplify
    calling code.
    
    * Made obtain the channel lock to do softhangup in some places.
    
    Patches:
          jira_ast_668_v1.8.patch (license #5621) patch uploaded by rmudgett
    
    JIRA AST-668
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      b3768f04
    • Gregory Nietsky's avatar
      Merged revisions 336936 via svnmerge from · 8493c463
      Gregory Nietsky authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

........
  r336936 | irroot | 2011-09-20 18:51:59 +0200 (Tue, 20 Sep 2011) | 14 lines
  
  
  Allow Setting Auth Tag Bit length Based on invite or config option
  
  Update the SIP SRTP API to allow use of 32 or 80 bit taglen.
  Curently only 80 bit is supported.
  
  The outgoing invite will use the taglen of the incoming invite preventing
  one-way audio.
  
  (Closes issue ASTERISK-17895)
  
  Review: https://reviewboard.asterisk.org/r/1173/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336937 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      8493c463
    • Russell Bryant's avatar
      Merged revisions 336878 via svnmerge from · 14d3f891
      Russell Bryant authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

................
  r336878 | russell | 2011-09-19 20:03:55 -0500 (Mon, 19 Sep 2011) | 43 lines
  
  Merged revisions 336877 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r336877 | russell | 2011-09-19 19:56:20 -0500 (Mon, 19 Sep 2011) | 36 lines
    
    Fix crashes in ast_rtcp_write().
    
    This patch addresses crashes related to RTCP handling.  The backtraces just
    show a crash in ast_rtcp_write() where it appears that the RTP instance is no
    longer valid.  There is a race condition with scheduled RTCP transmissions and
    the destruction of the RTP instance.  This patch utilizes the fact that
    ast_rtp_instance is a reference counted object and ensures that it will not get
    destroyed while a reference is still around due to scheduled RTCP
    transmissions.
    
    RTCP transmissions are scheduled and executed from the chan_sip scheduler
    context.  This scheduler context is processed in the SIP monitor thread.  The
    destruction of an RTP instance occurs when the associated sip_pvt gets
    destroyed (which happens when the sip_pvt reference count reaches 0).  However,
    the SIP monitor thread is not the only thread that can cause a sip_pvt to get
    destroyed.  The sip_hangup function, executed from a channel thread, also
    decrements the reference count on a sip_pvt and could cause it to get
    destroyed.
    
    While this is being changed anyway, the patch also removes calling
    ast_sched_del() from within the RTCP scheduler callback.  It's not helpful.
    Simply returning 0 prevents the callback from being rescheduled.
    
    (closes issue ASTERISK-18570)
    
    Related issues that look like they are the same problem:
    
    (issue ASTERISK-17560)
    (issue ASTERISK-15406)
    (issue ASTERISK-15257)
    (issue ASTERISK-13334)
    (issue ASTERISK-9977)
    (issue ASTERISK-9716)
    
    Review: https://reviewboard.asterisk.org/r/1444/
  ........
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      14d3f891
  15. Sep 19, 2011