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  1. Apr 27, 2011
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  15. Apr 18, 2011
    • Richard Mudgett's avatar
      Problems with ISDN MWI to phones. · 37274c73
      Richard Mudgett authored 
      The "controlling user number" is always the number of the voice mail box
which is identical with the subscriber number itself.  This number which
is listed in the ISDN phone MWI menu cannot be called back to contact the
voice mail box.  The controlling user number should be made configurable.

JIRA ABE-2738
JIRA SWP-2846


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314116 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      37274c73
    • Richard Mudgett's avatar
      Merged revisions 314069 via svnmerge from · 0a5c2d83
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r314069 | rmudgett | 2011-04-18 11:10:10 -0500 (Mon, 18 Apr 2011) | 22 lines
  
  The AsyncAGI command loop is lax in the value it returns for the return status.
  
  * Return correct status: SUCCESS/FAILED/HANGUP.  Previously, abnormal
  exits from the command loop such as hangup would return SUCCESS.
  
  * The "asyncagi break" command now returns SUCCESS and is now the only way
  to break the command loop with that status.  Previously, it returned
  FAILED.
  
  * The AMI event AsyncAGI End is no longer sent if the AsyncAGI Start event
  is not sent.  Previously, this happened because of an error setting up the
  AGI pipes.
  
  * All executed AGI commands now get an AsyncAGI Exec result event.
  Previously, if the command returned failure (because of hangup), the
  command loop just exited with FAILURE and did not send the AsyncAGI Exec
  result event.
  
  * Makes sure that the channel frame queue is empty on hangup.
  
  Review: https://reviewboard.asterisk.org/r/1183/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      0a5c2d83
    • Richard Mudgett's avatar
      Merged revisions 314068 via svnmerge from · 7c4fc0f0
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r314068 | rmudgett | 2011-04-18 11:02:12 -0500 (Mon, 18 Apr 2011) | 7 lines
  
  Unclear code in app_dial.c.
  
  Make code formatting clear.
  
  (closes issue #19134)
  Reported by: oej
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      7c4fc0f0
    • David Vossel's avatar
      Merged revisions 314067 via svnmerge from · 642249c3
      David Vossel authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r314067 | dvossel | 2011-04-18 10:23:45 -0500 (Mon, 18 Apr 2011) | 22 lines
  
  Remove the need for deadlock avoidance in chan_sip do_monitor.
  
  Deadlock avoidance between the sip pvt and the pvt->owner is
  very difficult.  Now that channel's are ao2 objects, this complication
  is no longer necessary.  It turns out the pvt's msg queue only
  exists because of deadlock avoidance (when deadlock avoidance fails
  msgs were added to a queue to be processed later), so this goes away as well.
  
  The technique used in the new sip_lock_pvt_full() function should
  be used as a template for replacing all locations where deadlock
  avoidance occurs between a channel tech_pvt and the pvt's owner.
  My hope is that this will begin a reversal of the invalid channel
  driver locking architecture we have been using for so long. 
  
  This patch also resolves an issue where the pvt->owner gets
  unlocked during processing the msg queue.
  
  (closes issue #18690)
  Reported by: dvossel
  
  Review: https://reviewboard.asterisk.org/r/1182/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      642249c3
    • David Vossel's avatar
      Merged revisions 314017 via svnmerge from · 4b454910
      David Vossel authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r314017 | dvossel | 2011-04-18 08:41:06 -0500 (Mon, 18 Apr 2011) | 17 lines
  
  sip codec negotiation of dynamic rtp payloads error fix
  
  This patch fixes how chan_sip handles dynamic rtp payload types
  it does not understand.  At the moment if a dynamic payload's mime
  type does not match one we understand, the payload does not get
  removed from our payload table.  As a result of this, the payload
  is set to whatever dynamic codec we use internally for that payload
  number on outgoing INVITES.  This is incorrect.
  
  This patch fixes this by properly checking the rtpmap set function's
  return code to make sure it was found.  The function can return both
  -1 and -2 depending on the source of the mismatch.  We were just
  checking -1 explicitly.
  
  Review: https://reviewboard.asterisk.org/r/1169/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      4b454910
  17. Apr 17, 2011