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  19. Nov 24, 2010
    • Andrew Parisio's avatar
      Meetme use voicemail greet for join/leave announce · 935930d8
      Andrew Parisio authored 
      Added option v(mailbox@[context]) which tells MeetMe where to look for a users greet file.  If one does not exist it clears the v option and defers to the functionality of i/I as/if set by the MeetMe() command.

Review: https://reviewboard.asterisk.org/r/1009/
(closes issue #18297)
Reported by: parisioa
Patches:
	meetme_final_patch_v.diff uploaded by parisioa (license 1153)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      935930d8
    • Russell Bryant's avatar
      Merged revisions 296230 via svnmerge from · 10f375f8
      Russell Bryant authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r296230 | russell | 2010-11-24 17:29:44 -0600 (Wed, 24 Nov 2010) | 20 lines
  
  Merged revisions 296221 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r296221 | russell | 2010-11-24 17:28:19 -0600 (Wed, 24 Nov 2010) | 13 lines
    
    Merged revisions 296213 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010) | 6 lines
      
      Make Asterisk less crashy.
      
      Since we might not put a new translation path on the channel, go ahead and
      set it to NULL right after destroying the old one to ensure we don't try
      to free an invalid translation path later on.
    ........
  ................
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      10f375f8
    • Richard Mudgett's avatar
      Merged revisions 296167 via svnmerge from · ccdc417a
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r296167 | rmudgett | 2010-11-24 16:49:48 -0600 (Wed, 24 Nov 2010) | 57 lines
  
  Merged revisions 296166 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r296166 | rmudgett | 2010-11-24 16:42:45 -0600 (Wed, 24 Nov 2010) | 50 lines
    
    Merged revisions 296165 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24 Nov 2010) | 43 lines
      
      Oneway audio to SIP phone from FXS port after FXS port gets a CallWaiting pip.
      
      The FXS connected phone has to have CW/CID support to fail, as it will
      send back a DTMF 'A' or 'D' when it's ready to receive CallerID.  A normal
      phone with no CID never fails.  Also the SIP phone does not hear MOH when
      the CW call is answered.
      
      The DTMF end frame is suppressed when the phone acknowledges the CW signal
      for CID.  The problem is the DTMF begin frame needs to be suppressed as
      well.  The DTMF begin frame is causing SIP to start sending the DTMF RTP
      frames.  Since the DTMF end frame is suppressed, SIP will not stop sending
      those DTMF RTP packets.
      
      * Suppress the DTMF begin and end frames when the channel driver is
      looking for DTMF digits.
      
      * Fixed a couple issues caused by not cleaning up the CID spill if you
      answer the CW call while it is sending the CID spill.
      
      * Fixed not sending CW/CID spill to the phone when the call is natively
      bridged.  (Fixed by not using native bridge if CW/CID is possible.)
      
      * Suppress received audio when sending CW/CID spills.  The other parties
      involved do not need to hear the CW/CID spills and may be confused if the
      CW call is for them.
      
      (closes issue #18129)
      Reported by: alecdavis
      Patches:
            issue_18129_v1.8_v3.patch uploaded by rmudgett (license 664)
      Tested by: alecdavis, rmudgett
      
      
      NOTE:
      
      * v1.4 does not have the main problem fixed by suppressing the DTMF start
      frames.  The other three items fixed are relevant.
      
      * If you really must restore native bridging between analog ports, you
      need to disable CW/CID either by configuring chan_dahdi.conf
      callwaitingcallerid=no or dialing *70 before dialing the number to
      temporarily disable CW.
    ........
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      ccdc417a
    • Russell Bryant's avatar
      Merged revisions 296084 via svnmerge from · ddd0ae53
      Russell Bryant authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r296084 | russell | 2010-11-24 14:23:46 -0600 (Wed, 24 Nov 2010) | 26 lines
  
  Merged revisions 296083 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r296083 | russell | 2010-11-24 14:23:11 -0600 (Wed, 24 Nov 2010) | 19 lines
    
    Merged revisions 296082 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010) | 12 lines
      
      Fix false reporting of an error by set_format().
      
      In the case that the native format was able to be changed to match the
      new requested format, the code proceeded to attempt to build a translation
      path, anyway.  The result would be NULL, since no translation path is
      necessary and resulted in this function thinking an error has occurred.
      This case is now specifically caught and no attempt to build a translation
      path is attempted.
      
      Thanks to our automated tests and bamboo.asterisk.org for catching this problem
      and making a whole lot of noise when things started failing.  :-)
    ........
  ................
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      ddd0ae53
    • Russell Bryant's avatar
      Merged revisions 296002 via svnmerge from · 712ba231
      Russell Bryant authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r296002 | russell | 2010-11-24 11:13:08 -0600 (Wed, 24 Nov 2010) | 52 lines
  
  Merged revisions 296001 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r296001 | russell | 2010-11-24 11:03:16 -0600 (Wed, 24 Nov 2010) | 45 lines
    
    Merged revisions 296000 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010) | 38 lines
      
      Handle failures building translation paths more effectively.
      
      The problem scenario occurred on a heavily loaded system that was using the
      codec_dahdi module and exceeded the hardware transcoding capacity.  The failure
      mode at that point was not good.  The report came in to us as an Asterisk
      lock-up.  The "core show locks" shows a ton of threads locked up (but no
      obvious deadlock).  Upon deeper investigation, when the system is in this
      state, the CPU was maxed out.  The CPU was being consumed by the Asterisk
      logger spewing messages on every audio frame for calls set up after transcoder
      capacity was reached.
      
      The purpose of this patch is to make Asterisk handle failures to create a
      translation path in a more graceful manner.  If we can't translate, then the
      call just needs to be dropped, as it's not going to work.  These are the
      changes:
      
      1) In set_format() of channel.c (which is called by set_read_format() and
      set_write_format()), it was ignoring if ast_translator_build_path() failed and
      returned NULL.  It now pays attention to that case and returns a result
      reflecting failure.  With this change in place, the bridging code will
      immediately detect a failure and end the bridge instead of proceeding to try to
      bridge frames that can't be translated and making channel drivers freak out by
      sending them frames in a format they weren't expecting.
      
      2) In ast_indicate_data() of channel.c, failure of ast_playtones_start() was
      ignored.  It is now reflected in the return value of the function.  This didn't
      turn out to have any affect on the bug, but seemed like a good change to leave
      in.
      
      3) In app_dial(), when only sending a call to a single endpoint, it will
      attempt to do some bridging of its own of early audio.  It uses
      make_compatible() when it's going to do this.  However, it ignored failure from
      make compatible.  So, even with the fix from #1, if there was early audio going
      through app_dial, there would still be a period of invalid frames passing
      through.  After detecting failure here, Dial() exits.
      
      ABE-2658
    ........
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      712ba231
  21. Nov 23, 2010
  23. Nov 22, 2010
  25. Nov 20, 2010
    • Richard Mudgett's avatar
      Merged revisions 295747 via svnmerge from · b1e7f85b
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  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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      b1e7f85b
    • Russell Bryant's avatar
      Merged revisions 295711 via svnmerge from · 9fbbdfb2
      Russell Bryant authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  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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      9fbbdfb2
  27. Nov 19, 2010
  29. Nov 18, 2010