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20 authors
  1. Mar 04, 2011
    • Richard Mudgett's avatar
      Merged revisions 309445 via svnmerge from · 928ec2b9
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r309445 | rmudgett | 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) | 46 lines
  
  Get real channel of a DAHDI call.
  
  Starting with Asterisk v1.8, the DAHDI channel name format was changed for
  ISDN calls to: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
  
  There were several reasons that the channel name had to change.
  
  1) Call completion requires a device state for ISDN phones.  The generic
  device state uses the channel name.
  
  2) Calls do not necessarily have B channels.  Calls placed on hold by an
  ISDN phone do not have B channels.
  
  3) The B channel a call initially requests may not be the B channel the
  call ultimately uses.  Changes to the internal implementation of the
  Asterisk master channel list caused deadlock problems for chan_dahdi if it
  needed to change the channel name.  Chan_dahdi no longer changes the
  channel name.
  
  4) DTMF attended transfers now work with ISDN phones because the channel
  name is "dialable" like the chan_sip channel names.
  
  For various reasons, some people need to know which B channel a DAHDI call
  is using.
  
  * Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and
  CHANNEL(dahdi_type) so the dialplan can determine the B channel currently
  in use by the channel.  Use CHANNEL(no_media_path) to determine if the
  channel even has a B channel.
  
  * Added AMI event DAHDIChannel to associate a DAHDI channel with an
  Asterisk channel so AMI applications can passively determine the B channel
  currently in use.  Calls with "no-media" as the DAHDIChannel do not have
  an associated B channel.  No-media calls are either on hold or
  call-waiting.
  
  (closes issue #17683)
  Reported by: mrwho
  Tested by: rmudgett
  
  (closes issue #18603)
  Reported by: arjankroon
  Patches:
        issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664)
  Tested by: stever28, rmudgett
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      928ec2b9
    • David Ruggles's avatar
      Merged revisions 309403 via svnmerge from · 3cda82a3
      David Ruggles authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r309403 | diruggles | 2011-03-03 20:50:44 -0500 (Thu, 03 Mar 2011) | 23 lines
  
  Merged revisions 309356 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r309356 | diruggles | 2011-03-03 19:42:28 -0500 (Thu, 03 Mar 2011) | 16 lines
    
    Merged revisions 309355 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r309355 | diruggles | 2011-03-03 19:34:13 -0500 (Thu, 03 Mar 2011) | 9 lines
      
      fix small memory leak
      
      fix small memory leak caused by a string allocation that wasn't freed
      
      (closes issue #18907)
      Reported by: andy11
      Patches: 
            asterisk_trunk-app_externalivr-leak.patch uploaded by andy11 (license 1224)
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      3cda82a3
  3. Mar 02, 2011
  5. Mar 01, 2011
  7. Feb 28, 2011
  9. Feb 25, 2011
  11. Feb 24, 2011
  13. Feb 23, 2011
  15. Feb 22, 2011
    • David Vossel's avatar
      Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd... · d760e81f
      David Vossel authored 
      Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff

-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
   using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
   and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate.  This allows
   for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
   updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.

-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c

Review: https://reviewboard.asterisk.org/r/1104/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      d760e81f
    • Andrew Latham's avatar
      Use ast_debug for console logging · 736133f8
      Andrew Latham authored 
      Guessed the log levels based on info that level 3
is the soft roof.  Can we create a page / document
to define the levels?



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      736133f8
  17. Feb 21, 2011
  19. Feb 19, 2011
  21. Feb 18, 2011
  23. Feb 17, 2011
  25. Feb 16, 2011
  27. Feb 15, 2011
  29. Feb 14, 2011
  31. Feb 13, 2011
  33. Feb 12, 2011