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  1. May 03, 2011
  2. Apr 22, 2011
  3. Apr 21, 2011
  4. Apr 18, 2011
    • 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
  5. Apr 14, 2011
  6. Apr 13, 2011
  7. Mar 18, 2011
  8. Mar 10, 2011
  9. Mar 08, 2011
  10. Mar 04, 2011
  11. 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
  12. Feb 14, 2011
  13. Feb 04, 2011
  14. Feb 03, 2011
  15. Feb 02, 2011
  16. Feb 01, 2011
  17. Jan 31, 2011
  18. Jan 30, 2011
  19. Jan 29, 2011
  20. Jan 27, 2011
  21. Jan 26, 2011
  22. Jan 24, 2011
    • Matthew Nicholson's avatar
      According to section 19.1.2 of RFC 3261: · e706b570
      Matthew Nicholson authored
        For each component, the set of valid BNF expansions defines exactly
        which characters may appear unescaped.  All other characters MUST be
        escaped.
      
      This patch modifies ast_uri_encode() to encode strings in line with this recommendation.  This patch also adds an ast_escape_quoted() function which escapes '"' and '\' characters in quoted strings in accordance with section 25.1 of RFC 3261.  The ast_uri_encode() function has also been modified to take an ast_flags struct describing the set of rules it should use when escaping characters to allow for it to escape SIP URIs in addition to HTTP URIs and other types of URIs or variations of those two URI types in the future.
      
      The ast_uri_decode() function has also been modified to accept an ast_flags struct describing the set of rules to use when decoding to enable decoding '+' as ' ' in legacy http URLs.
      
      The unit tests for these functions have also been updated.
      
      ABE-2705
      
      Review: https://reviewboard.asterisk.org/r/1081/
      
      
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      e706b570
  23. Jan 19, 2011
  24. Jan 07, 2011
  25. Jan 05, 2011
  26. Jan 04, 2011
  27. Dec 22, 2010
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