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20 authors
  1. May 03, 2011
  3. Apr 26, 2011
  5. Apr 21, 2011
  7. Apr 20, 2011
  9. 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
    • 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
  11. Apr 11, 2011
  13. Apr 01, 2011
  15. Mar 31, 2011
  17. Mar 11, 2011
  19. Mar 07, 2011
  21. Mar 04, 2011
  23. Feb 28, 2011
  25. 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
  27. Feb 15, 2011
    • Richard Mudgett's avatar
      Merged revisions 307879 via svnmerge from · b2ef13cb
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r307879 | rmudgett | 2011-02-15 10:13:55 -0600 (Tue, 15 Feb 2011) | 37 lines
  
  No response sent for SIP CC subscribe/resubscribe request.
  
  Asterisk does not send a response if we try to subscribe for call
  completion after we have received a 180 Ringing.  You can only subscribe
  for call completion when the call has been cleared.
  
  When we receive the 180 Ringing, for this call, its call-completion state
  is 'CC_AVAILABLE'.  If we then send a subscribe message to Asterisk, it
  trys to change the call-completion state to 'CC_CALLER_REQUESTED'.
  Because this is an invalid state change, it just ignores the message.  The
  only state Asterisk will accept our subscribe message is in the
  'CC_CALLER_OFFERED' state.
  
  Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears
  the call by sending a CANCEL.
  
  Asterisk should always send a response.  Even if its a negative one.
  
  
  The fix is to allow for the CCSS core to notify a CC agent that a failure
  has occurred when CC is requested.  The "ack" callback is replaced with a
  "respond" callback.  The "respond" callback has a parameter indicating
  either a successful response or a specific type of failure that may need
  to be communicated to the requester.
  
  (closes issue #18336)
  Reported by: GeorgeKonopacki
  Tested by: mmichelson, rmudgett
  
  JIRA SWP-2633
  
  (closes issue #18337)
  Reported by: GeorgeKonopacki
  Tested by: mmichelson
  
  JIRA SWP-2634
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      b2ef13cb
  29. Feb 10, 2011
  31. Feb 07, 2011
  33. Feb 04, 2011
  35. Feb 03, 2011
  37. Jan 31, 2011
  39. Jan 26, 2011
  41. Jan 24, 2011
    • Russell Bryant's avatar
      Merged revisions 303549 via svnmerge from · 09213439
      Russell Bryant authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r303549 | russell | 2011-01-24 14:51:37 -0600 (Mon, 24 Jan 2011) | 45 lines
  
  Merged revisions 303548 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r303548 | russell | 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) | 38 lines
    
    Merged revisions 303546 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines
      
      Fix channel redirect out of MeetMe() and other issues with channel softhangup.
      
      Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped
      working properly.  This issue includes a patch that resolves the issue by
      removing a call to ast_check_hangup() from app_meetme.c.  I left that in my
      patch, as it doesn't need to be there.  However, the rest of the patch fixes
      this problem with or without the change to app_meetme.
      
      The key difference between what happens before and after this patch is the
      effect of the END_OF_Q control frame.  After END_OF_Q is hit in ast_read(),
      ast_read() will return NULL.  With the ast_check_hangup() removed, app_meetme
      sees this which causes it to exit as intended.  Checking ast_check_hangup()
      caused app_meetme to exit earlier in the process, and the target of the
      redirect saw the condition where ast_read() returned NULL.
      
      Removing ast_check_hangup() works around the issue in app_meetme, but doesn't
      solve the issue if another application did the same thing.  There are also
      other edge cases where if an application finishes at the same time that a
      redirect happens, the target of the redirect will think that the channel hung
      up.  So, I made some changes in pbx.c to resolve it at a deeper level.  There
      are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to
      abort the hangup process.  My patch extends this to remove the END_OF_Q frame
      from the channel's read queue, making the "abort hangup" more complete.  This
      same technique was used in every place where a softhangup flag was cleared.
      
      (closes issue #18585)
      Reported by: oej
      Tested by: oej, wedhorn, russell
      
      Review: https://reviewboard.asterisk.org/r/1082/
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      09213439
    • 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
  43. Jan 19, 2011
  45. Dec 20, 2010
  47. Dec 18, 2010
  49. Dec 12, 2010
  51. Dec 10, 2010
  53. Dec 03, 2010
    • Matthew Nicholson's avatar
      Merged revisions 297157,297486,297495 via svnmerge from · 23d106b8
      Matthew Nicholson authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r297157 | mnicholson | 2010-12-01 13:47:33 -0600 (Wed, 01 Dec 2010) | 2 lines
  
  Changed some NOTICE and WARNING messages to DEBUG messages.
........
  r297486 | mnicholson | 2010-12-02 15:30:47 -0600 (Thu, 02 Dec 2010) | 6 lines
  
  Add support for reserving a fax session before answering the channel.
  
  Note: this change breaks ABI compatibility.
  
  FAX-217
........
  r297495 | mnicholson | 2010-12-03 09:21:52 -0600 (Fri, 03 Dec 2010) | 4 lines
  
  Print a DEBUG message instead of a WARNING message when the selected fax tech does not support reserving sessions.
  
  Answer the channel before quering it for t.38 support.  This is necessary for the query to work properly over local channels.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297496 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      23d106b8
  55. Dec 01, 2010
  57. Nov 30, 2010
  59. Nov 29, 2010
  61. Nov 27, 2010