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20 authors
  1. May 04, 2011
  3. May 03, 2011
  5. May 02, 2011
  7. Apr 27, 2011
  9. Apr 26, 2011
  11. Apr 21, 2011
  13. Apr 19, 2011
  15. Apr 18, 2011
  17. Apr 13, 2011
  19. Apr 11, 2011
    • Richard Mudgett's avatar
      Merged revisions 313368-313369 via svnmerge from · 663ed7fd
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r313368 | rmudgett | 2011-04-11 18:03:02 -0500 (Mon, 11 Apr 2011) | 2 lines
  
  Backport a restructuring change from trunk to make the next change stand out.
........
  r313369 | rmudgett | 2011-04-11 18:08:02 -0500 (Mon, 11 Apr 2011) | 13 lines
  
  Frames from the inbound channel should go to all outbound channels in app_dial.c.
  
  In app_dial.c:wait_for_answer() frames from the inbound channel should be
  sent to all outbound channels instead of only if there is just one
  outbound channel.
  
  Control frames like AST_CONTROL_CONNECTED_LINE need to be passed to all of
  the the outbound channels.  This can happen if a blond transfer is done by
  a remote switch on the inbound channel.
  
  JIRA AST-443
  JIRA SWP-2730
........


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      663ed7fd
  21. Apr 07, 2011
  23. Apr 05, 2011
  25. Apr 01, 2011
  27. Mar 28, 2011
  29. Mar 23, 2011
  31. Mar 22, 2011
  33. Mar 18, 2011
    • Jonathan Rose's avatar
      Adds an option to FollowMe that isn't useful for the bug it was made to solve.... · 18a6c3a4
      Jonathan Rose authored 
      Adds an option to FollowMe that isn't useful for the bug it was made to solve.  Still, due to the nature of FollowMe, it makes sense to have this option since it keeps apps bound to channels that would otherwise go away from being lost.

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      18a6c3a4
    • Richard Mudgett's avatar
      Merged revisions 311295 via svnmerge from · 4a8c7797
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r311295 | rmudgett | 2011-03-17 21:22:07 -0500 (Thu, 17 Mar 2011) | 35 lines
  
  Merged revision 310986 from
  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
  
  ..........
    r310986 | rmudgett | 2011-03-16 13:56:28 -0500 (Wed, 16 Mar 2011) | 28 lines
  
    Dial() o option broke when connected line feature added.
  
    The patch restores the o option behavior and adds the ability to specify
    the CallerID.  The Dial o and f options are complementary to each other.
    The o option stores the CallerID on the outgoing channel as the channel's
    CallerID.  The f option forces the CallerID sent by the outgoing channel.
  
    o(x) - The argument 'x' is optional.  If not present, then specify that
    the CallerID that was present on the *calling* channel be stored as the
    CallerID on the *called* channel.  This was the behavior of Asterisk 1.0
    and earlier.  If present, then specify the CallerID stored on the *called*
    channel.  Note that o(${CALLERID(all)}) is similar to option o without
    parameters.
  
    f(x) - The argument 'x' is optional and its presence changes the behavior
    of this option.  If not present, then force the outgoing CallerID on a
    call-forward or deflection to the dialplan extension for this Dial() using
    a dialplan 'hint'.  For example, some PSTNs do not allow CallerID to be
    set to anything other than the numbers assigned to you.  If present, then
    force the outgoing CallerID to 'x'.
  
    Patches:
  	jira_abe_2752_dial_fo_options.patch uploaded by rmudgett (license 664)
    Tested by: rmudgett
  
    JIRA ABE-2752
    JIRA SWP-3096
  ..........
................


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      4a8c7797
  35. Mar 17, 2011
  37. Mar 11, 2011
  39. Mar 10, 2011
  41. Mar 07, 2011
  43. Mar 04, 2011
  45. 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/


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      d760e81f
  47. Feb 15, 2011
  49. Feb 14, 2011
    • Tilghman Lesher's avatar
      Merged revisions 307750 via svnmerge from · 7800a1c3
      Tilghman Lesher authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r307750 | tilghman | 2011-02-14 00:50:23 -0600 (Mon, 14 Feb 2011) | 23 lines
  
  Calling a gosub routine defined in AEL from Dial/Queue ceased to work.
  
  A bug in AEL did not distinguish between the "s" extension generated by
  AEL and an "s" extension that was required to exist by the chan_dahdi
  (or another channel) that was not supplied with a starting extension.
  Therefore, AEL made incorrect assumptions about what commands were
  permissable in the context.  This was fixed by making AEL generate a
  different extension name.  However, Dial and Queue make additional
  assumptions about the name of the default gosub extension.  Therefore,
  they needed to be brought into line with a "macro" rendered by AEL (as
  a gosub), without breaking traditional dialplans written without the
  aid of AEL.
  
  Related to (issue #18480)
   Reported by: nivek
  
  (closes issue #18729)
   Reported by: kkm
   Patches: 
         20110209__issue18729.diff.txt uploaded by tilghman (license 14)
         018729-dial-queue-gosub-try3.patch uploaded by kkm (license 888)
   Tested by: kkm
........


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      7800a1c3
  51. Feb 09, 2011
  53. Feb 08, 2011
  55. Feb 04, 2011
    • Richard Mudgett's avatar
      Add ISDN display ie text handling options to chan_dahdi.conf. · a8aeb04a
      Richard Mudgett authored 
      The display ie handling can be controlled independently in the send and
receive directions with the following options:

* Block display text data.

* Use display text in SETUP/CONNECT messages for name.

* Use display text for COLP name updates (FACILITY/NOTIFY as appropriate).

* Pass arbitrary display text during a call.  Sent in INFORMATION
messages.  Received from any message that the display text was not used as
a name.

If the display options are not set then the options default to legacy
behavior.

The arbitrary display text is exchanged between bridged channels using the
AST_FRAME_TEXT frame type.

To send display text from the dialplan use the SendText() application when
the arbitrary display text option is enabled.

JIRA SWP-2688
JIRA ABE-2693


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      a8aeb04a
    • Jason Parker's avatar
      Merged revisions 306356 via svnmerge from · 0beeb00e
      Jason Parker authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r306356 | qwell | 2011-02-04 13:24:29 -0600 (Fri, 04 Feb 2011) | 16 lines
  
  Merged revisions 306346 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r306346 | qwell | 2011-02-04 13:21:43 -0600 (Fri, 04 Feb 2011) | 9 lines
    
    Don't fallthrough to 'unknown' in the 'ringing' case.
    
    This could cause improper exits from the queue.
    
    (closes issue #18499)
    Reported by: zaltar
    Patches: 
          app_queue.patch uploaded by zaltar (license 1148)
  ........
................


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