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  1. Apr 21, 2011
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  17. 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
  19. Feb 09, 2011
  21. Feb 07, 2011
  23. 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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      a8aeb04a
  25. Jan 21, 2011
    • Tilghman Lesher's avatar
      Add DB_KEYS. · 52dbebad
      Tilghman Lesher authored 
      Discussion on #asterisk on 2011-01-19:
(02:07:03 PM) boch: i wonder how to cycle all entries in a tree
(02:07:11 PM) leifmadsen: use While()
(02:07:17 PM) leifmadsen: you need to know the tree structure already though
(02:07:36 PM) boch: what you mean?
(02:09:02 PM) leifmadsen: you need to know the structure prior to looping, because you can't just return the structure from the dialplan
(02:09:43 PM) leifmadsen: the only way I can think of doing that is via something like writing the output of:  asterisk -rx "database show" to a file, then looping through that to know the structure of the database and check everything
(02:09:59 PM) leifmadsen: but at that point you're better off just using either a relational database or an external script
(02:10:13 PM) boch: for example i need to know all entries in the tree
(02:10:15 PM) boch: got it
(02:10:20 PM) leifmadsen: exactly
(02:10:22 PM) leifmadsen: that's the problem
(02:10:22 PM) boch: thank you
(02:13:09 PM) mateu: yeah, i'm surprised there isn't something from the dialplan like 'database show family' so one can get all keys in a family to loop over.
(02:15:35 PM) leifmadsen: database shows everything
(02:16:22 PM) mateu: i mean something from the dial plan that mimics 'database show <family>'
(02:16:41 PM) leifmadsen: guess no one has found that important enough to program :)
(02:16:52 PM) leifmadsen: at that point you should probably just use a relational database...
(02:17:10 PM) mateu: i dunno
(02:17:16 PM) mateu: seems pretty basic to me.
(02:17:16 PM) leifmadsen: me either
(02:17:19 PM) leifmadsen: sure does
(02:17:24 PM) leifmadsen: no one has programmed it though
(02:17:28 PM) ***leifmadsen shrugs
(02:17:43 PM) mateu: ok, well at least we know how it currently stands.  thanks leifmadsen
(02:28:52 PM) Corydon76-home: leifmadsen: something like HASHKEYS() ?
(02:30:11 PM) leifmadsen: Corydon76-home: ummm, I was thinking more like DUNDI_QUERY() and DUNDI_RESULT()
(02:30:31 PM) leifmadsen: although HASHKEYS() might work
(02:30:58 PM) leifmadsen: actually ya, looking at it, similar to HASHKEYS()
(02:31:01 PM) leifmadsen: DBKEYS() I guess?
(02:31:45 PM) Corydon76-home: So with no argument, retrieves families, with an argument, retrieves keys of that family?
(02:34:02 PM) leifmadsen: ya
(02:34:16 PM) leifmadsen: how would you iterate through layers of them?
(02:34:30 PM) leifmadsen: i.e. family/key/key/key ?
(02:34:43 PM) Corydon76-home: Essentially, yes


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      52dbebad
  27. Jan 13, 2011
  29. Jan 04, 2011
  31. Dec 31, 2010
  33. Nov 24, 2010
  35. Nov 02, 2010
  37. Oct 11, 2010
  39. Sep 23, 2010
  41. Sep 20, 2010
    • David Vossel's avatar
      Merged revisions 287647 via svnmerge from · 2f3dee23
      David Vossel authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r287647 | dvossel | 2010-09-20 17:09:16 -0500 (Mon, 20 Sep 2010) | 21 lines
  
  Addition of the FrameHook API (AKA AwesomeHooks)
  
  So far all our tools for viewing and manipulating media streams
  within Asterisk have been entirely focused on audio.  That made
  sense then, but is not scalable now.  The FrameHook API lets us
  tap into and manipulate _ANY_ type of media or signaling passed
  on a channel present today or in the future.  This tool is a step
  in the direction of expanding Asterisk's boundaries and will help
  generate some rather interesting applications in the future.
  
  In addition to the FrameHook API, a simple dialplan function
  exercising the api has been included as well.  This function
  is called FRAME_TRACE().  FRAME_TRACE() allows for the internal
  ast_frames read and written to a channel to be output.  Filters
  can be placed on this function to debug only certain types of frames.
  This function could be thought of as an internal way of doing
  ast_frame packet captures.
  
  Review: https://reviewboard.asterisk.org/r/925/
........



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      2f3dee23
  43. Sep 15, 2010
  45. Sep 10, 2010
  47. Sep 03, 2010
  49. Aug 13, 2010
  51. Aug 12, 2010
    • Russell Bryant's avatar
      Merged revisions 282066 via svnmerge from · 57535c59
      Russell Bryant authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r282066 | russell | 2010-08-12 15:41:17 -0500 (Thu, 12 Aug 2010) | 4 lines
  
  Add a "core reload" CLI command.
  
  Review: https://reviewboard.asterisk.org/r/859/
........


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      57535c59
    • David Vossel's avatar
      Merged revisions 282047 via svnmerge from · bbb32fe3
      David Vossel authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r282047 | dvossel | 2010-08-12 15:15:41 -0500 (Thu, 12 Aug 2010) | 35 lines
  
  improved translation paths for wideband codecs
  
  The problem I'm addressing is that Asterisk's current
  method of building the least cost translation paths
  between codecs does not take into account sample rate.
  For instance, it was possible for siren14 (a 32khz codec),
  to contain the a translation path to siren7 (a 16khz
  audio codec) that goes through slin at 8khz.  In this
  case Asterisk takes a 32khz codec, down samples it to
  8khz and then up samples it to 16khz which is terrible
  regardless if it is computationally less expensive.  This
  patch now builds translation paths that give priority to
  maintaining the best possible sample rate before taking
  into consideration computational cost.  This patch also
  adds cli commands to expose what translation paths are
  actually being used.
  
  Changes:
  1. Translation paths will never contain a step that changes
  the sample rate unless absolutely necessary.
  2. When choosing the best codec to make two channels compatible.
  Shared codecs with the highest sample rate are given priority.
  3. A new cli command to show all translation paths available
  for a specific codec 'core show translation paths [codec name]'
  has been added.
  4. 'core show translation' which displays the translation
  matrix now includes the new higher bit audio codecs in the table.
  5. 'core show channel [channel name]'  now displays the
  translation paths if translation is used.
  
  (closes issue #16841)
  Reported by: dvossel
  
  Review: https://reviewboard.asterisk.org/r/842/
........


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      bbb32fe3
  53. Aug 03, 2010
  55. Jul 29, 2010
  57. Jul 27, 2010
  59. Jul 26, 2010
  61. Jul 23, 2010
  63. Jul 20, 2010
  65. Jul 16, 2010