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20 authors
  1. 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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      52dbebad
  3. Jan 13, 2011
  5. Jan 04, 2011
  7. Dec 31, 2010
  9. Nov 24, 2010
  11. Nov 02, 2010
  13. Oct 11, 2010
  15. Sep 23, 2010
  17. 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/
........



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      2f3dee23
  19. Sep 15, 2010
  21. Sep 10, 2010
  23. Sep 03, 2010
  25. Aug 13, 2010
  27. 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/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      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/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      bbb32fe3
  29. Aug 03, 2010
  31. Jul 29, 2010
  33. Jul 27, 2010
  35. Jul 26, 2010
  37. Jul 23, 2010
  39. Jul 20, 2010
  41. Jul 16, 2010
  43. Jul 13, 2010
  45. Jul 10, 2010
  47. Jul 09, 2010
  49. Jul 08, 2010
    • Mark Michelson's avatar
      Add IPv6 to Asterisk. · cd4ebd33
      Mark Michelson authored 
      This adds a generic API for accommodating IPv6 and IPv4 addresses
within Asterisk. While many files have been updated to make use of the
API, chan_sip and the RTP code are the files which actually support
IPv6 addresses at the time of this commit. The way has been paved for
easier upgrading for other files in the near future, though.

Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne
for their hard work on this.

(closes issue #17565)
Reported by: russell
Patches: 
      asteriskv6-test-report.pdf uploaded by russell (license 2)

Review: https://reviewboard.asterisk.org/r/743



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      cd4ebd33
  51. Jul 07, 2010
  53. Jun 22, 2010
  55. Jun 21, 2010
  57. Jun 17, 2010
  59. Jun 16, 2010