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20 authors
  1. Jul 06, 2011
    • Terry Wilson's avatar
      Replace Berkeley DB with SQLite 3 · efd040cd
      Terry Wilson authored 
      There were some bugs in the very ancient version of Berkeley DB that Asterisk
used. Instead of spending the time tracking down the bugs in the Berkeley code
we move to the much better documented SQLite 3.

Conversion of the old astdb happens at runtime by running the included
astdb2sqlite3 utility. The ast_db API with SQLite 3 backend should behave
identically to the old Berkeley backend, but in the future we could offer a
much more robust interface.

We do not include the SQLite 3 library in the source tree, but instead rely
upon the distribution-provided libraries. SQLite is so ubiquitous that this
should not place undue burden on administrators.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      efd040cd
  3. Jul 05, 2011
  5. Jul 01, 2011
  7. Jun 30, 2011
  9. Jun 29, 2011
    • Gregory Nietsky's avatar
      · f99a06d0
      Gregory Nietsky authored 
      Commit "distrotech" app_queue changes to Trunk

 * Added general option negative_penalty_invalid default off. when set
   members are seen as invalid/logged out when there penalty is negative.  
   for realtime members when set remove from queue will set penalty to -1.  
 * Added queue option autopausedelay when autopause is enabled it will be
   delayed for this number of seconds since last successful call if there
   was no prior call the agent will be autopaused immediately.
 * Added member option ignorebusy this when set and ringinuse is not   
   will allow per member control of multiple calls as ringinuse does for
   the Queue.
  
 - Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members
 - QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty.

(closes issue ASTERISK-17421)
(closes issue ASTERISK-17391)
Reported by: irroot
Tested by: irroot, jrose
Review: https://reviewboard.asterisk.org/r/1119/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325483 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      f99a06d0
  11. Jun 15, 2011
  13. Jun 13, 2011
  15. Jun 01, 2011
    • Russell Bryant's avatar
      Support routing text messages outside of a call. · 3f4d0e87
      Russell Bryant authored 
      Asterisk now has protocol independent support for processing text messages
outside of a call.  Messages are routed through the Asterisk dialplan.
SIP MESSAGE and XMPP are currently supported.  There are options in sip.conf
and jabber.conf that enable these features.

There is a new application, MessageSend().  There are two new functions,
MESSAGE() and MESSAGE_DATA().  Documentation will be available on
the project wiki, wiki.asterisk.org.

Thanks to Terry Wilson for the assistance with development and to David Vossel
for helping with some additional testing.

Review: https://reviewboard.asterisk.org/r/1042/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      3f4d0e87
  17. May 27, 2011
  19. May 25, 2011
  21. May 23, 2011
    • Richard Mudgett's avatar
      Merged revisions 320650 via svnmerge from · 024e4bd0
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r320650 | rmudgett | 2011-05-23 12:53:44 -0500 (Mon, 23 May 2011) | 16 lines
  
  Add ConnectedLineNum/Name headers to output of AMI action Status.
  
  * Add ConnectedLineNum and ConnectedLineName headers to the output of the
  AMI action Status.  This makes it easier to find out who the channel is
  connected to without having to lookup BridgedChannel or when they are
  connected to an application (e.g.: VoiceMail) which has no bridged
  channel.
  
  * Bridged channels with no CallerID had "" instead of "<unknown>" output,
  that might be a bug as "<unknown>" was what older versions used.
  
  (closes issue #18158)
  Reported by: gareth
  Patches:
        svn-292308.diff uploaded by gareth (license 208)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      024e4bd0
  23. May 20, 2011
  25. May 16, 2011
  27. May 09, 2011
  29. May 06, 2011
  31. May 05, 2011
  33. May 04, 2011
  35. May 03, 2011
    • Tilghman Lesher's avatar
      If multiple [general] contexts occur from sip.conf (usually due to external includes), merge them. · ed56ae3e
      Tilghman Lesher authored 
      The original implementation of this did the merging of all contexts with the
same name in the realtime layer, but that implementation severely breaks
drivers which use the same context name (e.g. iax.conf, type={peer,user}).
Therefore, the implementation needs to do the merging for particular entries
only, based upon what contexts would allow that in the channel driver itself.
This implementation is for chan_sip only, but others could be added in the
future.

(closes issue #17957)
 Reported by: marcelloceschia
 Patches: 
       chan-sip_parsing-general_branch162.patch uploaded by marcelloceschia (license 1079)
 Tested by: tilghman


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316428 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      ed56ae3e
  37. Apr 21, 2011
  39. Apr 20, 2011
  41. Apr 13, 2011
  43. Apr 04, 2011
  45. Apr 01, 2011
  47. Mar 18, 2011
  49. Mar 11, 2011
  51. Mar 04, 2011
  53. 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
  55. Feb 09, 2011
  57. Feb 07, 2011
  59. 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