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20 authors
  1. Apr 05, 2011
  3. Apr 04, 2011
    • Richard Mudgett's avatar
      Remove the channel parameter from sig_pri_handle_subcmds(). · 121b90a4
      Richard Mudgett authored 
      It was only used in a debug message and may not be correct anyway.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312716 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      121b90a4
    • Richard Mudgett's avatar
      Merged revisions 312575 via svnmerge from · e1ceb52b
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r312575 | rmudgett | 2011-04-04 11:10:50 -0500 (Mon, 04 Apr 2011) | 52 lines
  
  Merged revisions 312574 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r312574 | rmudgett | 2011-04-04 11:00:02 -0500 (Mon, 04 Apr 2011) | 45 lines
    
    Merged revisions 312573 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r312573 | rmudgett | 2011-04-04 10:49:30 -0500 (Mon, 04 Apr 2011) | 38 lines
      
      Issues with ISDN calls changing B channels during call negotiations.
      
      The handling of the PROCEEDING message was not using the correct call
      structure if the B channel was changed.  (The same for PROGRESS.) The call
      was also not hungup if the new B channel is not provisioned or is busy.
      
      * Made all call connection messages (SETUP_ACKNOWLEDGE, PROCEEDING,
      PROGRESS, ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are
      using the correct structure and B channel.  If there is any problem with
      the operations then the call is now hungup with an appropriate cause code.
      
      * Made miscellaneous messages (INFORMATION, FACILITY, NOTIFY) find the
      correct structure by looking for the call and not using the channel ID.
      NOTIFY is an exception with versions of libpri before v1.4.11 because a
      call pointer is not available for Asterisk to use.
      
      * Made all hangup messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find
      the correct structure by looking for the call and not using the channel
      ID.
      
      (closes issue #18313)
      Reported by: destiny6628
      Tested by: rmudgett
      JIRA SWP-2620
      
      (closes issue #18231)
      Reported by: destiny6628
      Tested by: rmudgett
      JIRA SWP-2924
      
      (closes issue #18488)
      Reported by: jpokorny
      JIRA SWP-2929
      
      JIRA AST-437 (The issues fixed here are most likely causing this JIRA issue.)
      JIRA DAHDI-406
      JIRA LIBPRI-33 (Stuck resetting flag likely fixed)
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      e1ceb52b
  5. Apr 01, 2011
  7. Mar 31, 2011
    • Richard Mudgett's avatar
      Merged revisions 312022 via svnmerge from · ee44bf72
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r312022 | rmudgett | 2011-03-31 15:11:40 -0500 (Thu, 31 Mar 2011) | 14 lines
  
  chan_misdn segfaults when DEBUG_THREADS is enabled.
  
  The segfault happens because jb->mutexjb is uninitialized from the
  ast_malloc().  The internals of ast_mutex_init() were assuming a nonzero
  value meant mutex tracking initialization had already happened.  Recent
  changes to mutex tracking code to reduce excessive memory consumption
  exposed this uninitialized value.
  
  Converted misdn_jb_init() to use ast_calloc() instead of ast_malloc().
  Also eliminated redundant zero initialization code in the routine.
  
  (closes issue #18975)
  Reported by: irroot
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      ee44bf72
  9. Mar 30, 2011
  11. Mar 23, 2011
  13. Mar 18, 2011
  15. Mar 10, 2011
  17. Mar 08, 2011
  19. Mar 07, 2011
  21. Mar 05, 2011
  23. Mar 04, 2011
    • Russell Bryant's avatar
      Fix a buglet that prevented chan_nbs from loading (and subsequently stopped Asterisk). · c1ba13c1
      Russell Bryant authored 
      In passing, convert the return codes to be the proper AST_MODULE_LOAD_* constants.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309491 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      c1ba13c1
    • Richard Mudgett's avatar
      Merged revisions 309445 via svnmerge from · 928ec2b9
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r309445 | rmudgett | 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) | 46 lines
  
  Get real channel of a DAHDI call.
  
  Starting with Asterisk v1.8, the DAHDI channel name format was changed for
  ISDN calls to: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
  
  There were several reasons that the channel name had to change.
  
  1) Call completion requires a device state for ISDN phones.  The generic
  device state uses the channel name.
  
  2) Calls do not necessarily have B channels.  Calls placed on hold by an
  ISDN phone do not have B channels.
  
  3) The B channel a call initially requests may not be the B channel the
  call ultimately uses.  Changes to the internal implementation of the
  Asterisk master channel list caused deadlock problems for chan_dahdi if it
  needed to change the channel name.  Chan_dahdi no longer changes the
  channel name.
  
  4) DTMF attended transfers now work with ISDN phones because the channel
  name is "dialable" like the chan_sip channel names.
  
  For various reasons, some people need to know which B channel a DAHDI call
  is using.
  
  * Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and
  CHANNEL(dahdi_type) so the dialplan can determine the B channel currently
  in use by the channel.  Use CHANNEL(no_media_path) to determine if the
  channel even has a B channel.
  
  * Added AMI event DAHDIChannel to associate a DAHDI channel with an
  Asterisk channel so AMI applications can passively determine the B channel
  currently in use.  Calls with "no-media" as the DAHDIChannel do not have
  an associated B channel.  No-media calls are either on hold or
  call-waiting.
  
  (closes issue #17683)
  Reported by: mrwho
  Tested by: rmudgett
  
  (closes issue #18603)
  Reported by: arjankroon
  Patches:
        issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664)
  Tested by: stever28, rmudgett
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      928ec2b9
  25. Mar 02, 2011
  27. Mar 01, 2011
  29. Feb 25, 2011
  31. Feb 24, 2011
  33. Feb 23, 2011
  35. 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
  37. Feb 17, 2011
  39. Feb 15, 2011
    • Richard Mudgett's avatar
      Add CLI "pri show channels" command. · 4a486002
      Richard Mudgett authored 
      List the current mapping of DAHDI B channels to Asterisk channel names and
which calls are on hold or call-waiting.  Calls on hold or call-waiting
are not associated with any B channel.

JIRA LIBPRI-27
JIRA SWP-2547


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      4a486002
    • David Vossel's avatar
      Fixes compile error in chan_phone for big endian · 64ed1ba3
      David Vossel authored 
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307927 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      64ed1ba3
    • 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
  41. Feb 10, 2011
  43. Feb 08, 2011
  45. Feb 07, 2011
  47. Feb 05, 2011
  49. 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