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20 authors
  1. Apr 21, 2011
  3. Apr 18, 2011
    • Richard Mudgett's avatar
      Problems with ISDN MWI to phones. · 37274c73
      Richard Mudgett authored 
      The "controlling user number" is always the number of the voice mail box
which is identical with the subscriber number itself.  This number which
is listed in the ISDN phone MWI menu cannot be called back to contact the
voice mail box.  The controlling user number should be made configurable.

JIRA ABE-2738
JIRA SWP-2846


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314116 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      37274c73
  5. Apr 14, 2011
    • Richard Mudgett's avatar
      Add Device State Information CCSS for Generic Devices. · ae2926b5
      Richard Mudgett authored 
      Add Asterisk Device State information and callbacks to the Call Completion
Supplemental Services for generic agents.

There are currently not many devices that have native support for CCSS.
Even as the devices become available there may be other reasons why one
may choose to not take advantage of the native abilities and stick with
the generic implementation.  The generic implementation is quite capable
and could be greatly enhanced by adding device state capabilities.  A
phone could then subscribe to the device state with a BLF key in
conjunction with Asterisk hints.

The advantages of the device state information would allow a single button
to: request CCSS, cancel a CCSS request, and display the current state of
a CCSS request.

For example, you may have a single button that when not lit, there is no
active CCSS request.  When you press that button, the dialplan can query
the DEVICE_STATE() associated with that caller to determine whether they
should be calling CallCompletionRequest() or CallCompletionCancel().  If
there is currently a pending request, then the dialplan would cancel it.
This also has the advantage of showing the true state of a request, which
is an asynchronous call, even when CallCompletionRequest() thinks it was
successful.  The actual request could ultimately fail.  Once lit, further
feedback can be provided to the caller about the current state of their
request since it will be updated by the CCSS State Machine as appropriate.

The DEVICE_STATE mapping is configurable since the BLF being used on a
given phone type may vary.  The idea is to allow some level of
customization as to the phone's behavior.

As an example, you may want the BLF key to go solid once you have
requested a callback.  You may then want the LED to blink (typically
ringing) when either the callback is in process, which is a visual
indication that the incoming call is the desired callback.  You may want
it to blink when the callee is ready but you are busy, giving you a visual
indication that the target is available as you may want to get off the
line so that the callback can be successful.

Device state information is sent back via the ast_devstate_prov_add()
callback for any generic CCSS device as it traverses through the state
machine.  You simply provide a map between CC_STATE values and the
corresponding AST_DEVICE state values.

You could then generate hints against these states similar to what is
possible today with Custom Devstates or MeetMe states.  For example, you
may have an extension 3000 that is currently associated with device
SIP/3000.  You could then create a feature code for that extension that
may look something like:

exten => *823000,hint,ccss:sip/3000

You would then subscribe a BLF button to *823000 which would point to the
dialplan that handled CCSS requests/cancels using the available
DEVICE_STATE() information about ccss:sip/3000 to make the decision about
what to do.

(closes issue #18788)
Reported by: p_lindheimer
Patches:
      ccss.trunk.18788.patch uploaded by p lindheimer (license 558)
      Modified with final reviewboard comments.
Tested by: p_lindheimer, loloski

Review: https://reviewboard.asterisk.org/r/1105/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      ae2926b5
  7. Apr 13, 2011
  9. Apr 05, 2011
  11. Mar 31, 2011
  13. Mar 17, 2011
  15. Mar 07, 2011
  17. Mar 04, 2011
  19. Feb 24, 2011
  21. 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
  23. Feb 10, 2011
  25. Feb 08, 2011
  27. Feb 07, 2011
  29. 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
  31. Feb 03, 2011
  33. Feb 02, 2011
  35. Feb 01, 2011
  37. Jan 31, 2011
  39. Jan 27, 2011
  41. Jan 26, 2011
  43. Jan 20, 2011
  45. Jan 19, 2011
  47. Jan 17, 2011
  49. Jan 13, 2011
  51. Jan 11, 2011
  53. Jan 04, 2011
  55. Dec 31, 2010
  57. Dec 21, 2010
  59. Dec 17, 2010
  61. Dec 08, 2010
  63. Nov 22, 2010
  65. Nov 18, 2010
  67. Nov 17, 2010
  69. Nov 08, 2010