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20 authors
  1. Apr 22, 2011
  3. Apr 21, 2011
  5. Apr 20, 2011
  7. Apr 19, 2011
  9. 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
    • Richard Mudgett's avatar
      Merged revisions 314069 via svnmerge from · 0a5c2d83
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r314069 | rmudgett | 2011-04-18 11:10:10 -0500 (Mon, 18 Apr 2011) | 22 lines
  
  The AsyncAGI command loop is lax in the value it returns for the return status.
  
  * Return correct status: SUCCESS/FAILED/HANGUP.  Previously, abnormal
  exits from the command loop such as hangup would return SUCCESS.
  
  * The "asyncagi break" command now returns SUCCESS and is now the only way
  to break the command loop with that status.  Previously, it returned
  FAILED.
  
  * The AMI event AsyncAGI End is no longer sent if the AsyncAGI Start event
  is not sent.  Previously, this happened because of an error setting up the
  AGI pipes.
  
  * All executed AGI commands now get an AsyncAGI Exec result event.
  Previously, if the command returned failure (because of hangup), the
  command loop just exited with FAILURE and did not send the AsyncAGI Exec
  result event.
  
  * Makes sure that the channel frame queue is empty on hangup.
  
  Review: https://reviewboard.asterisk.org/r/1183/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      0a5c2d83
    • Richard Mudgett's avatar
      Merged revisions 314068 via svnmerge from · 7c4fc0f0
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r314068 | rmudgett | 2011-04-18 11:02:12 -0500 (Mon, 18 Apr 2011) | 7 lines
  
  Unclear code in app_dial.c.
  
  Make code formatting clear.
  
  (closes issue #19134)
  Reported by: oej
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      7c4fc0f0
    • David Vossel's avatar
      Merged revisions 314067 via svnmerge from · 642249c3
      David Vossel authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r314067 | dvossel | 2011-04-18 10:23:45 -0500 (Mon, 18 Apr 2011) | 22 lines
  
  Remove the need for deadlock avoidance in chan_sip do_monitor.
  
  Deadlock avoidance between the sip pvt and the pvt->owner is
  very difficult.  Now that channel's are ao2 objects, this complication
  is no longer necessary.  It turns out the pvt's msg queue only
  exists because of deadlock avoidance (when deadlock avoidance fails
  msgs were added to a queue to be processed later), so this goes away as well.
  
  The technique used in the new sip_lock_pvt_full() function should
  be used as a template for replacing all locations where deadlock
  avoidance occurs between a channel tech_pvt and the pvt's owner.
  My hope is that this will begin a reversal of the invalid channel
  driver locking architecture we have been using for so long. 
  
  This patch also resolves an issue where the pvt->owner gets
  unlocked during processing the msg queue.
  
  (closes issue #18690)
  Reported by: dvossel
  
  Review: https://reviewboard.asterisk.org/r/1182/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      642249c3
    • David Vossel's avatar
      Merged revisions 314017 via svnmerge from · 4b454910
      David Vossel authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r314017 | dvossel | 2011-04-18 08:41:06 -0500 (Mon, 18 Apr 2011) | 17 lines
  
  sip codec negotiation of dynamic rtp payloads error fix
  
  This patch fixes how chan_sip handles dynamic rtp payload types
  it does not understand.  At the moment if a dynamic payload's mime
  type does not match one we understand, the payload does not get
  removed from our payload table.  As a result of this, the payload
  is set to whatever dynamic codec we use internally for that payload
  number on outgoing INVITES.  This is incorrect.
  
  This patch fixes this by properly checking the rtpmap set function's
  return code to make sure it was found.  The function can return both
  -1 and -2 depending on the source of the mismatch.  We were just
  checking -1 explicitly.
  
  Review: https://reviewboard.asterisk.org/r/1169/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      4b454910
  11. Apr 17, 2011
  13. Apr 15, 2011
  15. Apr 14, 2011
    • Terry Wilson's avatar
      Sets video mark bit on format field correctly · e9ba0cba
      Terry Wilson authored 
      This fixes a regression in the media architecture change
where video frames did not have their video mark set
correctly.  dvossel wrote this. twilson kindly committed
this, mmichelson found the bug.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313822 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      e9ba0cba
    • Richard Mudgett's avatar
      Merged revisions 313780 via svnmerge from · 4f8d56a8
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r313780 | rmudgett | 2011-04-14 15:59:56 -0500 (Thu, 14 Apr 2011) | 20 lines
  
  Leftover debug messages unconditionally sent to the console.
  
  Executing Dial(DAHDI/1/18475551212,300,) with the echotraining config
  option enabled outputs the following debug messages unconditionally:
  
  Dialing T1847555121 on 1
  Dialing www2w on 1
  
  * Made debug messages in my_dial_digits() normal debug messages that do
  not get output unless enabled.
  
  * Reworded some debug messages in my_dial_digits() to be clearer.
  
  * Replace strncpy() with ast_copy_string() in my_dial_digits() which does
  the same job better.
  
  (closes issue #18847)
  Reported by: vmikhelson
  Tested by: rmudgett
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      4f8d56a8
    • 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
  17. Apr 13, 2011
  19. Apr 12, 2011
  21. Apr 11, 2011