-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/4046/
-----------------------------------------------------------

Review request for Asterisk Developers.


Bugs: ASTERISK-24195
    https://issues.asterisk.org/jira/browse/ASTERISK-24195


Repository: Asterisk


Description
-------

Adding a mixmonitor to a channel causes the bridge to change technologies from 
native to simple_bridge so the call can be recorded.  However, when the 
mixmonitor is stopped the bridge does not switch back to the native technology.

* Added unbridge requests to reevaluate the bridge when a channel audiohook is 
removed.

* Moved the unbridge request into ast_audiohook_attach() ensure that the bridge 
reevaluates whenever an audiohook is attached.  This simplified the mixmonitor 
and chan_spy start code as well.

* Added defensive code to stop_mixmonitor_full() in case additional arguments 
are ever added to the StopMixMonitor application.

* Made ast_framehook_detach() not do an unbridge request if the framehook does 
not exist.

* Made ast_framehook_list_fixup() do an unbridge request if there are any 
framehooks.  Also simplified the loop.


Diffs
-----

  /branches/12/main/framehook.c 424382 
  /branches/12/main/audiohook.c 424382 
  /branches/12/apps/app_mixmonitor.c 424382 
  /branches/12/apps/app_chanspy.c 424382 

Diff: https://reviewboard.asterisk.org/r/4046/diff/


Testing
-------

1 Made SIP A call SIP B such that directmedia was possible.
2 Added a MixMonitor recorder to SIP A using AMI MixMonitor action.
3 The bridge technology changed to simple_bridge
4 Stopped the MixMonitor recorder using AMI StopMixMonitor action.
5 The bridge technology now changes to native with the patch.


Thanks,

rmudgett

-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Reply via email to