[test-results] [Bamboo] Asterisk Testing > Asterisk Trunk > #568 has FAILED (2 tests failed, 1 failures were new). Change made by Mark Michelson and elguero.

[Bamboo]

-----------------------------------------------------------------------
Asterisk Testing > Asterisk Trunk > #568 failed.
-----------------------------------------------------------------------
Code has been updated by Mark Michelson, elguero.
2/283 tests failed, 1 failure was new.

http://bamboo.asterisk.org/browse/TESTING-ASTERISKTRUNK-568/


--------------
Failing Jobs
--------------
  - Asterisk CentOS 6 64-Bit (CentOS 6): 2 of 283 tests failed.



--------------
Code Changes
--------------
elguero (370955):

>Fix Not Unreferencing A Spied Channel
>
>When a channel hangs up while being spied upon and the option to exit the
>ChanSpy application when the spied on channel hangs up is set,
>ast_autochan_destroy is not being called and therefore a reference to the spied
>upon channel is not removed.
>
>The symptom being reported was that when using func_group in the dialplan and
>calling "group show channels" at the cli, the spied upon channel was still
>being shown while "core show channels" showed that the channel was not up.
>
>This patch calls ast_autochan_destroy when a spied upon channel hangs up and
>the option to exit the ChanSpy application is set, removing the reference to
>the channel allowing the count for the group that the spied channel was part of
>to be decremented.
>
>(closes issue ASTERISK-17515)
>Reported by: Arkadiusz Malka
>Tested by: Alexandr Gordeev, Michael L. Young
>Patches: 
>    asterisk-17515-destroy-autochan.diff
>                                    uploaded by Michael L. Young (license 5026)
>........
>
>Merged revisions 370952 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 370954 from http://svn.asterisk.org/svn/asterisk/branches/10
>

Mark Michelson (370953):

>Move a SIP change up to the other SIP changes in the CHANGES file.
>
>

Mark Michelson (370951):

>Allow support for early media on AMI originates and call files.
>
>This is based on the work done by Olle Johansson on review board.
>
>The idea is that the channel specified in an AMI originate or call
>file is typically not connected to the outgoing extension until the
>channel has been answered. With this change, an EarlyMedia header can
>be specified for AMI originates and an early_media option can
>be specified in call files. With this option set, once early media is
>received on a channel, it will be connected with the outgoing extension.
>
>(closes issue ASTERISK-18644)
>Reported by Olle Johansson
>
>Review: https://reviewboard.asterisk.org/r/1472
>
>



--------------
Tests
--------------
New Test Failures (1)
   - AsteriskTestSuite: S/channels/ s i p/sip blind transfer/callee refer only
Existing Test Failures (1)
   - AsteriskTestSuite: S/channels/ s i p/generic ccss

--
This message is automatically generated by Atlassian Bamboo
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/test-results/attachments/20120809/7a367ea6/attachment-0001.htm>