[test-results] [Bamboo] Asterisk Testing > Asterisk Trunk > #817 has FAILED (1 tests failed). Change made by 7 authors.

[Bamboo]

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Asterisk Testing > Asterisk Trunk > #817 failed.
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Code has been updated by Mark Michelson, oej, rmudgett, dlee, seanbright, file, elguero.
1/2 jobs failed, with 1 failing test.

http://bamboo.asterisk.org/browse/TESTING-ASTERISKTRUNK-817/


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Failing Jobs
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  - Asterisk CentOS 6 64-Bit (CentOS 6): 1 of 304 tests failed.



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Code Changes
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file (376984):

>Tweak extension used for incoming calls received on Motif.
>
>Based on feedback from numerous individuals this patch tweaks incoming calls
>to first look for an extension with the name of the endpoint. If no such extension
>exists the call will silently fall back to the "s" extension as it previously
>did.
>........
>
>Merged revisions 376983 from http://svn.asterisk.org/svn/asterisk/branches/11
>

rmudgett (376871):

>chan_local: Fix local_pvt ref leak in local_devicestate().
>
>Regression introduced by ASTERISK-20390 fix.
>........
>
>Merged revisions 376868 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 376869 from http://svn.asterisk.org/svn/asterisk/branches/10
>........
>
>Merged revisions 376870 from http://svn.asterisk.org/svn/asterisk/branches/11
>

Mark Michelson (376918):

>Fix potential crashes during SIP attended transfers.
>
>The principal behind this patch is simple. During a transfer,
>we manipulate channels that are owned by a separate thread than
>the one we currently are running in, so it makes sense that we
>need to grab a reference to the channels so that they cannot
>disappear out from under us.
>
>In the wild, crashes were sometimes seen when the transferring
>party would hang up the call before the transfer target answered
>the call. The most common place to see the crash occur was when
>attempting to send a connected line update to the transferer
>channel.
>
>(closes issue ASTERISK-20226)
>Reported by Jared Smith
>Patches:
>	ASTERISK-20226.patch uploaded by Mark Michelson (License #5049)
>Tested by: Jared Smith
>........
>
>Merged revisions 376901 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 376916 from http://svn.asterisk.org/svn/asterisk/branches/10
>........
>
>Merged revisions 376917 from http://svn.asterisk.org/svn/asterisk/branches/11
>



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Tests
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New Test Failures (1)
   - AsteriskTestSuite: S/channels/ s i p/sip custom presence/resubscribe
Fixed Tests (2)
   - AsteriskTestSuite: S/apps/voicemail/authenticate nominal
   - AsteriskTestSuite: S/channels/ s i p/ s d p offer answer

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