[test-results] [Bamboo] Asterisk - Team Branches > Pimp My SIP > #248 has FAILED (1 tests failed, no failures were new). Change made by root.

[Bamboo]

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Asterisk - Team Branches > Pimp My SIP > #248 failed.
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Code has been updated by root.
1/2 jobs failed, with 1 failing test, no failures were new.

http://bamboo.asterisk.org/browse/ASTTEAM-PIMPMYSIP-248/


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Failing Jobs
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  - Asterisk 1.8 CentOS 6 32-Bit (CentOS 6): 1 of 271 tests failed.



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Code Changes
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root (385176):

>Fix crash in chan_sip when a core initiated op occurs at the same time as a BYE
>
>When a BYE request is processed in chan_sip, the current SIP dialog is detached
>from its associated Asterisk channel structure. The tech_pvt pointer in the
>channel object is set to NULL, and the dialog persists for an RFC mandated
>period of time to handle re-transmits.
>
>While this process occurs, the channel is locked (which is good).
>Unfortunately, operations that are initiated externally have no way of knowing
>that the channel they've just obtained (which is still valid) and that they are
>attempting to lock is about to have its tech_pvt pointer removed. By the time
>they obtain the channel lock and call the channel technology callback, the
>tech_pvt is NULL.
>
>This patch adds a few checks to some channel callbacks that make sure the
>tech_pvt isn't NULL before using it. Prime offenders were the DTMF digit
>callbacks, which would crash if AMI initiated a DTMF on the channel at the
>same time as a BYE was received from the UA. This patch also adds checks on
>sip_transfer (as AMI can also cause a callback into this function), as well
>as sip_indicate (as lots of things can queue an indication onto a channel).
>
>Review: https://reviewboard.asterisk.org/r/2434/
>
>(closes issue ASTERISK-20225)
>Reported by: Jeff Hoppe
>........
>
>Merged revisions 385170 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 385173 from http://svn.asterisk.org/svn/asterisk/branches/11
>........
>
>Merged revisions 385174 from file:///srv/subversion/repos/asterisk/trunk
>



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Tests
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Existing Test Failures (1)
   - AsteriskTestSuite: S/channels/gulp/incoming calls without auth
Fixed Tests (1)
   - AsteriskTestSuite: S/channels/gulp/basic calls/incoming/nominal/authed/userpass/ident by user

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