[test-results] [Bamboo] Asterisk - Team Branches > Pimp My SIP > #157 has FAILED (2 tests failed). Change made by root.

[Bamboo]

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Asterisk - Team Branches > Pimp My SIP > #157 failed.
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Code has been updated by root.
1/2 jobs failed, with 2 failing tests.

http://bamboo.asterisk.org/browse/ASTTEAM-PIMPMYSIP-157/


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Failing Jobs
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  - Asterisk 1.8 CentOS 6 32-Bit (CentOS 6): 2 of 196 tests failed.



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Code Changes
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root (382673):

>Multiple revisions 382670-382671
>
>........
>  r382670 | mjordan | 2013-03-07 21:54:38 -0600 (Thu, 07 Mar 2013) | 21 lines
>  
>  Don't reset the RTP address on a glare re-INVITE
>  
>  Originally, way back in r201583, we added the alternate RTP address so
>  that the RTP engine would expect to receive audio from a new source
>  when a glare re-INVITE occurred. In r382589, we remove the alternate
>  RTP source, as the 'secret' probation mode allows for switching to a new
>  RTP source when a previous source stops sending RTP. At the time, it
>  seemed appropriate to set the RTP source based on the information in the
>  glared re-INVITE.
>  
>  Unfortunately, that doesn't work so well - in a glared re-INVITE that occurs
>  with no SDP - such as in a connected line update that glances - we'll set
>  the RTP source to an invalid address. In subsequent re-INVITE requests from
>  this Asterisk instance, we'll then send an invalid media address, which will
>  result in the remote side sending a 488. Whoops.
>  
>  There isn't any need to reset the RTP source - if we're using strictrtp, we'll
>  simply synchronize to a new source when we stop getting packets from the old
>  one. If we aren't using strictrtp, then again there shouldn't be a problem.
>  
>  Note that the Asterisk Test Suite's connectedline test caught this error.
>........
>  r382671 | mjordan | 2013-03-07 22:11:12 -0600 (Thu, 07 Mar 2013) | 4 lines
>  
>  Remove unused function
>  
>  After r382670, get_ip_and_port_from_sdp was no longer used.
>........
>
>Merged revisions 382670-382671 from file:///srv/subversion/repos/asterisk/trunk
>



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Tests
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New Test Failures (2)
   - AsteriskTestSuite: S/channels/gulp/incoming calls without auth
   - AsteriskTestSuite: S/channels/gulp/handle options request

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