[test-results] [Bamboo] Asterisk Testing > Asterisk Trunk > #1274 has FAILED (3 tests failed, 1 failures were new). Change made by elguero.

Bamboo bamboo at asterisk.org
Mon May 13 22:24:14 CDT 2013


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Asterisk Testing > Asterisk Trunk > #1274 failed.
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Code has been updated by elguero.
1/2 jobs failed, with 3 failing tests, 1 failure was new.

http://bamboo.asterisk.org/browse/TESTING-ASTERISKTRUNK-1274/


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Failing Jobs
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  - Asterisk CentOS 6 32-Bit (CentOS 6): 3 of 249 tests failed.



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Code Changes
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elguero (388602):

>Fix Crash Caused By One-way Audio With auto_* NAT Settings Fix
>
>The prior code committed, r385473, failed to take into consideration that not
>all outgoing calls will be to a peer.  My fault.
>
>This patch does the following:
>
>* Check if there is a related peer involved.  If there is, check and set NAT 
>  settings according to the peer's settings.
>
>* Fix a problem with realtime peers.  If the global setting has auto_force_rport
>  set and we issued a "sip reload" while a peer is still registered, the peer's
>  flags for NAT are reset to off.  When this happens, we were always setting the
>  contact address of the peer to that of the full contact info that we had.
>
>(closes issue ASTERISK-21374)
>Reported by: jmls
>Tested by: Michael L. Young
>Patches:
>   asterisk-21374-fix-crash-and-rt-peers.diff by Michael L. Young (license 5026)
>
>Review: https://reviewboard.asterisk.org/r/2524/
>........
>
>Merged revisions 388601 from http://svn.asterisk.org/svn/asterisk/branches/11
>



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Tests
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New Test Failures (1)
   - AsteriskTestSuite: S/apps/confbridge/confbridge marked
Existing Test Failures (2)
   - AsteriskTestSuite: S/channels/ s i p/sip tls call
   - AsteriskTestSuite: S/bridge/transfer failure

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