[test-results] [Bamboo] Asterisk - Team Branches > Pimp My SIP > #257 has FAILED (1 tests failed, no failures were new). Change made by root.

[Bamboo]

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Asterisk - Team Branches > Pimp My SIP > #257 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-257/


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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 (385476):

>Fix One-Way Audio With auto_* NAT Settings When SIP Calls Initiated By PBX
>
>When we reload Asterisk or chan_sip, the flags force_rport and comedia that are
>turned on and off when using the auto_force_rport and auto_comedia nat settings
>go back to the default setting off.  These flags are turned on when needed or
>off when not needed at the time that a peer registers, re-registers or initiates
>a call.  This would apply even when only the default global setting
>"nat=auto_force_rport" is being used, which in this case would only affect the
>force_rport flag.
>
>Everything is good except for the following:  The nat setting is set to
>auto_force_rport and auto_comedia.  We reload Asterisk and the peer's
>registration has not expired.  We load in the settings for the peer which turns
>force_rport and comedia back to off.  Since the peer has not re-registered or
>placed a call yet, those flags remain off.  We then initiate a call to the peer
>from the PBX.  The force_rport and comedia flags stay off.  If NAT is involved,
>we end up with one-way audio since we never checked to see if the peer is behind
>NAT or not.
>
>This patch does the following:
>
>* Moves the checking of whether a peer is behind NAT into its own function
>
>* Create a function to set the peer's NAT flags if they are using the auto_* NAT
>  settings
>
>* Adds calls in sip_request_call() to these new functions in order to setup the
>  dialog according to the peer's settings
>
>(closes issue ASTERISK-21374)
>Reported by: Michael L. Young
>Tested by: Michael L. Young
>Patches:
>    asterisk-21374-auto-nat-outgoing-fix_v2.diff Michael L. Young (license 5026)
>
>Review: https://reviewboard.asterisk.org/r/2421/
>........
>
>Merged revisions 385473 from http://svn.asterisk.org/svn/asterisk/branches/11
>........
>
>Merged revisions 385474 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

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