[asterisk-dev] [Code Review] 3297: Testsuite: Add test for direct RTP reinvite failure

[Matt Jordan]

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asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/test-config.yaml
<https://reviewboard.asterisk.org/r/3297/#comment20704>

    This seems excessive. I'd just remove it.


- Matt Jordan


On March 4, 2014, 11:30 a.m., opticron wrote:
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> (Updated March 4, 2014, 11:30 a.m.)
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> 
> Review request for Asterisk Developers.
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> Bugs: ASTERISK-23310
>     https://issues.asterisk.org/jira/browse/ASTERISK-23310
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> Repository: testsuite
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> Description
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> This adds a test for the scenario where Asterisk attempts to initiate a remote RTP native bridge, but one side declines and hangs up. This could previously cause a crash in Asterisk 1.8 and 11.
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> Diffs
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>   asterisk/trunk/tests/channels/SIP/tests.yaml 4745 
>   asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/test-config.yaml PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/sipp/uas-reinvite.xml PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/sipp/uas-no-reinvite.xml PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/configs/ast1/sip.conf PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/configs/ast1/extensions.conf PRE-CREATION 
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> Diff: https://reviewboard.asterisk.org/r/3297/diff/
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> Testing
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> Verified that the indicators of the crash did not show up when running this test and after Asterisk was patched to fix the problem. Also verified that the indicators did show up when Asterisk was unpatched.
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> Thanks,
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> opticron
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