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

[opticron]

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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3297/
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(Updated March 4, 2014, 11:30 a.m.)


Review request for Asterisk Developers.


Changes
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Fix the test's minversion and remove extraneous code from a SIPp scenario.


Bugs: ASTERISK-23310
    https://issues.asterisk.org/jira/browse/ASTERISK-23310


Repository: testsuite


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.


Diffs (updated)
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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 

Diff: https://reviewboard.asterisk.org/r/3297/diff/


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.


Thanks,

opticron

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