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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: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/3297/ > ----------------------------------------------------------- > > (Updated March 4, 2014, 11:30 a.m.) > > > Review request for Asterisk Developers. > > > Bugs: ASTERISK-23310 > https://issues.asterisk.org/jira/browse/ASTERISK-23310 > > > Repository: testsuite > > > Description > ------- > > 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 > ----- > > 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 > ------- > > 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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