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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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