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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3297/">https://reviewboard.asterisk.org/r/3297/</a>
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<div>Review request for Asterisk Developers.</div>
<div>By opticron.</div>
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<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23310">ASTERISK-23310</a>
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<b style="color: #575012; font-size: 10pt;">Repository: </b>
testsuite
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">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.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">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.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>asterisk/trunk/tests/channels/SIP/tests.yaml <span style="color: grey">(4745)</span></li>
<li>asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/test-config.yaml <span style="color: grey">(PRE-CREATION)</span></li>
<li>asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/sipp/uas-reinvite.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/sipp/uas-no-reinvite.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/configs/ast1/sip.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/configs/ast1/extensions.conf <span style="color: grey">(PRE-CREATION)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3297/diff/" style="margin-left: 3em;">View Diff</a></p>
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