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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/2147/">https://reviewboard.asterisk.org/r/2147/</a>
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<p>Ship it!</p>
<p>- Mark</p>
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<p>On October 10th, 2012, 2:07 p.m., opticron wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By opticron.</div>
<p style="color: grey;"><i>Updated Oct. 10, 2012, 2:07 p.m.</i></p>
<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 is nearly a complete rewrite of the IPv6 SIP Attended Transfer Test using SIPpTest with AMIEventModule in the configuration-driven test framework and SIPp's 3PCC extended mode for higher level call control and message passing among SIPp instances.
The test had previously been set to be skipped because it was failing randomly, presumably because of race conditions in the test itself.
This also resulted in a few enhancements to the testsuite to allow for the use of IPv6 targets (already committed).</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;">The test runs (seemingly) reliably on my box.</pre>
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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/SWP-4661">SWP-4661</a>
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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/sip_attended_transfer_v6/configs/ast1/sip.conf <span style="color: grey">(3490)</span></li>
<li>asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/hangup_call.py <span style="color: grey">(PRE-CREATION)</span></li>
<li>asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/run-test <span style="color: grey">(3490)</span></li>
<li>asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/slave_cfg.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/uac-call.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/uac-calls-and-refer.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/uas-no-hangup.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/test-config.yaml <span style="color: grey">(3490)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/2147/diff/" style="margin-left: 3em;">View Diff</a></p>
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