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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3525/">https://reviewboard.asterisk.org/r/3525/</a>
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<p style="margin-top: 0;">On May 12th, 2014, 8:58 p.m. CDT, <b>Matt Jordan</b> wrote:</p>
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<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">I feel like we're missing a SIPp scenario here. We have the initial receiver of the call, and we have the scenario that sends the REFER request - but where is the second SIP channel from the initiator of the transfer, and where is the destination of the transfer?</pre>
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<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">The only two SIPp scenarios involved here are the ones involved in the REFER. Both scenarios initiate a call into the dialplan: one into Stasis() and one into Echo(). After those are established, the leg that calls Echo() relays the requisite call information to the leg that calls Stasis() and the REFER is sent from there. Neither of these legs receive an invite.</pre>
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<p>- opticron</p>
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<p>On May 9th, 2014, 2:13 p.m. CDT, opticron wrote:</p>
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<div>Review request for Asterisk Developers and Joshua Colp.</div>
<div>By opticron.</div>
<p style="color: grey;"><i>Updated May 9, 2014, 2:13 p.m.</i></p>
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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-23641">ASTERISK-23641</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 reworks a significant portion of the ARI attended transfer test to avoid dependence on pjsua since it has the tendency to cause sporadic (and sometimes consistent) test failures. The reworked test uses SIPp with 3PCC to manage the transfer scenario.
This change also gives WebSocketEventModule (and anything else using EventMatcher) the ability to spawn SIPp scenarios in response to websocket events.</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;">Ensured that the test was operating as expected.</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/rest_api/bridges/attended_transfer/test-config.yaml <span style="color: grey">(5020)</span></li>
<li>asterisk/trunk/tests/rest_api/bridges/attended_transfer/sipp/call_stasis.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>asterisk/trunk/tests/rest_api/bridges/attended_transfer/sipp/call_echo.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>asterisk/trunk/tests/rest_api/bridges/attended_transfer/attended_transfer.py <span style="color: grey">(5020)</span></li>
<li>asterisk/trunk/lib/python/asterisk/ari.py <span style="color: grey">(5020)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3525/diff/" style="margin-left: 3em;">View Diff</a></p>
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