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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/2592/">https://reviewboard.asterisk.org/r/2592/</a>
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<a href="https://reviewboard.asterisk.org/r/2592/diff/3/?file=40409#file40409line75" style="color: black; font-weight: bold; text-decoration: underline;">/asterisk/trunk/tests/feature_call_pickup/run-test</a>
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(Diff revision 3)
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> self.ami[0].hangup(channel2).addCallback(self.hangup)</pre></td>
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<pre style="margin-left: 2em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">channel2 doesn't exist here. Now hanging up channel. It didn't cause the test to fail, but it did raise an exception.</pre>
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<p>- jrose</p>
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<p>On June 25th, 2013, 10:05 p.m. UTC, jrose wrote:</p>
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<div>Review request for Asterisk Developers, kmoore and Matt Jordan.</div>
<div>By jrose.</div>
<p style="color: grey;"><i>Updated June 25, 2013, 10:05 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-21544">ASTERISK-21544</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 test is based roughly on the directed pickup application test (but that test doesn't really work anymore). Basically it does the following:
1) A local channel is originated to an extension that will dial a SIP peer (faker). This SIP peer points to an unused address, so it won't answer.
2) Once the dial starts, Asterisk 2 dials Asterisk 1 via SIP to extension *8 (the pickup extension assigned in features.conf)
3) At this point the SIP channel on Asterisk 2 (sip_receive is its peername on Asterisk 1) should answer the call made by the local channel since faker is in a callgroup set for use by sip_receive. Both channels will then enter a simple bridge together.
Once https://reviewboard.asterisk.org/r/2582/ is committed I can also add the pickupsound being played as a condition for completing the test.</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;">Ran it a few times and made sure what was happening matched with log messages after the test was over. Made sure all pass conditions were actually met as well.</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/feature_call_pickup/configs/ast1/extensions.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/feature_call_pickup/configs/ast1/features.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/feature_call_pickup/configs/ast1/sip.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/feature_call_pickup/configs/ast2/extensions.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/feature_call_pickup/configs/ast2/sip.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/feature_call_pickup/run-test <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/feature_call_pickup/test-config.yaml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/tests.yaml <span style="color: grey">(3820)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/2592/diff/" style="margin-left: 3em;">View Diff</a></p>
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