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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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<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>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Changes</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 now works with 11. Also, it's a lot more specific about which channels it expects than it used to be.
One quirk of Asterisk 11 is that Pickup events and Bridge events don't happen in an obviously defined order while in 12 it seems fairly certain that the Pickup will occur before the Bridge. Because of this, the test now no longer cares which order the events appear in (which includes Asterisk 12's bridge_enter events); merely that they are all received as expected.</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/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> (updated)</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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