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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/4411/">https://reviewboard.asterisk.org/r/4411/</a>
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<p>Ship it!</p>
<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 caveat here is that you may want to watch automated runs of the SIP info_dtmf test to be sure that on awful hardware the 5 second Wait() isn't too short for the test to complete. I suspect it will be okay though.
On a side note, I have another test to add to my list of tests that could be rewritten to not rely on timing, though :)</pre>
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<p>- Mark Michelson</p>
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<p>On February 9th, 2015, 5:50 p.m. UTC, Corey Farrell wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By Corey Farrell.</div>
<p style="color: grey;"><i>Updated Feb. 9, 2015, 5:50 p.m.</i></p>
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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;">* Add Hangup() to priority after Dial() where needed. This prevents auto-fallthrough from playing 10 seconds of BUSY or CONGESTION tone.
* Decrease Wait(10) to Wait(5) in tests/channels/SIP/info_dtmf.
* Maintain list of AGI connections where needed so they can all be agi.finish().
* Replace calls to reactor.stop() with self.stop_reactor(), remove test.start_asterisk()/test.stop_asterisk() from main().
* Delay self.stop_reactor() in tests/channels/SIP/sip_tls_call by 2 seconds. This gives the calls enough time to end and avoid shutdown timeout.</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 all effected tests against Asterisk 11 with REF_DEBUG. Prior to these fixes graceful shutdown of Asterisk timed out, causing reference leaks to be reported. These tests now shutdown gracefully and have no reference leaks.</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/funcs/func_srv/run-test <span style="color: grey">(6377)</span></li>
<li>/asterisk/trunk/tests/funcs/func_presencestate/run-test <span style="color: grey">(6377)</span></li>
<li>/asterisk/trunk/tests/fastagi/stream-file/run-test <span style="color: grey">(6377)</span></li>
<li>/asterisk/trunk/tests/fastagi/database/run-test <span style="color: grey">(6377)</span></li>
<li>/asterisk/trunk/tests/fastagi/control-stream-file/run-test <span style="color: grey">(6377)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_tls_call/run-test <span style="color: grey">(6377)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_srtp/run-test <span style="color: grey">(6377)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_cause/configs/ast1/extensions.conf <span style="color: grey">(6377)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/secure_bridge_media/run-test <span style="color: grey">(6377)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/noload_res_srtp/run-test <span style="color: grey">(6377)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/info_dtmf/configs/ast1/extensions.conf <span style="color: grey">(6377)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/hangupcause/configs/ast1/extensions.conf <span style="color: grey">(6377)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/generic_ccss/configs/ast1/extensions.conf <span style="color: grey">(6377)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/4411/diff/" style="margin-left: 3em;">View Diff</a></p>
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