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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1407/">https://reviewboard.asterisk.org/r/1407/</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;">Looks like its is in order</pre>
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<p>- irroot</p>
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<p>On September 5th, 2011, 5:15 a.m., Alec Davis wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By Alec Davis.</div>
<p style="color: grey;"><i>Updated Sept. 5, 2011, 5:15 a.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;">If a call arrives before asterisk is fully booted generally it will segfault.</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;">restarted asterisk, and before all modules have finished loading made a call into it.
Warning message appears, and call is dropped.
No orphaned channels.
== Using SIP RTP CoS mark 5
== Registered translator 'slin 96000khz -> 32000khz' from format slin96 to slin32, table cost, 850000, computational cost 999999
[2011-09-05 21:57:11.617417] WARNING[30782]: pbx.c:5363 ast_pbx_start: PBX requires Asterisk to be fully booted
[2011-09-05 21:57:11.617973] WARNING[30782]: chan_sip.c:22917 handle_request_invite: Failed to start PBX :(
== Registered translator 'slin 96000khz -> 44100khz' from format slin96 to slin44, table cost, 850000, computational cost 999999
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>trunk/main/pbx.c <span style="color: grey">(333893)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1407/diff/" style="margin-left: 3em;">View Diff</a></p>
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