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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3116/">https://reviewboard.asterisk.org/r/3116/</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;">I don't have anything to say about the ast_setstate, but Matt called it, so I guess it's good.</pre>
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<p>- wdoekes</p>
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<p>On January 10th, 2014, 4:49 p.m. UTC, Scott Griepentrog wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By Scott Griepentrog.</div>
<p style="color: grey;"><i>Updated Jan. 10, 2014, 4:49 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-23010">ASTERISK-23010</a>
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<b style="color: #575012; font-size: 10pt;">Repository: </b>
Asterisk
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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;">Setting the channel state to down is not done anywhere in chan_sip except handle_invite_replaces() after masquerading, and can interfere with the subsequent hangup of the channel. Removing it appears to make sense, and causes no problems, although I was unable to observe a new BYE message in the packet trace of test sip_one_legged_transfer.
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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 SIP tests in testsuite before and after, no new failures.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/branches/1.8/channels/chan_sip.c <span style="color: grey">(405265)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3116/diff/" style="margin-left: 3em;">View Diff</a></p>
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