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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1769/">https://reviewboard.asterisk.org/r/1769/</a>
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<div>Review request for Asterisk Developers and Mark Michelson.</div>
<div>By jrose.</div>
<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;">A while back, when 1.6.3 was a thing, A largish patch was introduced that enabled SIP diversion headers to be sent, presumably for call forwarding. This adds an option to keep diversion headers introduced with that patch from being added to SIP dialogs because of interoperability issues that were introduced with that support.</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;">I actually couldn't find a useful test scenario. Call forwarding is a pretty roughly defined feature that tends to be implemented in the dialplan, and I couldn't find a method for call forwarding that actually included the Diversion header. Still, the approach is really simple. I don't think it'll be a problem.</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-16862">ASTERISK-16862</a>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/trunk/channels/chan_sip.c <span style="color: grey">(355731)</span></li>
<li>/trunk/channels/sip/include/sip.h <span style="color: grey">(355731)</span></li>
<li>/trunk/configs/sip.conf.sample <span style="color: grey">(355731)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1769/diff/" style="margin-left: 3em;">View Diff</a></p>
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