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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1159/">https://reviewboard.asterisk.org/r/1159/</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've tried this patch and it is working for me both on trunk and backported to 1.8 (where I have tested it more heavily since my production systems use it).</pre>
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<p>- Marquis</p>
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<p>On April 3rd, 2011, 9:36 p.m., kkm wrote:</p>
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<div>Review request for Asterisk Developers, Russell Bryant and Olle E Johansson.</div>
<div>By kkm.</div>
<p style="color: grey;"><i>Updated April 3, 2011, 9:36 p.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;">There is currently no way to augment a REFER message from Transfer with extra headers. The attached patch implements the feature.
The feature is enabled by default, with a new sip.conf setting to disable it. The rationale for it to be on by default is that it is in fact very controllable from the dialplan: It takes one application call to SIPRemoveHeader with no argument to remove all previously accumulated additional SIP headers. Since Transfer normally terminates the channel, there is no need in practice to keep any SIP headers beyond it in the channel, so that removing these does not impose any dialplan programming complexity.
A hunk near chan_sip.c line 11697 also fixes an issue with Refer-To header gaining an extra set of <> around the address only when retransmitted due to an authentication request.</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;">Confirmed working per spec with sip set debug on packet dump.
Deployed and used on a production server under 1.8.3 and working for 2 weeks already.</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/19059">19059</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/CHANGES <span style="color: grey">(312554)</span></li>
<li>/trunk/channels/chan_sip.c <span style="color: grey">(312554)</span></li>
<li>/trunk/channels/sip/include/sip.h <span style="color: grey">(312554)</span></li>
<li>/trunk/configs/sip.conf.sample <span style="color: grey">(312554)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1159/diff/" style="margin-left: 3em;">View Diff</a></p>
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