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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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<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 might be confused, but you have removed a lot of headers that won't come back if you disable this functionality. That's not good. Or have I missed something?</pre>
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<a href="https://reviewboard.asterisk.org/r/1159/diff/2/?file=16051#file16051line12911" style="color: black; font-weight: bold; text-decoration: underline;">/trunk/channels/chan_sip.c</a>
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(Diff revision 2)
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static int transmit_refer(struct sip_pvt *p, const char *dest)</pre></td>
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<td bgcolor="#ffc5ce" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> <span class="n">add_supported_header</span><span class="p">(</span><span class="n">p</span><span class="p">,</span> <span class="o">&</span><span class="n">req</span><span class="p">);</span></pre></td>
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<pre style="margin-left: 2em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">WHy delete all this code? Are you saying that the user has to add these headers in the dial plan now?</pre>
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<p>- Olle E</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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