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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/2466/">https://reviewboard.asterisk.org/r/2466/</a>
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<p style="margin-top: 0;">On April 25th, 2013, 3:41 p.m. UTC, <b>Joshua Colp</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/2466/diff/2/?file=36390#file36390line331" style="color: black; font-weight: bold; text-decoration: underline;">/team/include/asterisk/res_sip.h</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; ">struct ast_sip_endpoint {</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="cm">/*! Do we add Diversion headers to applicable outgoing requests/responses? */</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="kt">unsigned</span> <span class="kt">int</span> <span class="n">send_diversion</span><span class="p">;</span></pre></td>
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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;">chan_sip has an option called "promiscredir" which follows the explicit SIP URI and does not just go to the user portion within the dialplan. Is this something we want to do as well here? (albeit maybe not the same way)</pre>
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<p>On April 25th, 2013, 5:11 p.m. UTC, <b>Kevin Harwell</b> wrote:</p>
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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 can add this option...and a call to ast_channel_call_forward_build right? Or is that what you meant by not the same way?</pre>
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<p>On April 25th, 2013, 5:20 p.m. UTC, <b>Joshua Colp</b> wrote:</p>
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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;">Yeah. The conundrum for supporting it is how to expose it to the dialplan. Right now in chan_sip if you have the option enabled it sends a call to SIP/<the URI, minus the sip: at the front> - This won't really work since it has no endpoint and using the dialed endpoint seems wrong. How about exposing the SIP URI via a dialplan variable and sending the call to an explicit extension?
ie: If enabled all diverted calls go to the "diverted" extension and the user can then decide what to do with the SIP URI they have been diverted to.
Thoughts?</pre>
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<p>On April 25th, 2013, 5:49 p.m. UTC, <b>Kevin Harwell</b> wrote:</p>
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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;">If I understand correctly do we even need a "promiscredir" option? Of do we just want to have the dialplan variable always available with the SIP URI for every redirect? Or would that option disallow redirection all together?</pre>
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<p>On April 25th, 2013, 5:51 p.m. UTC, <b>Joshua Colp</b> wrote:</p>
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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;">If you don't have the option then we'd go to the user portion of the URI in all cases, which means the user would then have to do a match all and direct it elsewhere using the URI. It also means you couldn't easily support a case with that and without that. I think having an option is best.</pre>
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<pre style="margin-left: 1em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">So I started to implement this and there does not seem to be a way to access the set dialplan variable on the new outgoing channel. Is there something I am missing? Is it possible to do this at redirection time?</pre>
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<p>- Kevin</p>
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<p>On April 24th, 2013, 8:09 p.m. UTC, Kevin Harwell wrote:</p>
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<div>Review request for Asterisk Developers, Joshua Colp and Mark Michelson.</div>
<div>By Kevin Harwell.</div>
<p style="color: grey;"><i>Updated April 24, 2013, 8:09 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-21426">ASTERISK-21426</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;">Adds call forwarding support (Josh's patch) to the new SIP work being done in Asterisk. This also includes the ability to add a diversion header to an outgoing response/request when appropriate. The diversion header feature can be turned off by setting the send_diversion=false (defaults to true) on an endpoint within the configuration file.</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;">Wrote a few testsuite tests to make sure calls are being forwarded and the diversion header is added/propagated appropriately.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/team/group/pimp_my_sip/channels/chan_gulp.c <span style="color: grey">(386451)</span></li>
<li>/team/group/pimp_my_sip/include/asterisk/res_sip.h <span style="color: grey">(386451)</span></li>
<li>/team/group/pimp_my_sip/res/res_sip.c <span style="color: grey">(386451)</span></li>
<li>/team/group/pimp_my_sip/res/res_sip/sip_configuration.c <span style="color: grey">(386451)</span></li>
<li>/team/group/pimp_my_sip/res/res_sip_diversion.c <span style="color: grey">(PRE-CREATION)</span></li>
<li>/team/group/pimp_my_sip/res/res_sip_session.c <span style="color: grey">(386451)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/2466/diff/" style="margin-left: 3em;">View Diff</a></p>
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