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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/2728/">https://reviewboard.asterisk.org/r/2728/</a>
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<p style="margin-top: 0;">On August 20th, 2013, 11:43 a.m. UTC, <b>wdoekes</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/2728/diff/2/?file=44230#file44230line7401" 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 sip_hangup(struct ast_channel *ast)</pre></td>
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<th bgcolor="#f0f0f0" style="border-right: 1px solid #C0C0C0;" align="right"><font size="2">7380</font></th>
<td bgcolor="#ffffff" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span>return;</pre></td>
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<td bgcolor="#ffffff" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> return;</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;">This accidentally turned into 4 spaces?</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;">Editor :-(
Fixed.</pre>
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<p>- Matt</p>
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<p>On August 19th, 2013, 11:36 p.m. UTC, Matt Jordan wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By Matt Jordan.</div>
<p style="color: grey;"><i>Updated Aug. 19, 2013, 11:36 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-21976">ASTERISK-21976</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;">{quote}
For video calls, we would like to set the codecs in the dialplan using
SIP_CODEC. However, if SIP_CODEC is set, all codecs except the ONE set are disallowed and thus either audio or video is available.
Attached is a patch for 11.4 that allows SIP_CODEC to contain a list of codecs , e.g. "gsm,h264".
{quote}
As an aside, chan_pjsip has an analogous dialplan function "PJSIP_MEDIA_OFFER". While this doesn't allow for setting multiple codecs, it does handle multiple media types, as you can specify both video or audio for the codec you want to apply - hence I didn't port this patch/feature over to chan_pjsip. If we think it needs it, it would be reasonably easy to do so.</pre>
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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">(396943)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/2728/diff/" style="margin-left: 3em;">View Diff</a></p>
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