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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3742/">https://reviewboard.asterisk.org/r/3742/</a>
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<h1 style="margin-right: 0.2em; padding: 0; font-size: 10pt;">This change has been marked as submitted.</h1>
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<div>Review request for Asterisk Developers.</div>
<div>By opticron.</div>
<p style="color: grey;"><i>Updated July 11, 2014, 7:47 a.m.</i></p>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Changes</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;">Committed in revision 418365</pre>
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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;">During the media rework, setting of nativeformats for chan_pjsip channels was removed for inbound channels. This prevents chan_pjsip channels from dialing other channels due to having (nothing) as their native format capability. It's hard to transcode from (something) to (nothing). This tweak gives chan_pjsip channels back their native formats on creation.</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;">Made sure the incoming call could dial other channels and pass audio.</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/media_formats-reviewed-trunk/channels/chan_pjsip.c <span style="color: grey">(418253)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3742/diff/" style="margin-left: 3em;">View Diff</a></p>
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