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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1640/">https://reviewboard.asterisk.org/r/1640/</a>
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<p style="margin-top: 0;">On December 22nd, 2011, 8:50 a.m., <b>jrose</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;">Hey, this looks right to me, but could you describe the call scenario to me a little better? I want to know things like what types of channels you were using and the settings on them, especially for directmedia in the case of SIP. The patch you are referencing was made to deal with nasty directmedia issues involving multiple servers, and those haven't been completely resolved yet, so I'm interested in what that could have broken and what this possibly could fix.</pre>
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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 have attached my sip.conf and extension.conf to reproduce this to the linked issue. My call setup is very simple. two sip phones behind a sip proxy and behind nat. both with directmedia=no. I also tried it with two sip phones directly connected to asterisk to make sure its not my proxy.
i also had some problems with directmedia enabled across two asterisk servers but i just disabled directmedia and then the one way audio problems stoped.
Maybe you could take a look at the debug log i have attached to the issue. I can also try to enable directmedia but normally all my phones are behind nat so i dont think this will work well.
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<p>- schmidts</p>
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<p>On December 22nd, 2011, 3:58 a.m., schmidts wrote:</p>
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<div>Review request for Asterisk Developers and jrose.</div>
<div>By schmidts.</div>
<p style="color: grey;"><i>Updated Dec. 22, 2011, 3:58 a.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;">with adding the AST_CONTROL_UPDATE_RTP_PEER control frame music on hold stoped working when a call was put on hold. The problem was that the control frame was only handled when received in a remote_bridge but not in a local_bridge.
By adding the handling of the UPDATE_RTP_PEER frame also to local_bridge moh works again.</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;">tested several calls. Moh is working again when a call is put on hold.</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/ASTERISK-19095">ASTERISK-19095</a>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>team/schmidts/unleash-the-beast/main/rtp_engine.c <span style="color: grey">(348832)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1640/diff/" style="margin-left: 3em;">View Diff</a></p>
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