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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/4103/">https://reviewboard.asterisk.org/r/4103/</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 Joshua Colp.</div>
<p style="color: grey;"><i>Updated Nov. 3, 2014, 8:45 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 427112</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;">Currently when musiconhold is started or stopped in PJSIP it is always locally generated using res_musiconhold. This change adds an option, moh_passthrough, that allows musiconhold requests to be passed through chan_pjsip. This is done by sending a re-INVITE with recvonly state on the streams when the channel is put on hold and sending a re-INVITE with sendrecv state on the streams when the channel is taken off hold. The end result of this being that an upstream entity (such as another PBX) can generate the musiconhold instead.</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;">Enabled option. Placed call to a remote server. Put call on hold and off hold. Confirmed re-INVITEs were sent with proper state.</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/res/res_pjsip_sdp_rtp.c <span style="color: grey">(426095)</span></li>
<li>/trunk/res/res_pjsip/pjsip_configuration.c <span style="color: grey">(426095)</span></li>
<li>/trunk/res/res_pjsip.c <span style="color: grey">(426095)</span></li>
<li>/trunk/include/asterisk/res_pjsip_session.h <span style="color: grey">(426095)</span></li>
<li>/trunk/include/asterisk/res_pjsip.h <span style="color: grey">(426095)</span></li>
<li>/trunk/contrib/ast-db-manage/config/versions/339e1dfa644d_add_moh_passthrough_option_to_pjsip.py <span style="color: grey">(PRE-CREATION)</span></li>
<li>/trunk/channels/pjsip/dialplan_functions.c <span style="color: grey">(426095)</span></li>
<li>/trunk/channels/chan_pjsip.c <span style="color: grey">(426095)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/4103/diff/" style="margin-left: 3em;">View Diff</a></p>
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