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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3535/">https://reviewboard.asterisk.org/r/3535/</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 and Joshua Colp.</div>
<div>By Matt Jordan.</div>
<p style="color: grey;"><i>Updated May 18, 2014, 3:29 p.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 414122</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;">This patch fixes the currently failing pjsip/transfers/blind_transfer/caller_direct_media test (with a few small tweaks to the test as well).
The test currently fails primarily for two reasons:
(1) When Bob and Charlie (the transfer target and the transfer destination) enter a bridge together, the framehook remains on the transfer target channel until both channels are in the bridge. As it consumes voice frames, the initial bridge type is a simple bridge. The framehook is removed when both channels are in the bridge; however, this does not currently cause the bridging framework to re-evaluate the bridge. This patch adds a AST_SOFTHANGUP_UNBRIDGE poke to the transfer target channel when a framehook is removed so the bridge can re-evaluate itself.
(2) When a channel leaves a native RTP bridge, it may be leaving due to being hung up. Sending a re-INVITE to a channel that is about to be hung up is not nice - in fact, there's a good chance we'll send the BYE request before the channel has had a chance to send back a 200 OK. To be somewhat nicer, this patch makes it so that we only send the re-INVITE if there's a chance the channel will survive the native bridging experience.</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;">Once some timing issues were removed from the test, it passes with this patch.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/branches/12/res/res_pjsip_session.c <span style="color: grey">(413948)</span></li>
<li>/branches/12/main/framehook.c <span style="color: grey">(413948)</span></li>
<li>/branches/12/main/channel.c <span style="color: grey">(413948)</span></li>
<li>/branches/12/main/bridge_channel.c <span style="color: grey">(413948)</span></li>
<li>/branches/12/include/asterisk/channel.h <span style="color: grey">(413948)</span></li>
<li>/branches/12/bridges/bridge_native_rtp.c <span style="color: grey">(413948)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3535/diff/" style="margin-left: 3em;">View Diff</a></p>
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