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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3900/">https://reviewboard.asterisk.org/r/3900/</a>
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<div>Review request for Asterisk Developers, Matt Jordan and rmudgett.</div>
<div>By Jonathan Rose.</div>
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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-24027">ASTERISK-24027</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;">An odd little issue that can interrupt features and even hangup calls entirely in an unwanted fashion.
Reproduction is fairly trivial, simply have a dynamic bridge feature that plays audio or executes an AGI or something that just generally takes a little while and is sensitive to hangups.
Call into an extension that puts that feature on the channel and call another device.
Execute the feature.
While the feature is running, start mixmonitor on the channel executing the feature and watch as the feature is prematurely terminated and the call itself is hung up entirely.
Having the flag to re-evaluate the status of the bridge be a hangup flag seems to have been a little off point and trying to have everything that pays attention to hangups specifically ignore it seems a little wacky, so instead I've pulled AST_SOFTHANGUP_UNBRIDGE out of the soft hangup flags and turned it into its own thing.</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;">Performed the above reproduction steps with the patch and the call no longer hangs up and the feature completes normally. Mixmonitor captures all the audio as well.
Made sure native RTP bridges would still be re-evaluated and become simple bridges when a hook such as mixmonitor is placed on one of the bridged channels.
Ran through chan_sip and chan_pjsip testsuite tests to make sure the patch didn't introduce any failures.</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/main/pbx.c <span style="color: grey">(420558)</span></li>
<li>/branches/12/main/framehook.c <span style="color: grey">(420558)</span></li>
<li>/branches/12/main/channel_internal_api.c <span style="color: grey">(420558)</span></li>
<li>/branches/12/main/channel.c <span style="color: grey">(420558)</span></li>
<li>/branches/12/main/bridge_channel.c <span style="color: grey">(420558)</span></li>
<li>/branches/12/main/bridge_after.c <span style="color: grey">(420558)</span></li>
<li>/branches/12/include/asterisk/channel.h <span style="color: grey">(420558)</span></li>
<li>/branches/12/apps/app_stack.c <span style="color: grey">(420558)</span></li>
<li>/branches/12/apps/app_mixmonitor.c <span style="color: grey">(420558)</span></li>
<li>/branches/12/apps/app_chanspy.c <span style="color: grey">(420558)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3900/diff/" style="margin-left: 3em;">View Diff</a></p>
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