<html>
<body>
<div style="font-family: Verdana, Arial, Helvetica, Sans-Serif;">
<table bgcolor="#f9f3c9" width="100%" cellpadding="8" style="border: 1px #c9c399 solid;">
<tr>
<td>
This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3431/">https://reviewboard.asterisk.org/r/3431/</a>
</td>
</tr>
</table>
<br />
<table bgcolor="#fefadf" width="100%" cellspacing="0" cellpadding="8" style="background-image: url('https://reviewboard.asterisk.org/static/rb/images/review_request_box_top_bg.ab6f3b1072c9.png'); background-position: left top; background-repeat: repeat-x; border: 1px black solid;">
<tr>
<td>
<div>Review request for Asterisk Developers.</div>
<div>By rmudgett.</div>
<div style="margin-top: 1.5em;">
<b style="color: #575012; font-size: 10pt;">Repository: </b>
Asterisk
</div>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
<table width="100%" bgcolor="#ffffff" cellspacing="0" cellpadding="10" style="border: 1px solid #b8b5a0">
<tr>
<td>
<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;">The failing assertion ensures that the final snapshot gets generated so CDR records can get finalized. The only place where a channel staging snapshot flag could be left set is in the handle_request_bye(). The function could return before clearing the flag because the channel could dissappear while the function had to have the channel unlocked.
* Fixed handle_request_bye() channel snapshot staging coverage area to not have a return in the middle of it and be unable to clear the staging flag.
* Pushed the channel snapshot staging coverage area into ast_rtp_instance_set_stats_vars() to ensure that the staging is not interrutped.
* Made callers of ast_rtp_instance_set_stats_vars() not call it with channels or channel driver private locks held to eliminate the deadlock potential. The callers must hold references to the passed in channel and rtp objects.
* Eliminated sip_hangup() trying to get the bridge peer. It is futile at this point because the channel could never be in a bridge.
* Moved sip_pvt unref in ast_hangup() and handle_request_do() to the end of the function. The unref needs to happen after the last use of the pointer.</pre>
</td>
</tr>
</table>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
<table width="100%" bgcolor="#ffffff" cellspacing="0" cellpadding="10" style="border: 1px solid #b8b5a0">
<tr>
<td>
<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;">I was unsuccessful in reproducing the testsuite channel staging assertion failure.
However, SIP calls can still setup and teardown with the patch installed.</pre>
</td>
</tr>
</table>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
<ul style="margin-left: 3em; padding-left: 0;">
<li>/branches/12/main/rtp_engine.c <span style="color: grey">(412047)</span></li>
<li>/branches/12/channels/chan_sip.c <span style="color: grey">(412047)</span></li>
</ul>
<p><a href="https://reviewboard.asterisk.org/r/3431/diff/" style="margin-left: 3em;">View Diff</a></p>
</td>
</tr>
</table>
</div>
</body>
</html>