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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1128/">https://reviewboard.asterisk.org/r/1128/</a>
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<div>Review request for Asterisk Developers.</div>
<div>By Alec Davis.</div>
<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;">Symptom:
When an external call originating from IAX2 a SEGFAULT (or from DAHDI a BAD MAGIC NUMBER) happens as the transfer between 2 internal SIP extensions completes.
Initially before the transfer is complete and the before masquerade happens the glue0 and glue1 pointers are pointing to update_peer (which is a reference to sip_set_rtp_peer)
Then after transfer completes and masquerading happens, glue0 and glue1 don't change, but the channel is now an IAX channel.
calling glue0->update_peer(c0, NULL, NULL, NULL, 0, 0)) is now fatal.
Solution:
If the channel condition is one of the following after breaking out of the loop, don't try to update_peer
1). ZOMBIE
2). cx->tech_pvt != pvtx
3). gluex != ast_rtp_instance_get_glue(c0->tech->type))
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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;">Multiple transfers for now nearly 2 weeks, on a production 1.8.2.3 box handling ~ 800 calls a day.
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<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/view.php?id=18781">18781</a>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>trunk/main/rtp_engine.c <span style="color: grey">(309081)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1128/diff/" style="margin-left: 3em;">View Diff</a></p>
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