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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1397/">https://reviewboard.asterisk.org/r/1397/</a>
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<div>Review request for Asterisk Developers.</div>
<div>By irroot.</div>
<p style="color: grey;"><i>Updated Aug. 29, 2011, 8:30 a.m.</i></p>
<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;">A sip phone tries to pickup a call but for some reason looses the pvt this causes all sorts of mischief.
Im proposing that sip_hangup return -1 when there is no private it has failed what it should be doing.
if the hangup fails ast_do_masquerade will kill the channel and return not leaving the channel orphaned.
[Aug 26 10:50:21] NOTICE[29961] app_directed_pickup.c: pickup SIP/2201-0000167c attempt by SIP/2225-0000168c
[Aug 26 10:50:22] WARNING[29961] chan_sip.c: No SIP tech_pvt! Fixup of SIP/2225-0000168c failed.
[Aug 26 10:50:22] WARNING[29961] channel.c: Fixup failed on channel SIP/2225-0000168c<MASQ>, strange things may happen.
[Aug 26 10:51:23] NOTICE[21093] chan_sip.c: Disconnecting call 'SIP/2225-0000168c' for lack of RTP activity in 61 seconds
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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-18273">ASTERISK-18273</a>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/team/irroot/distrotech-customers-1.8/channels/chan_sip.c <span style="color: grey">(333499)</span></li>
<li>/team/irroot/distrotech-customers-1.8/main/channel.c <span style="color: grey">(333499)</span></li>
<li>/team/irroot/distrotech-customers-1.8/main/features.c <span style="color: grey">(333499)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1397/diff/" style="margin-left: 3em;">View Diff</a></p>
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