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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/2868/">https://reviewboard.asterisk.org/r/2868/</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, Joshua Colp and Matt Jordan.</div>
<div>By jrose.</div>
<p style="color: grey;"><i>Updated Sept. 20, 2013, 5:04 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 399553</pre>
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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-22487">ASTERISK-22487</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;">When forwarding a call (basically a blind transfer when ringing) that has been originated, Asterisk would crash because a the original channel would be hung up twice. Also the context of the forwarding channel wouldn't be respected. This patch brings things more in line with how forwarding is handled in basic calls.</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;">Originated a call to SIP/A which would send the call into a holding bridge
While SIP/A was ringing, it requests forwarding to SIP/B
SIP/B picks up.
Prior to changes:
SIP/B wouldn't ring
Asterisk would crash due to hanging up a channel without a technology (since the first hangup strips the technology)
After the change:
SIP/B rings, picking up the call causes SIP/B to enter the holding bridge as expected (once local channels optimize out anyway).
Asterisk no longer crashes.</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/dial.c <span style="color: grey">(399197)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/2868/diff/" style="margin-left: 3em;">View Diff</a></p>
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