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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3505/">https://reviewboard.asterisk.org/r/3505/</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.</div>
<div>By Jonathan Rose.</div>
<p style="color: grey;"><i>Updated May 9, 2014, 11:10 a.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 413551</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-23381">ASTERISK-23381</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;">This is a cleanup of the patches posted by Robert Moss for ASTERISK-23381 - A stalled review of the patch exists here: https://reviewboard.asterisk.org/r/3331/diff/1/#index_header
This version is made to apply to Asterisk 11 since the bug was first reported there.</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;">1. SIP/A calls a spy application which spies on SIP/B with the DTMF option (4 for spy mode, 5 for whisper mode, 6 for barge)
2. SIP/B calls SIP/C
3. Check to see if audio is flowing from A to B and from A to C... should start as neither. (A always hears both of the others though)
* press DTMF 5
- now audio should flow from A to B, but from not A to C
* press DTMF 6
- now audio should flow from A to B and A to C
4. Hang up SIP/B
5. Have SIP/B call SIP/C again
6. The spy should reconnect automatically and be in barge mode
- audio should flow from A to B and A to C
* press DTMF 5
- audio should flow from A to B, but not from A to C
* press DTMF 4
- audio should not flow from A to either phone
Prior to patch: Audio from A to C could not be heard in the first check
After the patch: Audio from A to C can be heard in the first check
All other checks work as expected in both situations.</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/11/apps/app_chanspy.c <span style="color: grey">(413071)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3505/diff/" style="margin-left: 3em;">View Diff</a></p>
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