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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3917/">https://reviewboard.asterisk.org/r/3917/</a>
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<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Just retested this with the SVN Asterisk 12 - works well!</pre>
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<p>- Krandon Bruse</p>
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<p>On August 15th, 2014, 3:53 p.m. UTC, Jonathan Rose wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By Jonathan Rose.</div>
<p style="color: grey;"><i>Updated Aug. 15, 2014, 3:53 p.m.</i></p>
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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-24043">ASTERISK-24043</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;">Steps for reproduction:
* one extension in dial plan
[default]
exten => 1000,1,NoOp(Show me something running in verbosity 3 please)
* Start a websocket listening for a stasis application
I use the websocket echo test to listen for application 'hello'
* Use the command POST /channels to call an endpoint and put it into stasis
I used the args endpoint='SIP/1601', app='hello', channelID='test'
* Use the POST /channels/{channelID}/continue command to make the channel
continue into the PBX I used the args channelID='test', context='default',
extension='1000', priority=1
Expectations:
* The channel should leave stasis and start a new life in the PBX. Or to
simplify, I should see a StasisEnd event on the websocket and I should see
extension 1000 executing on the SIP/1601-xxxxxxxx channel since I run at
verbosity 3.
Actual results prior to the patch:
* StasisEnd is received, but the channel never enters the PBX and I don't
see my NoOp application being ran.
The patch solves this problem by calling ast_pbx_run_args on the channel
before it would leave the stasis_app_exec function when there isn't a PBX
running on the channel and when the channel isn't hung up.
Patch: stasis-continue.diff submitted by Krandon Bruze</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;">Ran through reproduction with and without patch to confirm that the issue existed and that the patch fixes the issue.</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/res/res_stasis.c <span style="color: grey">(421079)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3917/diff/" style="margin-left: 3em;">View Diff</a></p>
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