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      This is an automatically generated e-mail. To reply, visit:
      <a href="https://reviewboard.asterisk.org/r/3297/">https://reviewboard.asterisk.org/r/3297/</a>
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<blockquote style="margin-left: 1em; border-left: 2px solid #d0d0d0; padding-left: 10px;">
 <p style="margin-top: 0;">On March 4th, 2014, 4:45 p.m. CST, <b>Matt Jordan</b> wrote:</p>
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    <a href="https://reviewboard.asterisk.org/r/3297/diff/2/?file=55259#file55259line26" style="color: black; font-weight: bold; text-decoration: underline;">asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/test-config.yaml</a>
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     (Diff revision 2)

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    <td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "></pre></td>
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    <td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">    <span class="l-Scalar-Plain">reactor-timeout</span><span class="p-Indicator">:</span> <span class="l-Scalar-Plain">80</span></pre></td>
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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;">This seems excessive. I'd just remove it.</pre>
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<pre style="margin-left: 1em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">This is necessary to avoid a reactor timeout in the nominal case (the failure case will end at around 40 seconds) since the test must wait for a SIP dialog to be destroyed which is what actually triggered the error condition this is testing for.</pre>
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<p>- opticron</p>


<br />
<p>On March 4th, 2014, 11:30 a.m. CST, opticron wrote:</p>








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<div>Review request for Asterisk Developers.</div>
<div>By opticron.</div>


<p style="color: grey;"><i>Updated March 4, 2014, 11:30 a.m.</i></p>







<div style="margin-top: 1.5em;">
 <b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>


 <a href="https://issues.asterisk.org/jira/browse/ASTERISK-23310">ASTERISK-23310</a>


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 <b style="color: #575012; font-size: 10pt;">Repository: </b>
testsuite
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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 adds a test for the scenario where Asterisk attempts to initiate a remote RTP native bridge, but one side declines and hangs up. This could previously cause a crash in Asterisk 1.8 and 11.</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;">Verified that the indicators of the crash did not show up when running this test and after Asterisk was patched to fix the problem. Also verified that the indicators did show up when Asterisk was unpatched.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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 <li>asterisk/trunk/tests/channels/SIP/tests.yaml <span style="color: grey">(4745)</span></li>

 <li>asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/test-config.yaml <span style="color: grey">(PRE-CREATION)</span></li>

 <li>asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/sipp/uas-reinvite.xml <span style="color: grey">(PRE-CREATION)</span></li>

 <li>asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/sipp/uas-no-reinvite.xml <span style="color: grey">(PRE-CREATION)</span></li>

 <li>asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/configs/ast1/sip.conf <span style="color: grey">(PRE-CREATION)</span></li>

 <li>asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/configs/ast1/extensions.conf <span style="color: grey">(PRE-CREATION)</span></li>

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<p><a href="https://reviewboard.asterisk.org/r/3297/diff/" style="margin-left: 3em;">View Diff</a></p>







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