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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1093/">https://reviewboard.asterisk.org/r/1093/</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;">If you used SIPp to test this, will you add this as a test case in the test suite?</pre>
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<p>- Russell</p>
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<p>On February 3rd, 2011, 1:27 p.m., Terry Wilson wrote:</p>
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<div>Review request for Asterisk Developers, Olle E Johansson and David Vossel.</div>
<div>By Terry Wilson.</div>
<p style="color: grey;"><i>Updated 2011-02-03 13:27:31</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;">Asterisk currently accepts a REFER with a Refer-To with an embedded Replaces header that matches the dialog of the REFER. This would be a situation like A calls B, A calls C, A transfers B to A, which is just silly. This patch makes the transfer fail instead of making Asterisk freak out and forget to hang other channels up.</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;">Used sipp to test that the transfer failed properly in the aforementioned case. Used polycom phones to ensure that normal transfers still worked.</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/1.4/channels/chan_sip.c <span style="color: grey">(306052)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1093/diff/" style="margin-left: 3em;">View Diff</a></p>
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