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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/2337/">https://reviewboard.asterisk.org/r/2337/</a>
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<div>Review request for Asterisk Developers.</div>
<div>By Kevin Harwell.</div>
<p style="color: grey;"><i>Updated March 4, 2013, 3:15 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;">Addressed comments about removing items from the Allow header.</pre>
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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;">Tests to make sure the music on hold event is not triggered if the "discard_remote_hold_retrieval" option is set to "yes". In the test multiple scenarios are ran where one SIP phone puts another SIP phone on hold by sending a re-INVITE with a modified SDP containing either a restricted audio direction, an IP address of 0.0.0.0, or a combination thereof. This is tested both for a local RTP bridge, and a non-bridged scenario.</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 the test and made sure all scenarios passed. Also set the discard_remote_hold_retrieval to "no" and made sure the test failed. </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/ABE-2899">ABE-2899</a>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> (updated)</h1>
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<li>/asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/configs/ast1/extensions.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/configs/ast1/sip.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/sipp/phone_A.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/sipp/phone_B.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/test-config.yaml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/tests.yaml <span style="color: grey">(3642)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/2337/diff/" style="margin-left: 3em;">View Diff</a></p>
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