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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1647/">https://reviewboard.asterisk.org/r/1647/</a>
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<div>Review request for Asterisk Developers, otherwiseguy, schmidts, and jrose.</div>
<div>By Matt Jordan.</div>
<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 series of tests to the testsuite that cover SIP hold. The tests use two phones (A / B), wherein Phone A calls Phone B, Phone B puts Phone A on hold, waits a period of time, removes the hold on Phone A, then sends a BYE. The test checks that MOH is started / stopped in each scenario by subscribing to the MusicOnHold AMI event.
The SIP hold tests include Phone B sending a re-INVITE containing in the SDP either a restricted audio direction, a receiving IP address of 0.0.0.0, or a combination thereof. The tests cover the two SIP endpoints interacting either directly or through a local RTP bridge.
Note that this test will fail in 1.8.8.0, and was used to test the regression identified in ASTERISK-19095. The test will pass in 1.8.7.2, 1.8.8.1, and the current 1.8 branch.</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-19095">ASTERISK-19095</a>
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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">(2951)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_hold/test-config.yaml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_media_restrict.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_restrict.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_hold/sipp/inject_bridge.csv <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_hold/sipp/inject_bypass.csv <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_media_restrict.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_hold/configs/ast1/sip.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_hold/run-test <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_hold/configs/ast1/extensions.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/lib/python/asterisk/sipp.py <span style="color: grey">(2951)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1647/diff/" style="margin-left: 3em;">View Diff</a></p>
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