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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1992/">https://reviewboard.asterisk.org/r/1992/</a>
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<div>Review request for Asterisk Developers.</div>
<div>By Mark Michelson.</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 scenario sets up three SIP peers.
One is a UAC (Alice), bound to port 5063. And two are UASes, bound to port 5061 (Carol) and 5065 (Bob). We also have set outboundproxy to be the localhost, port 5062 (Percy).
We run three SIPp scenarios. One corresponds to Carol, one corresponds to Percy, and one corresponds to Alice. Alice attempts to call Bob. Those of you who are observant will notice that we do not have a SIPp scenario running for Bob.
Here is the expected call flow
Alice * Percy Carol
---INVITE (Bob)-->
---INVITE (Bob)------->
<-401 (Contact: Carol)-
--------ACK----------->
---INVITE (Bob)------->
<-200 (Contact: Carol)-
-------------------------ACK-------------->
<-----200 OK------
------ACK-------->
------BYE-------->
<-----200 OK------
-------------------------BYE-------------->
<---------------------200 OK---------------
The idea is that the outboundproxy should be the destination for the initial INVITE, despite the fact that Bob is the target. Then, despite the fact that Carol is specified as the Contact in the 401 to Asterisk, the ensuing ACK and INVITE with authentication still should go to Percy. Once Percy sends a 200 OK with Carol's contact, however, then further SIP traffic should be routed to Carol.
I've linked in ASTERISK-20008 because without the patch attached to that issue, this test fails.</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-20008">ASTERISK-20008</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/sip_outbound_proxy/configs/ast1/extensions.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_outbound_proxy/configs/ast1/sip.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_outbound_proxy/run-test <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_outbound_proxy/sipp/uac.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_outbound_proxy/sipp/uas-ackonly.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_outbound_proxy/sipp/uas.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_outbound_proxy/test-config.yaml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/tests.yaml <span style="color: grey">(3263)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1992/diff/" style="margin-left: 3em;">View Diff</a></p>
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