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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/4006/">https://reviewboard.asterisk.org/r/4006/</a>
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<p style="margin-top: 0;">On September 19th, 2014, 2:34 p.m. UTC, <b>Mark Michelson</b> wrote:</p>
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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;">There are a couple of problems with this test:
1) It's quite a bit more complicated than it needs to be. What's actually being tested here is that Asterisk does not send a 503 in addition to a 486 on an INVITE retransmission. This only requires a UA to send and retransmit the INVITE and Asterisk. This should be doable entirely within a SIPp scenario and does not need voxcallcontrol, a Perl AGI load balancer, or grepping any logs.
2) The included sip.conf file is about 95% comments. This makes reviewing the config file much more difficult than it needs to be.
I think this test can be accomplished using the SIPpTestCase and defining the test details entirely within test-config.yaml, with no run-test file necessary. If you're unfamiliar with how this is done, there are several examples of this in the testsuite. A simple example can be found at tests/channels/SIP/directrtpsetup/test-config.yaml. In that test, there is a test-modules section that tells the testsuite to use sipp.SIPpTestCase as the main test object for the test (it also has an unnecessary add-test-to-search-path option set. You can ignore that). The corresponding test-object-config provides details about the SIPp scenarios to run. In that test, there are two scenarios run, but I suspect that for your test, you would only need a single scenario to run. If you're curious about what options are available for configuring the SIPpTestCase, you can look in sample-yaml/sipptestcase-config.yaml.sample for some more details. Your test can pass or fail based on whether the SIPp scenario succeeds or fails.</pre>
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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;">The problem is that I have to ignore INVITES for awhile by using a pause block. The 503 happens during that ignoring time after loosing a day and a half I couldn't find a better way then to pause then grep the log to see what happened. If you can make a version that can work entirely in sipp let me know :-)
agiload balancer and voxcallcontrol is there by mistake, (thought I had removed all references to them already) I'll check again and remove the references and post the new patch</pre>
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<p>- Torrey</p>
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<p>On September 19th, 2014, 8:09 a.m. UTC, Torrey Searle wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By Torrey Searle.</div>
<p style="color: grey;"><i>Updated Sept. 19, 2014, 8:09 a.m.</i></p>
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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-24335">ASTERISK-24335</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 is a test for the test suite to reproduce the issue described in ASTERISK-24335</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;">test passes when 4003 patch applied, fails when patch not applied</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">(5608)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/invite_retransmit/test-config.yaml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/invite_retransmit/sipp/A_PARTY.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/invite_retransmit/run-test <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/invite_retransmit/configs/ast1/sip.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/invite_retransmit/configs/ast1/extensions.conf <span style="color: grey">(PRE-CREATION)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/4006/diff/" style="margin-left: 3em;">View Diff</a></p>
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