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      This is an automatically generated e-mail. To reply, visit:
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 <p>Ship it!</p>









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    <a href="https://reviewboard.asterisk.org/r/4055/diff/2/?file=67967#file67967line10" style="color: black; font-weight: bold; text-decoration: underline;">/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/test-config.yaml</a>
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     (Diff revision 2)

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    <td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "></pre></td>
    <th bgcolor="#b1ebb0" style="border-left: 1px solid #C0C0C0; border-right: 1px solid #C0C0C0;" align="right"><font size="2">10</font></th>
    <td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">    <span class="l-Scalar-Plain">minversion</span><span class="p-Indicator">:</span> <span class="s">'1.8.0.0'</span></pre></td>
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<pre style="margin-left: 2em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Nitpick: since this would fail in most versions of 1.8, the minversion should be the next scheduled release of 1.8 that would contain this fix, i.e., 1.8.32.0.

The 1.8 branch is always considered to be 'greater' than an explicit tag version.</pre>
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<p>- Matt Jordan</p>


<br />
<p>On October 9th, 2014, 4:16 p.m. CDT, wdoekes wrote:</p>








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<div>Review request for Asterisk Developers.</div>
<div>By wdoekes.</div>


<p style="color: grey;"><i>Updated Oct. 9, 2014, 4:16 p.m.</i></p>







<div style="margin-top: 1.5em;">
 <b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>


 <a href="https://issues.asterisk.org/jira/browse/ASTERISK-22791">ASTERISK-22791</a>


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<div style="margin-top: 1.5em;">
 <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;">ASTERISK-22791 details how asterisk resends a reINVITE even though the
call has already been hung up by a BYE.

This tests that problem.

Also note how the From/To are also reversed, since this is a reINVITE
*to* alice where alice is in the From.</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;">Before it is fixed:


<?xml version="1.0" encoding="utf-8"?>
<testsuite errors="0" failures="1" name="AsteriskTestSuite" tests="1" time="2.84">
  <testcase name="tests/channels/SIP/no_reinvite_after_491" time="2.84">
    <failure>Running ['./lib/python/asterisk/test_runner.py', 'tests/channels/SIP/no_reinvite_after_491'] ...
[Oct 08 17:54:30] WARNING[4582]: sipp:437 processEnded: Resolving remote host '127.0.0.1'... Done.

[Oct 08 17:54:30] WARNING[4582]: sipp:437 processEnded: 2014-10-08      17:54:30.202158 1412783670.202158: Aborting call on unexpected message for Call-Id '1-4636@127.0.0.1': while pausing (index 10), received 'INVITE sip:alice@127.0.0.1:5062 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK717504e3;rport
Max-Forwards: 70
From: alice &lt;sip:alice@127.0.0.1:5062&gt;;tag=4636SIPpTag001
To: bob &lt;sip:bob@127.0.0.1:5060&gt;;tag=as7d7023cd
Contact: &lt;sip:bob@127.0.0.1:5060&gt;
Call-ID: 1-4636@127.0.0.1
CSeq: 103 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r424181
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 296

v=0
o=root 30542954 30542956 IN IP4 127.0.0.1
s=Asterisk PBX SVN-branch-1.8-r424181
c=IN IP4 127.0.0.1
t=0 0
m=image 4725 udptl t38
c=IN IP4 127.0.0.1
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:389
a=T38FaxUdpEC:t38UDPRedundancy
'.

[Oct 08 17:54:30] WARNING[4582]: sipp:539 __scenario_callback: SIPp Scenario alice.xml Failed [1]
[Oct 08 17:54:30] WARNING[4582]: sipp:548 __evaluate_scenario_results: SIPp Scenario alice.xml Failed
[Oct 08 17:54:30] WARNING[4582]: sipp:402 kill: Killing SIPp Scenario bob.xml
</failure>
  </testcase>
</testsuite>


After a possible fix:


<?xml version="1.0" encoding="utf-8"?>
<testsuite errors="0" failures="0" name="AsteriskTestSuite" tests="1" time="9.95">
  <testcase name="tests/channels/SIP/no_reinvite_after_491" time="9.95"/>
</testsuite></pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
<ul style="margin-left: 3em; padding-left: 0;">

 <li>/asterisk/trunk/tests/channels/SIP/tests.yaml <span style="color: grey">(5684)</span></li>

 <li>/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/test-config.yaml <span style="color: grey">(PRE-CREATION)</span></li>

 <li>/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/bob.xml <span style="color: grey">(PRE-CREATION)</span></li>

 <li>/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/alice.xml <span style="color: grey">(PRE-CREATION)</span></li>

 <li>/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/sip.conf <span style="color: grey">(PRE-CREATION)</span></li>

 <li>/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/extensions.conf <span style="color: grey">(PRE-CREATION)</span></li>

</ul>

<p><a href="https://reviewboard.asterisk.org/r/4055/diff/" style="margin-left: 3em;">View Diff</a></p>







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