[asterisk-dev] [Code Review] 4055: testsuite: chan_sip: Test unscheduling reINVITE after call hangup.

[wdoekes]

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Review request for Asterisk Developers.


Bugs: ASTERISK-22791
    https://issues.asterisk.org/jira/browse/ASTERISK-22791


Repository: testsuite


Description
-------

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.


Diffs
-----

  /asterisk/trunk/tests/channels/SIP/tests.yaml 5684 
  /asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/test-config.yaml 
PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/bob.xml 
PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/alice.xml 
PRE-CREATION 
  
/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/sip.conf 
PRE-CREATION 
  
/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/extensions.conf
 PRE-CREATION 

Diff: https://reviewboard.asterisk.org/r/4055/diff/


Testing
-------

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>


Thanks,

wdoekes


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