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

[Matt Jordan]

-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/4055/#review13493
-----------------------------------------------------------

Ship it!



/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/test-config.yaml
<https://reviewboard.asterisk.org/r/4055/#comment24012>

    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.


- Matt Jordan


On Oct. 9, 2014, 4:16 p.m., wdoekes wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4055/
> -----------------------------------------------------------
> 
> (Updated Oct. 9, 2014, 4:16 p.m.)
> 
> 
> 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 at 127.0.0.1': while pausing (index 10), received 'INVITE sip:alice at 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 <sip:alice at 127.0.0.1:5062>;tag=4636SIPpTag001
> To: bob <sip:bob at 127.0.0.1:5060>;tag=as7d7023cd
> Contact: <sip:bob at 127.0.0.1:5060>
> Call-ID: 1-4636 at 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
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20141011/b19238c4/attachment-0001.html>