[asterisk-dev] [Code Review] sip_tls_call test added to external test suite

Paul Belanger reviewboard at asterisk.org
Fri Jun 17 10:54:06 CDT 2011


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A few comments about your test.  I'll hold off on commenting on the test util you look at the TestCase class.


/asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/ast1/manager.conf
<https://reviewboard.asterisk.org/r/1276/#comment7500>

    This file is not needed, the default configs will include this information.
    
    You just need to create a manager.general.conf.inc file with:
    enabled = yes.
    
    Again, checkout manager/login test.



/asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/ast1/sip.conf
<https://reviewboard.asterisk.org/r/1276/#comment7501>

    Bind to loopback: 127.0.0.1



/asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/ast1/sip.conf
<https://reviewboard.asterisk.org/r/1276/#comment7502>

    same



/asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/ast2/manager.conf
<https://reviewboard.asterisk.org/r/1276/#comment7503>

    same comment as above



/asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/ast2/sip.conf
<https://reviewboard.asterisk.org/r/1276/#comment7504>

    same



/asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/ast2/sip.conf
<https://reviewboard.asterisk.org/r/1276/#comment7505>

    same



/asterisk/trunk/tests/channels/SIP/sip_tls_call/run-test
<https://reviewboard.asterisk.org/r/1276/#comment7498>

    Look into manager/login test about inheriting the TestCase class.  This will help remove alot of duplicate code in your test.
    
    EG:
    
    class SIPCallTest(TestCase):



/asterisk/trunk/tests/channels/SIP/sip_tls_call/run-test
<https://reviewboard.asterisk.org/r/1276/#comment7499>

    Not sure this is the best approach.  Perhaps we can use #exec to build some of the dynamic information.


- Paul


On 2011-06-17 10:41:49, jrose wrote:
> 
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> https://reviewboard.asterisk.org/r/1276/
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> 
> (Updated 2011-06-17 10:41:49)
> 
> 
> Review request for Asterisk Developers, Russell Bryant, David Vossel, and Paul Belanger.
> 
> 
> Summary
> -------
> 
> First, you can ignore the text files.  spacespacespace.txt was just used while I was working out shell command stuff and I accidentally added it to this diff and I've also removed test-output.txt.
> 
> I'm still not perfectly sure how this is going to work with the cert files.  I'd guess it'll be fine regardless of the machine, but I haven't tested this on another machine yet that didn't create these cert files.
> 
> This test uses the basic-call test in IAX2 as a base.
> 
> 
> Diffs
> -----
> 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/ast1/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/ast1/manager.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/ast1/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/ast1/spacespacespace.txt PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/ast2/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/ast2/manager.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/ast2/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/keys/ca.cfg PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/keys/ca.crt PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/keys/ca.key PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/keys/serverA.crt PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/keys/serverA.csr PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/keys/serverA.key PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/keys/serverA.pem PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/keys/serverB.crt PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/keys/serverB.csr PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/keys/serverB.key PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/keys/serverB.pem PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/keys/tmp.cfg PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/sip_1_a.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/sip_1_b.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/sip_2_a.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/sip_2_b.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/run-test PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/test-config.yaml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/test-output.txt PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/tests.yaml 1633 
> 
> Diff: https://reviewboard.asterisk.org/r/1276/diff
> 
> 
> Testing
> -------
> 
> How did I test the test?  Mostly by checking the sip debug logs to make sure the call was going through as a SIP/TLS call.  The usual flow of SIP messages was there and it resembled my regular SIP/TLS calls.
> 
> 
> Thanks,
> 
> jrose
> 
>

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