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

[jrose]

> On 2011-06-21 15:14:35, Russell Bryant wrote:
> > /asterisk/trunk/tests/tests.yaml, lines 3-27
> > <https://reviewboard.asterisk.org/r/1276/diff/3/?file=17107#file17107line3>
> >
> >     You might want to revert the changes that disable all other tests.  :-)
> >     
> >     You can run a single test directly by using the -t option.
> >     
> >     ./runtests.py -t tests/channels/SIP/mytest

I was planning on it.  Didn't realize I could run an individual test otherwise, so that's helpful information going into the next one.

The next big step is converting it over to use AMI to , but getting the whole thing syncing up properly is driving me nuts at the moment.  Once I get that done, doing the registration test should be pretty trivial, but getting there is proving complicated.


- jrose


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On 2011-06-21 12:33:58, jrose wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1276/
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> 
> (Updated 2011-06-21 12:33:58)
> 
> 
> 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.general.conf.inc 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/ast2/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/ast2/manager.general.conf.inc 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/helper1 PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/helper2 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/run-test PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/test-config.yaml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/tests.yaml 1657 
>   /asterisk/trunk/tests/tests.yaml 1657 
> 
> 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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