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

[jrose]

> On 2011-06-17 10:54:06, Paul Belanger wrote:
> > /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/ast1/sip.conf, line 4
> > <https://reviewboard.asterisk.org/r/1276/diff/1/?file=17032#file17032line4>
> >
> >     Bind to loopback: 127.0.0.1

I assume this goes for tlsbindaddr as well?  This and all the other similar problems have been fixed now.


> On 2011-06-17 10:54:06, Paul Belanger wrote:
> > /asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/ast1/manager.conf, line 1
> > <https://reviewboard.asterisk.org/r/1276/diff/1/?file=17031#file17031line1>
> >
> >     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.

Got it.


> On 2011-06-17 10:54:06, Paul Belanger wrote:
> > /asterisk/trunk/tests/channels/SIP/sip_tls_call/run-test, line 20
> > <https://reviewboard.asterisk.org/r/1276/diff/1/?file=17053#file17053line20>
> >
> >     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):

Switching it over will take some time, but I'll work on it.


> On 2011-06-17 10:54:06, Paul Belanger wrote:
> > /asterisk/trunk/tests/channels/SIP/sip_tls_call/run-test, lines 40-57
> > <https://reviewboard.asterisk.org/r/1276/diff/1/?file=17053#file17053line40>
> >
> >     Not sure this is the best approach.  Perhaps we can use #exec to build some of the dynamic information.

One thing this has going for it is that it's really easy, but if you don't like it, I'll put in some effort at learning what this #exec you speak of is and how I can work with it.


- jrose


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On 2011-06-17 10:41:49, jrose wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1276/
> -----------------------------------------------------------
> 
> (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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