[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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https://reviewboard.asterisk.org/r/1276/#review3748
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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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