[asterisk-dev] [Code Review] sip_tls_call test added to external test suite
Paul Belanger
reviewboard at asterisk.org
Fri Jun 17 14:39:04 CDT 2011
-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1276/#review3753
-----------------------------------------------------------
/asterisk/trunk/tests/channels/SIP/sip_tls_call/run-test
<https://reviewboard.asterisk.org/r/1276/#comment7512>
Well Asterisk already have logic for both #include and #exec, if we can utilize it more code coverage :)
- Paul
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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20110617/b6762766/attachment-0001.htm>
More information about the asterisk-dev
mailing list