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

[Paul Belanger]

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Almost there! :) I'm going to spend some time this afternoon playing with this patch.


/asterisk/trunk/tests/channels/SIP/sip_tls_call/configs/ast1/extensions.conf
<https://reviewboard.asterisk.org/r/1276/#comment7572>

    We may have to change this logic a bit, I've found that Wait() logic in the Dialplan not to be reliable. Many outside factors could affect the timing of this.
    
    For an example, look at feature_attended_transfer and how we use Background() with SendDTMF().



/asterisk/trunk/tests/channels/SIP/sip_tls_call/run-test
<https://reviewboard.asterisk.org/r/1276/#comment7569>

    These should be removed, move information below.
    



/asterisk/trunk/tests/channels/SIP/sip_tls_call/run-test
<https://reviewboard.asterisk.org/r/1276/#comment7568>

    I'm still not a fan of this code.  One thing that I do notice, we are modify configuration files out side of the sandbox'd instances of Asterisk. 
    
    As you re-run the test, we update files that previous sandbox'd versions of the test depend on.  Changing the behavior of a previous run test.
    
    I started work on something last night which should help address my comments, but requires an update to the testsuite.



/asterisk/trunk/tests/channels/SIP/sip_tls_call/run-test
<https://reviewboard.asterisk.org/r/1276/#comment7570>

    Move this to the top of your code block.  This will sandbox the 2 instances of asterisk first, allowing you to get access to self.options.base, self.options.test_name and self.asterisk.base, the path names for directories, allowing you to build the path names.



/asterisk/trunk/tests/channels/SIP/sip_tls_call/run-test
<https://reviewboard.asterisk.org/r/1276/#comment7571>

    Be sure to check the ami.id, to find out which instance of asterisk this call back is running on.  Then you can confirm you have the right instance to check the digit event.


- Paul


On 2011-06-24 10:10:21, jrose wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
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> 
> (Updated 2011-06-24 10:10:21)
> 
> 
> 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 1671 
> 
> 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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