[asterisk-dev] [Code Review] External test for verifying SIP-related CHANNEL parameters

[Mark Michelson]

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/asterisk/trunk/tests/sip_channel_params/sipp/call.xml
<https://reviewboard.asterisk.org/r/589/#comment3836>

    Hmm, while it doesn't actually cause the test to fail, I should update the From and Contact headers of the BYE to mirror those used in the INVITE.



/asterisk/trunk/tests/sip_channel_params/test.lua
<https://reviewboard.asterisk.org/r/589/#comment3837>

    yuck


- Mark


On 2010-03-25 11:44:34, Mark Michelson wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/589/
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> (Updated 2010-03-25 11:44:34)
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> 
> Review request for Asterisk Developers.
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> 
> Summary
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> The summary says it nicely. In this test, a SIPp client calls Asterisk. While on the call, the lua script will connect to Asterisk via AMI and query the values of SIP-related parameters to the CHANNEL dialplan function.
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> 
> Diffs
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>   /asterisk/trunk/tests/sip_channel_params/configs/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/sip_channel_params/configs/rtp.conf PRE-CREATION 
>   /asterisk/trunk/tests/sip_channel_params/configs/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/sip_channel_params/run-test PRE-CREATION 
>   /asterisk/trunk/tests/sip_channel_params/sipp/call.xml PRE-CREATION 
>   /asterisk/trunk/tests/sip_channel_params/sipp/register.xml PRE-CREATION 
>   /asterisk/trunk/tests/sip_channel_params/test.lua PRE-CREATION 
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> Diff: https://reviewboard.asterisk.org/r/589/diff
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> 
> Testing
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> I have run this test many times and have ensured that the results are correct.
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> 
> Thanks,
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> Mark
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>