[asterisk-dev] [Code Review] Add SIP INFO DTMF test to the Asterisk Test Suite

[Paul Belanger]

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/asterisk/trunk/tests/channels/SIP/info_dtmf/configs/ast1/sip.conf
<https://reviewboard.asterisk.org/r/1723/#comment10058>

    Yes, and that is the problem.  Not in this case when we launch 1 instance of asterisk, but 2 or more then each instance will attempt to bind to the same 0.0.0.0 address.
    
    I updated manager.conf a while back to automagically handle the bindaddr, when I have some spare time I want to do the same for chan_sip.  Then we don't have to worry about it.


- Paul


On Feb. 9, 2012, 9:15 a.m., Matt Jordan wrote:
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> (Updated Feb. 9, 2012, 9:15 a.m.)
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> 
> Review request for Asterisk Developers.
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> Summary
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> This adds a test to the Asterisk Test Suite to cover the regression that was introduced in ASTERISK-18924.  This test sends a sequence of SIP INFO requests containing DTMF events covering 0-9, A-D, a-d, and 10-16.  This checks only for SIP INFO requests that have a Content-Type of application/dtmf-relay; the test could be expanded to cover the less used Content-Type of application/dtmf if we feel its necessary.
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> This addresses bug ASTERISK-19290.
>     https://issues.asterisk.org/jira/browse/ASTERISK-19290
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> Diffs
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>   /asterisk/trunk/tests/channels/SIP/info_dtmf/configs/ast1/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/info_dtmf/configs/ast1/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/info_dtmf/run-test PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/info_dtmf/sipp/dtmf-relay.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/info_dtmf/sipp/dtmf.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/info_dtmf/test-config.yaml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/tests.yaml 3027 
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> Diff: https://reviewboard.asterisk.org/r/1723/diff
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> Testing
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> Thanks,
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> Matt
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>

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