[asterisk-dev] [Code Review] Fix IPv6 attended transfer test

[Matt Jordan]

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asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/hangup_call.py
<https://reviewboard.asterisk.org/r/2147/#comment13968>

    Even though this is a very small file, it still needs the standard preamble.



asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/test-config.yaml
<https://reviewboard.asterisk.org/r/2147/#comment13970>

    This should state that its the IPv6 version of the attended transfer tests.



asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/test-config.yaml
<https://reviewboard.asterisk.org/r/2147/#comment13969>

    It may be worthwhile updating the test description to explain how the various SIPp scenarios interact.
    
    In particular, this should not who initiates the attended transfer.


- Matt


On Oct. 4, 2012, 1:48 p.m., opticron wrote:
> 
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> https://reviewboard.asterisk.org/r/2147/
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> 
> (Updated Oct. 4, 2012, 1:48 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> This is nearly a complete rewrite of the IPv6 SIP Attended Transfer Test using SIPpTest with AMIEventModule in the configuration-driven test framework and SIPp's 3PCC extended mode for higher level call control and message passing among SIPp instances.
> 
> The test had previously been set to be skipped because it was failing randomly, presumably because of race conditions in the test itself.
> 
> This also resulted in a few enhancements to the testsuite to allow for the use of IPv6 targets (already committed).
> 
> 
> This addresses bug SWP-4661.
>     https://issues.asterisk.org/jira/browse/SWP-4661
> 
> 
> Diffs
> -----
> 
>   asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/configs/ast1/sip.conf 3476 
>   asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/hangup_call.py PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/run-test 3476 
>   asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/slave_cfg.conf PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/uac-call.xml PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/uac-calls-and-refer.xml PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/uas-no-hangup.xml PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/test-config.yaml 3476 
> 
> Diff: https://reviewboard.asterisk.org/r/2147/diff
> 
> 
> Testing
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> 
> The test runs (seemingly) reliably on my box.
> 
> 
> Thanks,
> 
> opticron
> 
>

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