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

[Mark Michelson]

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Ship it!


- Mark


On Oct. 10, 2012, 2:07 p.m., opticron wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2147/
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> (Updated Oct. 10, 2012, 2:07 p.m.)
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> 
> Review request for Asterisk Developers.
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> Summary
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> 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.
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> The test had previously been set to be skipped because it was failing randomly, presumably because of race conditions in the test itself.
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> This also resulted in a few enhancements to the testsuite to allow for the use of IPv6 targets (already committed).
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> 
> This addresses bug SWP-4661.
>     https://issues.asterisk.org/jira/browse/SWP-4661
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> 
> Diffs
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>   asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/configs/ast1/sip.conf 3490 
>   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 3490 
>   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 3490 
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> Diff: https://reviewboard.asterisk.org/r/2147/diff
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> 
> Testing
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> The test runs (seemingly) reliably on my box.
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> 
> Thanks,
> 
> opticron
> 
>

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