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

[Mark Michelson]

> On Oct. 4, 2012, 4:12 p.m., Paul Belanger wrote:
> > asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/hangup_call.py, lines 1-8
> > <https://reviewboard.asterisk.org/r/2147/diff/1/?file=31724#file31724line1>
> >
> >     I'm trying to understand what we are doing here with the new yaml files.  But from the looks of it, we need to create a python file to call AMI events, is that correct?
> >     
> >     Is there not the ability to directly call AMI events with parameters via the yaml file?

<pedantic>
You don't call AMI events. You call AMI commands, also known as AMI actions
</pedantic>

There's nothing yet that allows for this. The best you can do is to register a callback that can be called when an AMI event occurs. Matt Jordan currently has a review up [1] that starts to pave the way for performing actions (be they AMI or otherwise) based on the progress of the test.

[1] https://reviewboard.asterisk.org/r/2130/


- Mark


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2147/#review7225
-----------------------------------------------------------


On Oct. 4, 2012, 1:48 p.m., opticron wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2147/
> -----------------------------------------------------------
> 
> (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
> -------
> 
> The test runs (seemingly) reliably on my box.
> 
> 
> Thanks,
> 
> opticron
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20121004/a14a6d05/attachment.htm>