[asterisk-dev] [Code Review] 'directrtpsetup' option test

[opticron]

-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2185/#review7400
-----------------------------------------------------------



/asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_A.xml
<https://reviewboard.asterisk.org/r/2185/#comment14182>

    You can add this after the </recv> to avoid the obscure use of strcmp:
    <Reference variables="1"/>



/asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_B.xml
<https://reviewboard.asterisk.org/r/2185/#comment14181>

    You can add this after the </recv> to avoid the obscure use of strcmp:
    <Reference variables="1"/>



/asterisk/trunk/tests/channels/SIP/directrtpsetup/test-config.yaml
<https://reviewboard.asterisk.org/r/2185/#comment14183>

    Is there anything preventing this from running on earlier versions of Asterisk?


- opticron


On Nov. 15, 2012, 7:07 a.m., Joshua Colp wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2185/
> -----------------------------------------------------------
> 
> (Updated Nov. 15, 2012, 7:07 a.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> This test covers the 'directrtpsetup' option present within chan_sip. It creates a call going through Asterisk and examines the SDP to ensure that the address for media is of the caller, and not that of Asterisk. It also examines the answer SDP to confirm that the address for media is that of the called party.
> 
> 
> Diffs
> -----
> 
>   /asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_A.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_B.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/directrtpsetup/test-config.yaml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/tests.yaml 3508 
> 
> Diff: https://reviewboard.asterisk.org/r/2185/diff
> 
> 
> Testing
> -------
> 
> Ran test to confirm it works, purposely broke test to confirm it fails. Ate some toast. It was crunchy.
> 
> 
> Thanks,
> 
> Joshua
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20121115/bbfacd12/attachment.htm>