[asterisk-dev] [Code Review] External test for verifying SIP-related CHANNEL parameters

[Nick Lewis]

> On 2010-03-25 12:34:17, Nick Lewis wrote:
> > /asterisk/trunk/tests/sip_channel_params/sipp/call.xml, line 10
> > <https://reviewboard.asterisk.org/r/589/diff/2/?file=8909#file8909line10>
> >
> >     I am not really sure of the purpose of this test but the calleridname and calleridnum could be pushed a little harder e.g.
> >     
> >     From: "ben&jerry; mailto:bj at bj.com"<sip:+44(0)3303338258_b&j@[local_ip]:[local_port]>;tag=[call_number]
> 
> Mark Michelson wrote:
>     The purpose of this test was to exercise the new rtpsource options that I added in review 542. I basically expanded the test to make sure that the values we read when executing the CHANNEL() function are what we would expect them to be. I just used the strings "wienerschnitzel" and "kartoffelsalat" so there would be no confusion as to where these values were derived in the original scenario.
>     
>     While this wasn't necessarily meant to be a stress test, a second scenario could be added which has more bizarre elements to retrieve.

Thanks for explaining the purpose. I remember that some other dialplan functions have problems with the & character and with unicode so it may be worth using a calleridname with some of these characters


- Nick


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/589/#review1760
-----------------------------------------------------------


On 2010-03-25 11:48:39, Mark Michelson wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/589/
> -----------------------------------------------------------
> 
> (Updated 2010-03-25 11:48:39)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> The summary says it nicely. In this test, a SIPp client calls Asterisk. While on the call, the lua script will connect to Asterisk via AMI and query the values of SIP-related parameters to the CHANNEL dialplan function.
> 
> 
> Diffs
> -----
> 
>   /asterisk/trunk/tests/sip_channel_params/configs/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/sip_channel_params/configs/rtp.conf PRE-CREATION 
>   /asterisk/trunk/tests/sip_channel_params/configs/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/sip_channel_params/run-test PRE-CREATION 
>   /asterisk/trunk/tests/sip_channel_params/sipp/call.xml PRE-CREATION 
>   /asterisk/trunk/tests/sip_channel_params/sipp/register.xml PRE-CREATION 
>   /asterisk/trunk/tests/sip_channel_params/test.lua PRE-CREATION 
> 
> Diff: https://reviewboard.asterisk.org/r/589/diff
> 
> 
> Testing
> -------
> 
> I have run this test many times and have ensured that the results are correct.
> 
> 
> Thanks,
> 
> Mark
> 
>