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

[Kevin P. Fleming]

Nick Lewis wrote:

> /asterisk/trunk/tests/sip_channel_params/sipp/call.xml
> <https://reviewboard.asterisk.org/r/589/#comment3842>
> 
>     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]

The purpose of the test is to verify that chan_sip's parameters
available in the CHANNEL() dialplan function operate correctly, so the
only value in providing a more complex CLID and CNAM in the SIP INVITE
would be if they were somehow likely to cause a problem in the dialplan
function itself... this is not a test for the SIP message or header parsers.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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