[asterisk-users] evaluate SIP response codes in dialplan

Johansson Olle E oej at edvina.net
Mon Jan 19 03:17:18 CST 2009


>
>
> I still think we need a SIP_CAUSE channel variable. :-)
>
Then we need to start working on aggregation rules, like what if one  
IAX channel answers and one SIP channel is busy?

For SIP-only calls, we need to add a lot of code from proxy rules for  
call forking and response aggregation. It's not an
easy task.

/O



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