[asterisk-users] evaluate SIP response codes in dialplan

[Johansson Olle E]

>
>
> 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