[asterisk-users] evaluate SIP response codes in dialplan

[Philipp Kempgen]

Johansson Olle E schrieb:
>>
>>
>> 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.

I know it's not an easy task if you'd want it to be done properly.
But then again Asterisk is not a SIP softswitch but a PBX.  :-)
I've never seen people who are asking for SIP_CAUSE expect it
to work under all circumstances. All the use cases are pretty
simple:

	Dial(SIP/buddy);  // single argument

When dialling to more than 1 SIP peer

	Dial(SIP/busy&SIP/answers_the_call);

the best thing to do would be to store the last cause code that
we receive i.e. the one of the peer who answered.

In a multi-protocol situation

	Dial(SIP/busy&IAX/answers_the_call);

I don't expect SIP_CAUSE to be anything meaningful. It could be
set to "000" or somesuch.


   Philipp Kempgen

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