[asterisk-users] evaluate SIP response codes in dialplan
Johansson Olle E
oej at edvina.net
Mon Jan 19 08:09:24 CST 2009
19 jan 2009 kl. 11.10 skrev 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:
Well, but if we implement a half-done implementation, we will get a
ton of bug reports within days... We can't do it like that, Philipp.
(Well, looking at TLS/TCP in 1.6 I guess we can do anything... ;-) )
/O
>
>
> 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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* Olle E Johansson - oej at edvina.net
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