[asterisk-users] evaluate SIP response codes in dialplan

[Philipp Kempgen]

Hi Olle,

Johansson Olle E schrieb:
> 14 jan 2009 kl. 18.57 skrev Philipp Kempgen:
>> Klaus Darilion schrieb:
>>> Philipp Kempgen schrieb:
>>>> Klaus Darilion schrieb:
>>>>> Is it somehow possible to evaluate the SIP response code inside the
>>>>> dialplan?
>>>>
>>>> No.
>>>> Part of the reasoning is that Asterisk is meant to be a multi-
>>>> protocol PBX, not a SIP softswitch.
>>>
>>> This is IMO a stupid limitation. There are dozens of ISDN cause  
>>> codes,
>>> dozens of SIP response codes and similar in other protocols, but  
>>> Dial()
>>> only exports BUSY or CONGESTION ......
>>
>> I know. But the developers didn't want to add it.
> 
> Which is incorrect. We don't want to add expose every protocol to the  
> dialplan if not needed.

The "if not needed" part causes lots of discussions in this
case.

> As Josh and I've stated, we have the  
> HANGUPCAUSE that gives you this level of detail, but in a  
> multiprotocol way.

Some (no so) subtle differences get lost.

> It would be really bad if I had to  
> write one app for every protocol covered by my dialplan.

True. But it would be a plus if you *could* do that in order to
fine-tune the behavior if you wanted to.

I still think we need a SIP_CAUSE channel variable. :-)


   Philipp Kempgen

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