[asterisk-users] evaluate SIP response codes in dialplan

[Johansson Olle E]

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

The most important feature of Asterisk is that it's a multiprotocol  
PBX. Even if I think there's only one protocol for the future, there's  
still a lot of old stuff out there and the beauty is that I can  
produce services in asterisk covering all of these without knowing the  
details of all these protocols. It would be really bad if I had to  
write one app for every protocol covered by my dialplan.

/O