[asterisk-users] evaluate SIP response codes in dialplan

[Johansson Olle E]

15 jan 2009 kl. 12.42 skrev Klaus Darilion:

>
>
> 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. 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.
>
> That's OK. HANGUPCAUSE is OK. Nevertheless a configurable mapping  
> cause
> codes <-> SIP response codes would be nice :-)
Absolutely - contact me off line to discuss such a project :-)

In the meantime, we could document this a bit better.

/O