[asterisk-users] evaluate SIP response codes in dialplan

[Klaus Darilion]

Hi Olle!

Currently we have the problem that due to 
SIP<->hangupcause<->SIP<->hangupcause.... conversions the original 
hangupcause gets lost in a chain of Asterisk servers using SIP.

In chan_sip there is already code for adding the X-Asterisk-Hangupcode 
header. What about reading this header on the receiving side for setting 
the hangupcause instead of doing SIP->hangupcause mapping ?

Then the original hangupcause is not lost.

regards
klaus

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.
> 
> /O
> 
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