[asterisk-users] evaluate SIP response codes in dialplan

[Klaus Darilion]

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 ......

thanks
klaus