[asterisk-users] evaluate SIP response codes in dialplan

[Klaus Darilion]

Hi!

Is it somehow possible to evaluate the SIP response code inside the 
dialplan?

I have an Asterisk server which forwards requests to various PSTN 
gateways with SIP. If the Dial() attempt is not successful I want to 
differ at least these 3 options:
- called destination is busy (486): e.g. activate auto-redial
- called destination does not exist, unassigned number (404)
- gateway is broken, error, circuit busy (e.g. 503)

486 is mapped to DIALSTATUS=BUSY
but both 503 and 404 is mapped to DIALSTATUS=CONGESTION

As when Asterisk forwards the response with SIP to the caller the same 
response code is used, I suspect this information must be stored 
somewhere inside the channel variable. So, are there any means to access it?

thanks
klaus