[asterisk-users] evaluate SIP response codes in dialplan

[SIP]

Take a look (if it still exists) at the Asterisk B2BUA project. It has a 
patch that adds direct access to SIP response codes. It takes a little 
modification of the patch file to use in some of the newer asterisks 
(and to strip out the one codec option that's somewhat irrelevant), but 
it's a good starting spot.

N.

Klaus Darilion wrote:
> 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
>
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