[asterisk-users] evaluate SIP response codes in dialplan

[Johansson Olle E]

14 jan 2009 kl. 14.02 skrev 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?

Check the HANGUPCAUSE, it's much more detailed than DIALSTATUS.

We do map the SIP (and all other protocol errors in various channel  
drivers) codes to ISDN hangup causes, which gives you much more  
information about
why a call failed.

The conversion we're using follows the RFC, and where that doesn't  
cover it, Cisco's documentation.

/Olle