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 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
