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 > > _______________________________________________ > -- 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 > _______________________________________________ -- 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
