That is very nice, but where are the HANGUPCAUSE values documented? Thanks.
on Thursday 01/15/2009 Johansson Olle E([email protected]) wrote > > 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 > > _______________________________________________ > -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [email protected] _______________________________________________ -- 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
