15 jan 2009 kl. 13.02 skrev John covici: > That is very nice, but where are the HANGUPCAUSE values documented? That's the issue...
include/asterisk/causes.h is a good reference for now. /O > > > Thanks. > > on Thursday 01/15/2009 Johansson Olle E(o...@edvina.net) 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 > cov...@ccs.covici.com > > _______________________________________________ > -- 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 --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden _______________________________________________ -- 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