Johansson Olle E schrieb: > 15 jan 2009 kl. 12.42 skrev Klaus Darilion: > >> >> Johansson Olle E schrieb: >>> 14 jan 2009 kl. 18.57 skrev Philipp Kempgen: >>> >>>> Klaus Darilion schrieb: >>>>> Philipp Kempgen schrieb: >>>>>> Klaus Darilion schrieb: >>>>>>> Is it somehow possible to evaluate the SIP response code inside >>>>>>> the >>>>>>> dialplan? >>>>>> No. >>>>>> Part of the reasoning is that Asterisk is meant to be a multi- >>>>>> protocol PBX, not a SIP softswitch. >>>>> This is IMO a stupid limitation. There are dozens of ISDN cause >>>>> codes, >>>>> dozens of SIP response codes and similar in other protocols, but >>>>> Dial() >>>>> only exports BUSY or CONGESTION ...... >>>> I know. But the developers didn't want to add it. >>> Which is incorrect. We don't want to add expose every protocol to the >>> dialplan if not needed. As Josh and I've stated, we have the >>> HANGUPCAUSE that gives you this level of detail, but in a >>> multiprotocol way. >>> >>> The most important feature of Asterisk is that it's a multiprotocol >>> PBX. Even if I think there's only one protocol for the future, >>> there's >>> still a lot of old stuff out there and the beauty is that I can >>> produce services in asterisk covering all of these without knowing >>> the >>> details of all these protocols. It would be really bad if I had to >>> write one app for every protocol covered by my dialplan. >> That's OK. HANGUPCAUSE is OK. Nevertheless a configurable mapping >> cause >> codes <-> SIP response codes would be nice :-) > Absolutely - contact me off line to discuss such a project :-) > > In the meantime, we could document this a bit better.
Yes - for example a note in the documentation of DIALSTATUS which refers to HANGUPCAUSE. One of the problems with hangupcause is, that is might get changed from one Asterisk to another - e.g. Hangup(3) generates a SIP 404 response which gets translated to hangupcause 1. So, a mechanism to signal Asterisk hangupcauses from one Asterisk to another Asterisk would be nice. IIRC I once saw a prorietary Asterisk header (X-Hangupcause or similar) in a SIP response, but I could find it currently. regards klaus regards 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
