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 :-) 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