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. /O _______________________________________________ -- 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
