Hi Olle, 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.
The "if not needed" part causes lots of discussions in this case. > As Josh and I've stated, we have the > HANGUPCAUSE that gives you this level of detail, but in a > multiprotocol way. Some (no so) subtle differences get lost. > It would be really bad if I had to > write one app for every protocol covered by my dialplan. True. But it would be a plus if you *could* do that in order to fine-tune the behavior if you wanted to. I still think we need a SIP_CAUSE channel variable. :-) Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany -> http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- _______________________________________________ -- 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