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. 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
