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 ...... thanks 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
