I've configured our PBX so that when a user dials 80 on the PBX extension, it goes out an ISDN TE interface on the PBX and into an NT interface on my asterisk machine, where it jumps into the 's' extension.
Asterisk then does a DISA(no-password|sip_provider_out) which allows the call to go out via a sip provider, to give us cheaper calls. Unfortunately if the user doesn't wait for DISA to give dialtone, asterisk doesn't hear all of the digits. When they dial '0' on the PBX there is no need to wait for dialtone, so it is a bit confusing for the users. Any suggestions? I'm using misdn, and 'immediate' and 'always_immediate' in the config for that port, so maybe there is something I could do there to take whatever digits have been dialled so far... Thanks James _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
