Hi, Just got hold of Ericsson webswitch 100 G4 (4 FXO ports). IT uses H323 as codec. The plan is to use it for incoming/outgoing calls on two PSTN lines. I have ATA 186 which is using SIP to use asterisk services.
I can not figure out: 1. where in asterisk do I edit conf files so it uses webswitch for incoming/outgoing calls 2. how come there is no username+password on webswitch. (apparently just an IP and port number) 3. is there any docs on gateway/gatekeeper for using H323 with asterisk. If some could please explain this to me, I would be very grateful. Thanks Senad _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
