Sounds Good. Keep us updated!
-bk Iñaki Baz Castillo wrote: > El Wednesday 18 July 2007 21:24:21 bkruse escribió: > >> right, >> >> Thats what I thought. >> >> >> Let me explain this again... >> >> You setup extensions in your asterisk box, with callerID, or something >> similar. Then pass those to open SER over a sip trunk. >> >> Ser then takes the call, parses the callerID, and routes based on that. >> >> I have done setups similar to you, and put openser in front >> of a mass of users, and let asterisk handle inbound, except I >> usually do the opposite, but in this case, it will still work. >> > > Ok, thanks. I do that with OpenSer and Asterisk, but I didn0t know if I could > route calls to openser users with AsteriskGUI. > > So thanks, I'll try it. > > Regards. > > > > _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-gui mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-gui
