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. -- Iñaki Baz Castillo [EMAIL PROTECTED] _______________________________________________ --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
