Thanks David, That is exactly what we had to do. We got some help from Digium as well and have it taken care of.
Lane Hoskins, MCP Network Engineer 540.767.7626 -----Original Message----- From: David Gomillion [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 10:33 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] inbound call routing problem Lane Hoskins <> wrote: > I have come to a stumbling block. > > We have 8 lines coming into an ADTRAN channelbank that then goes to > the * server via a T100P card. I need to route lines 1 and 2 to > everyone when a call comes in on either of them. I also need lines 3 > - 8 to ring first at specific sip extensions (direct dials for staff > here) and then to go to voicemail or fwd to a cellphone after that if > the extension is not answered. Has anyone done this that could > provide an example for me or point me to better documentation? We > have searched extensively and not found anything yet. > > Lane Hoskins, MCP > Network Engineer > 540.767.7626 I have not done it yet, but it would seem to me that the key to this exercise would be having 7 contexts: 1 for lines 1+2 (which rings all lines or a queue or IVR/ACD) and then one for each line 3-8. This means that each of your incoming lines can have their very own s extension. You can define each line's context in the .conf in Asterisk's etc directory. Hope this helps, David Gomillion _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
