Hello, I've found a little documentation on voip-info and on the asterisk- users list, although I was hoping for an example of a tried-and-true failover setup between PRI and SIP.
We are an outgoing call center that uses asterisk 1.4 connected to 2 PRIs from the local telephone company in one group (g1) and a SIP trunk from bandwidth.com. The PRIs are the primary outgoing service, however we have been experiencing some issues where one or both of them can fail randomly. We are working with the telephone company to have this resolved. In the meantime, we want to have a good failover solution where if both PRIs fail, asterisk will dial out through the SIP trunk. I've found solutions as simple as two Dial commands one after the other, and others where the failover Dial is in a jump to CONGESTION. Unfortunately we don't have a testing environment, so the solution really has to work. Does anyone else on the list have a PRI to VoIP failover setup that's worked for them in a high volume environment? Thanks! Jason Martin Metrix Matrix, Inc. 785 Elmgrove Rd, Bldg 1 Rochester, NY 14624 Office: 888-865-0065 x202 Mobile: 585-705-1400 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
