I have a single server running asterisk 1.6.0.8 with a few sip voip providers and a tdm card for redundancy. It has a caching name server and the sip providers are hard coded in the hosts file.
When the internet connection dies, it fails over to the dahdi channel as it should, but slowly the sip phones loose registration and the incoming dahdi channel can still answer the incoming call, but it doesn't pass it off the mailbox, it just says the "person at extension... is not available"? There is a custom recording setup that otherwise works? What does a guy got to do to keep asterisk up when the net connection fails? This is becoming a show stopper :( Thanks! jlc _______________________________________________ -- 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
