Douglas Garstang wrote: > I know there's bugs open on this.
This is not a bug. There is no practical way to handle a SIP client who tries to transfer a call between Asterisk servers directly. The proper way to handle is this to ensure that your proxy/load balancer ensures that all SIP calls placed by a phone go to the same Asterisk server as long as that phone has any active calls. It should only randomly pick a server when it is placing a call and has nothing else active. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
