Fred Posner wrote: > If you're using just SIP to SIP, a better option would be a pure sip proxy, > ala Kamailio/SER, etc. They can survive a failover without a drop.
Agreed. Even if using transaction-stateful relay mode, as long as a dialog is nailed up, sequential in-dialog messages (re-INVITEs, BYEs, etc.) can be routed based on the Route header even if the runtime transaction state has been lost. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 _______________________________________________ -- 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
