On 13/03/06, Douglas Garstang <[EMAIL PROTECTED]> wrote: > Now that I've read that paragraph of Kevin's a few times, it strikes me that > this is not a redundant configuration. If the call is handled by the Asterisk > system where the phone registered, what happens if that system becomes > available? Can another system (one that did not handle the registration) > process the call?
(Any chance you could format your emails for easier quoting? Thanks) Something like this: Server A [sip-registrations] exten => peer1,2,Dial(SIP/peer1) exten => peer2,2,Dial(SIP/peer2) include => switch-server-b [switch-server-b] switch => IAX/user:[EMAIL PROTECTED]/sip-registrations So a call arriving in context sip-registrations will hit any peer which has registered (with the regcontext trick), and fall through to the 'switch' for any which hasn't. Server B has the opposite. This won't help with a failure for an in-progress call, but should automatically distrubute calls around your peers which are registered with one server or the other. If the phones know how to re-register in the event of a server failure (and I think you said you use a SRV-based system for this), then something good should be able to happen. -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 _______________________________________________ --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
