I think I have 4 options. 1, Modify chan_sip.c to update a new field in sipusers realtime table with the status of the sip peer/user. Then use agi to dial sip calls. Check the status field if OK then dial the fullcontact from the sip table. If not goto voicemail or where ever else I want the call to go.. The UA would only register to one server, so only one server *should* be writing to the database. (If not the code will be modified to do so.) Other servers would be read only from my AGI.
2, Use Asterisk management interface to find the status of the sip peer. Then dial fullcontact if peer is active. Should be easy to implement. Problem is I would have to actively poll each server in the farm. 3, Use SER as the sip router and asterisk as an application/media server. Then all sip UA would register to the SER. Should scale higher, but does add a level of complexity. 4, Continue to use IAX trunk to dial the other switch. Then hope that realtime has been improved by the time I need the 3rd server. It is a failover but not a load balancer. Any thoughts? Am I completely off here? Doug -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Bowyer Sent: Wednesday, January 04, 2006 7:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: Using *RT for HA purposes was: [Asterisk-Users] RealtimeMultipleAsterisk boxes, iaxusers On 04/01/06, Alistair Cunningham <[EMAIL PROTECTED]> wrote: > Tijmen, > > We use SER for this to load balance across multiple Asterisks. We then > use a custom program to monitor the health of the Asterisks and update > SER's configuration should one go down. 2 SERs share a single IP address > for users to contact using heartbeat. I was thinking along the same lines, but for a dynamic setup it should be possible to have SER/OpenSER load balance REGISTER requests according to some strategy/metrics, and then forward INVITEs and other call-related traffic to the 'right' back-end server. Probably lots of reasons why this is too complicated, though.... Peter -- 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 _______________________________________________ --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
