Asterisk redundant

I am working an a multiple box asterisk solution. I need phones to be able to login to multiple asterisk servers. I need Phone A to be able to register to switch A and call Phone B that is registered to switch B.

With rtcachfriends=no this can be done, However I then loss MWI and “sip show peers� plus if a Phone becomes unreachable the phone I get dead air until the dial timeout reached.

With rtcachfriends=yes I get MWI & “Sip show peers�, However I cannot call phones that register to a different switch.

My current working solution is to have rtcachfriends=yes. Place the call via sip if dialstatus= chanunavaliable
I then route the call to the other switch via an IAX trunk. Everything works but I don't have a true load balance soltuion. Plus it really only works for 2 boxes. It get out of hand when I add more..

I have tried using AGI and dialing the “full contact� found in the SIP realtime table. It works if the phone is active, but if the phone is no active I get dead air until the dial timeout is reached. This will not work as I cannot have 12 sec of dead air. So is there a way know the status of a SIP UA? It is it in the SIP realtime data? I looked at regseconds but it does not seem to be it because I can have a UA that is unreachable and the regseconds are not expired.

Could realtime be altered to add a status filed to the SIP realtime table?

Or is there a asterisk configuration option that I missed?
I am working an a multiple box asterisk solution. I need phones to be able to login to multiple asterisk servers. I need Phone A to be able to register to switch A and call Phone B that is registered to switch B.

With rtcachfriends=no this can be done, However I then loss MWI and “sip show peers� plus if a Phone becomes unreachable the phone I get dead air until the dial timeout reached.

With rtcachfriends=yes I get MWI & “Sip show peers�, However I cannot call phones that register to a different switch.

My current working solution is to have rtcachfriends=yes. Place the call via sip if dialstatus= chanunavaliable
I then route the call to the other switch via an IAX trunk. Everything works but I don't have a true load balance soltuion. Plus it really only works for 2 boxes. It get out of hand when I add more..

I have tried using AGI and dialing the “full contact� found in the SIP realtime table. It works if the phone is active, but if the phone is no active I get dead air until the dial timeout is reached. This will not work as I cannot have 12 sec of dead air. So is there a way know the status of a SIP UA? It is it in the SIP realtime data? I looked at regseconds but it does not seem to be it because I can have a UA that is unreachable and the regseconds are not expired.

Could realtime be altered to add a status filed to the SIP realtime table?

Or is there a asterisk configuration option that I missed?
Created by: rehan, Last modification: Thu 29 of Dec, 2005 (23:33 UTC)
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