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?
This is my first post so please forgive me if I posted
this in the wrong list.
Many
thanks!
Doug
Gillespie
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