[asterisk-users] load balancing SIP extensions

[Andres Jimenez]

On Fri, Feb 22, 2008 at 5:49 PM, Vieri <rentorbuy at yahoo.com> wrote:


>  Thanks. I'll try that although I hope it won't go into
>  an infinite loop between the 2 servers.

You are right. That could happen if the phone is not registered anywhere


You can put some security in the dialplan.
 if calls comes from IAX it means that PHONE is not registered in the
other server.
Just create special extensions to take the IAX calls (instead of GoTo):

PHONE  is 101

SERVER 1

exten => 101,1, Dial SIP/101
exten => 101,1, Dial IAX-SERVER2/55101

exten => 55101,1, Dial SIP/101
exten => 55101,1, Hangup

SERVER 2

exten => 101,1, Dial SIP/101
exten => 101,1, Dial IAX-SERVER1/55101

exten => 55101,1, Dial SIP/101
exten => 55101,1, Hangup


I hope it helps,


-- 
Andres Jimenez

GPG : http://www.andresin.com/gpg/gandresin@gmail.com.asc