I suppose this is better for you:
http://kamailio.org/docs/modules/3.2.x/modules/carrierroute.html#id2548186
But I never used it - we just did not need anything alike.
>From overview - you can define routing tree per user.
On Mon, Feb 6, 2012 at 8:13 PM, Greg Mannie wrote:
> Thank you again for
Thank you again for your help.
I have been reading on the dispatcher module and have a question. It
would seem many people use Kamailio as more than proxy and register
sip extensions against it. Since we host virtual pbx (asterisk 1.8,
Freepbx) for a few different clients, each instance i
I dont know about siremis, but you can forward calls to different groups of
Asterisk servers - using
ds_select_dst(set, alg);
Where set is set of Asterisk servers - you can check that module.
The problem is - I have no idea how you can select different sets in
kamailio.cfg, except by length, or som
Thank you for your detailed response. Sorry for the trouble but would
you be able to also answer the following.
Do you know if this same type of deployment would be suited to our
needs. Many of the Asterisk servers we host are for clients, who have
their own extensions, voicemail, ivr etc
We were in similar situation. Many years with Asterisk and then we were
forced to use ser - and we preferred Kamailio.
Now we do:
Kamailio has global IP address and clients register to it.
Kamailio forward all calls to Asterisk boxes using following:
ds_select_dst("1","4");#You can use many aster
Hello
After much reading I have come to the realization that after years of
using Asterisk I know very little about Sip.
I have my Kamailio box up, I have Asterisk 1.8.x running with realtime
working. I thought it would be just a case of registering SIP trunks
from my provider to the kam
