From overview - you can define routing tree per user.

On Mon, Feb 6, 2012 at 8:13 PM, Greg Mannie <greg@latigi.com> wrote:

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 is separate with it's own database.

I am having problems wrapping my head around the configuration I should use. Is there not a method just to add DID in the same fashion as asterisk. So I register a trunk on Kamailio and based on incoming DID it sends it to the correct asterisk server?

I have my asterisk 1.8 on a public ip address with a trunk registered to the Kamailio server which also has a public ip address.

Regards,

Greg

Quoting Stoyan Mihaylov <stoyan.v.mihaylov@gmail.com>:

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 some matching pattern in CallerID.
But if you put whole logic used in Asterisk in DB - then you dont care
which server will take the call, because you can put whole logic purely in
DB - including extensions etc.
At least I prefer to have almost nothing in extensions.conf - and
everything to stay either in DB or in AGI scripts.

My knowledge of Kamailio is very very basic - I know only few things there.
Asterisk and Kamailio can run on same server, but I cant see any reason for
that. I mean you will have lot of troubles in such case, and nothing
"good". This is only if you want to make some tests. But you can expect lot
of troubles.

On Fri, Feb 3, 2012 at 9:40 PM, Greg Mannie <greg@latigi.com> wrote:

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. I was hoping I could setup routes on the
kamailio and direct them to the appropriate asterisk server.

Initially I thought it would be as simple as setting up an inbound route
on Asterisk. Ha.. I also installed siremis 3.2 and perhaps reading on how
to use it will provide clearer details.

I know so little, I'm not even sure if I need to have Kamailio and
Asterisk running on the same server, since I only want Kamailio as a proxy.

Regards,

Greg

Quoting Stoyan Mihaylov <stoyan.v.mihaylov@gmail.com>:

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 asterisk boxes this way
$sht(forw=>$ft)=$du; #this way I store used path
I used t_relay, instead of forward, because my Asterisks are with local
IP.
Calls from Asterisk are send to Kamailio if they are to local user, or to
our SIP provider. There are no problems with calls from Asterisk to SIP
provider, even if Asterisk is behind NAT.
Asterisk accepts calls from SIP provider though registrar lines in
sip.conf. Asterisk can forward calls from our SIP provider to local users
in Kamailio.
I got problems with ACK and BYE. To solve them, I used
if(($td=="sip.name.of.**kamailio.server.com<http://sip.name.of.kamailio.server.com>
")||($si=="**IPofServer")){

$du=$sht(forw=>$ft);
}

On Fri, Feb 3, 2012 at 8:13 PM, Greg Mannie <greg@latigi.com> wrote:

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 kamailio and registering our internal asterisk servers
to the kamailio.

Much of what I read talks about using Asterisk as the PSTN interface, but
that interface is through a sip trunk purchased from a provider. Won't
Kamailio be the PSTN gateway? The idea here is to pool all the sip
trunks
from the various hosted asterisk solutions (VM running asterisk) and
point
them all to a proxy to facilitate the aggregation of traffic.

I have been reading SIP tutorials and the mailing list archives. If
anyone has a sample config and perhaps a little direction it would be
highly appreciated.

Thank you

Greg

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