16 Feb
2009
16 Feb
'09
4:40 p.m.
Hello,
your issue has nothing to do with a call stateful server.
You simply do record route in your config so for within dialog requests
the routing is don properly.
For initial invite, I assume (as I never used it) carrierroute sets the
R-URI so you just use:
t_relay("udp:asteriskip:asteriskport");
and in asterisk you dial the address in the r-uri.
If carrierroute is setting the dst_uri, then add it as special header,
use same to forward to asterisk, but there use the special header to
dial out.
Cheers,
Daniel
On 02/16/2009 02:54 PM, Juan Asencio wrote:
Hi list,
I need some explanation about using Kamailio as a Call Stateful Proxy.
I understand that Kamailio is a transaction stateful proxy. And I would
like to learn how I can achieve a Call Stateful Proxy using Kamailio.
I'm going to use Asterisk for transcoding to receive and send calls to the
users and Kamailio should do the routing process using carrierroute,
deciding to which carrier the calls should be sent to.
Customer Carrier1
Group1 \ /
\ /
\ /
Customer ---->GW ---> Kamailio ---> GW ---> Carrier2
Group2 Asterisk Asterisk
/ \
/ \
Customer / Carrier3
Group3
Now it works with x-lite, if customer1 calls to certain number the call is
send to the correspondent carrier base on the prefix number.
But when I add the GWs, how I can send the call to the GW IP address
(Asterisk) and then forward to the IP address of the carrier that I
assigned for that prefix number in my carrierroute table.
Can any of you could help me to clarify this or guide me about what to
read, so I could implement this.
Thanks in advance,
Juan.-
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Daniel-Constantin Mierla
http://www.asipto.com