Hello,
if kamailio is not running yet for you, check this installation tutorial:
http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git
When you need to bridge sip and rtp, kamailio does it for sip and you
have to use rtpproxy to do the bridging for rtp - there is an example
config in the rtpproxy module.
Cheers,
Daniel
On 5/1/11 1:28 AM, Eliezer Croitoru wrote:
i have a main office and a branch office.
the main office and the branch office connected via vpn connection the
branch office is natted to the main office.
so i want to put on the gateway machine a SIP proxy server to nat the
sip connection and also to proxy all the connections to SIP uses of
the office.
means i have one asterisk machine on the main site and other SIP
providers that in a case i want to use i can use them also without
connecting to the main office.
i dont need asterisk on the branch office.
the gateway server is ubuntu 10.04 amd64
what i should use Kamailio or the openser on the ubuntu repos?
for now the Kamailio is installed.(not working).
another feature i want to add is to be able to make calls between SIP
phones using this server such as berni(a)mydomain.com
from outside the office and vise verse... to dial people at
bob(a)otherdomain.com
hope for little help
Eliezer
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Daniel-Constantin Mierla
http://www.asipto.com