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 [email protected]
from outside the office and vise verse... to dial people at
[email protected]
hope for little help
Eliezer
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--
Daniel-Constantin Mierla
http://www.asipto.com
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