On 02/05/2011 15:44, Daniel-Constantin Mierla wrote:
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
Thanks Daniel
i have a running kamailio using the debian repo so i have the binaries
but i dont know where to start from.
the documents were pretty obscure to me.
if you have simple sip proxy example for
"nat" like proxy.
i have a machine that connects to a remote network and if im using
iptables to nat the sip extention\register
the register wont work but i can do outgoing calls without registering.
so i need to use the middle machine to be a proxy for the sip
server.(asterisk)
i will read more about it..
Thanks
Eliezer
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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