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 [email protected]

from outside the office and vise verse... to dial people at [email protected]


hope for little help


Eliezer




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