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@mydomain.com
from outside the office and vise verse... to dial people at bob@otherdomain.com
hope for little help
Eliezer
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:
On 02/05/2011 15:44, Daniel-Constantin Mierla wrote:
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
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Daniel-Constantin Mierla -
Eliezer Croitoru