I need OpenSIPS 1.6.4 configuration that will just transparently route to our Asterisk 1.6.2.x Server every SIP connection it gets (also you need to use rtpproxy).
No DB backend. No OpenSIPS authentication of peer. Just routing of SIP and RTP traffic to Asterisk.
Your job is to make a working [url removed, login to view] file to route SIP traffic between a SIP UA and Asterisk with the help of rtpproxy. SIP UA should connect to the IP of OpenSIPS, and OpenSIPS should route everything transparently with the help of rtpproxy to Asterisk.
Why I need this? Because due to some routing problems I'm unable to connect some client phones in some cities to our main Asterisk server. I've setup a server that is visible to those clients but I want it to be only a SIP router. I will provide client with alternative router's IP and client should be able to connect to my Asterisk server using that IP.
Please only bid if you can 100% do it and have experience. Don't waste my and your time.
I will give info on the VPS server with OpenSIPS and RTPProxy and 2 SIP accounts to test connectivity.
Hello, I have a 5 years experience in the VoIP area with Asterisk/OpenSIPS and another applications. I can quickly finish this project and configure all requested there. Feel free contatc me if you require more i Plus