SIP hybrid server behind pfSense and setting up remote phone over openvpn - NAT?
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I wanted to figure out what is the best way to setup a sip phone remotely and I talked to a friend of mine that said I should look into NAT as my possible problem for audio problem.
I have a hybrid sip phone system internally that I thought I could use openvpn to setup a phone to work offsite. I figured it would be better to use openvpn to make the connection secure and not open up the firewall for the sip server to be opened up for access. I was able to get audio from the remote phone to cell or phone in the building, but no audio back to the remote phone. I've looked at siproxy and wondered if that would even work for our setup. Hopefully someone can steer me the right direction.Thanks in advance.
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You have a firewall rule that would allow the traffic from the LAN to the OpenVPN. If it's site-to-site OpenVPN consider the both subnets involved!
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It definitely works, from my experience in Site To Site and Road Warrior scenarios with OpenVPN and Asterisk, it works. Check if the phone in the remote site registers correctly with the PBX, you can sniff some traffic to get some impression. Last but not least check that everything is configured with the routing and that the required ports are accessible.