Hello- We upgraded our phone system from an analog system to a Digium Switchvox D65 PBX. I would like to replace our home brewed Linux router with pfsense 2.0.2 but am having trouble making a good phone connection. While the home-brewed router has worked well in the past, traffic-shaping is not one of its strengths.
Our PBX sits inside the firewall with 15 SIP phones connected to it. The PBX is connected via SIP to our SIP provider. What I did was to create port forwarding for a variety of ports mainly 5060 UDP and 10000-20000 UDP in the NAT section. I saw that the port forwarding rules were automatically created. So far so good. However when we attempt to connect from the outside, all we hear is blip blip blip. When we attempt to call from the inside, the connection is made but there is no sound. I saw a bunch of support documents at the pfsense web site. One document suggested to install sipproxy and configure it for the internal PBX. I also came across an external document (http://www.voipvoip.com/switchvox/) suggesting turning on NAT port forwarding within the Switchvox configuration (see the Notes section toward the end). I tried this with the Linux router but it didn't like it. I haven't tried that with pfsense. Another consideration would be to assign a public IP address to the PBX box and have all SIP phones connect to it from the inside- would that be the better scenario? What I would be interested in knowing at this point what your experiences with attempting to set up a SIP connection from an internal PBX box with a SIP provider (in our case, Cbeyond) using a pfsense 2.0.2 box. Are there any recommendation that you could offer to effect a fully functional SIP connection? Most of the SIP phones are internal but it is planned to have a few outside the LAN as well. Would it be more effective to place the PBX outside the LAN? Any thoughts that you can offer would be greatly appreciated! ~Doug
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