I receive phone calls in just fine but when I dial out it sometimes dial out to a different phone number and its always the same phone number it calls to and that person is never happy.
I also, have issues trying to connect my second SIP phones at different location. I had setup phone to my public ip address but it still won’t connect.
I have freepbx installed as a virtual machine on windows server 2016. My server is setup as Nat server and my server is behind ISP router. I’m using a as400 fxo gateway that connects to magicjack go. I have 2 SIP phones.
I doubt that this is a supported configuration, and I’m pretty sure the Magic Jack people are looking for you now. This definitely violates their Usage Policy.
Do you know what number the failing call reaches? Is it related to the number you are trying to call? On a failing call, does the FreePBX CDR show the desired destination or the incorrect one? I know nothing about the AS400, but assume that it can also log calls (using a debug command if there is nothing better). Does it show the correct number being dialed? Approximate percentage of calls that fail?
To use external extensions, the SIP port (defaults are 5060 for pjsip and 5160 for chan_sip) and the RTP port range must be forwarded in your ISP router to the private IP address of the PBX. Your VM’s network interface should be bridged, i.e. it gets its IP address from the ISP router, avoiding double NAT.
Aside from the ToS issue noted by jynjut, using MJ via FOX is a pretty low quality and limited solution. Get a real SIP trunk. One that’s easy to set up is Localphone; US $0.005/min. + $0.99/mo. for a number. Calls to toll-free numbers are free, so you can test without making a payment. Google Voice is free but more complex to set up. Others to consider include Callcentric, Anveo, Flowroute, Vitelity (better quality, reliability, support, but more expensive), or AnveoDirect, ThinQ, AlcazarNetworks (complex but very low cost).