I have a switchvox server at site a. The site has multiple external IP's and using PFSense I've routed all outgoing data from the switchvox server to a specific ip.
For NAT, I have 5060 and 10000-10500 forwarded from external ip to switchvox server.
Switchvox has NAT mapping enabled and the correct external IP.. and also the /24 set for local networks.
I've not done any test calls yet (silly I know!) but after setting it up I have gone to site b.
Site b just has a Cisco/Linksys SPA303 phone that I have set up to point to switchvox server at site a.
The phone registers fine and if I call the number, it rings... and I can hear the caller, but he cannot hear me.
On the phone In the sip setting, I've changed the RTP port to 10000 to 10500 and in EXT1 I have NAT Mapping Enable: and NAT Keep Alive Enable: both set to yes.
I have also forwarded ports 10000-10500 from the external ip on site b to the voip fone.
I'm not home till Saturday, so can't do any testing on the phone on the internal network on site A, but has anyone got any suggestions as to why the caller cannot hear me when the phone rings at site b?
Thanks in advance