Ok, so I've probably made this install more complicated than it needs to be but here is my issue:
Calls set up ok but there's no audio on 75 percent of the calls, so obviously a media issue (RTP)
Now the complicated bit, how the network is setup
The Fusion pbx has its own public IP on a pfsense firewall with SIP (UDP/TCP 5060, 5061) and RTP (UDP 10,000 to 40,000) nat'd to the fusion pbx box on the internal network.
The ASTPP box has its own public IP on a pfsense firewall with SIP (UDP/TCP 5060, 5061) and RTP (UDP 10,000 to 40,000) nat'd to the ATPP pbx box on the internal network. The ASTPP box is registered to a carrier via SIP trunk.
The Fusion PBX box has a gateway setup to the ASTPP box on the internal network.
You with me..
Now the kicker. The phone registers from the public network to the Fusion PBX (reserve judgment for later please) via SIP.
So like I said calls setup ok, but no audio for three quarters of the calls. Also all calls to voice mail or other internal extensions are no problem. I'm guessing all this nating is really screwing with the RTP stream somewhere. At first I thought it might be a port or codec mismatch but I've eliminated that.
My next step is to hang the ASTPP box out on the public network and just lock it down but I'd rather not. Any ideas?
Calls set up ok but there's no audio on 75 percent of the calls, so obviously a media issue (RTP)
Now the complicated bit, how the network is setup
The Fusion pbx has its own public IP on a pfsense firewall with SIP (UDP/TCP 5060, 5061) and RTP (UDP 10,000 to 40,000) nat'd to the fusion pbx box on the internal network.
The ASTPP box has its own public IP on a pfsense firewall with SIP (UDP/TCP 5060, 5061) and RTP (UDP 10,000 to 40,000) nat'd to the ATPP pbx box on the internal network. The ASTPP box is registered to a carrier via SIP trunk.
The Fusion PBX box has a gateway setup to the ASTPP box on the internal network.
You with me..
Now the kicker. The phone registers from the public network to the Fusion PBX (reserve judgment for later please) via SIP.
So like I said calls setup ok, but no audio for three quarters of the calls. Also all calls to voice mail or other internal extensions are no problem. I'm guessing all this nating is really screwing with the RTP stream somewhere. At first I thought it might be a port or codec mismatch but I've eliminated that.
My next step is to hang the ASTPP box out on the public network and just lock it down but I'd rather not. Any ideas?