No audio between phones

I have the FreePBX with CENTOS on our school network on one ISP and our Church network is on another ISP. Our PBX is connected to a netgear FVS318. I have rule that allows traffic from the church’s network to the school across ports 5060. If I am on the Church’s network I can call through the cloud to the school. when we pick up the phone, we aren’t getting audio.
I’m sure it’s a simple fix, but not sure what I’m missing. Any help would be appreciated.


The payload, in your case audio, is not carried on port 5060 but by RTP which uses a UDP connection on a port within the limits defined in /etc/asterisk/rtp.conf negotiated by the SIP protocol which by default uses UDP 5060 but just for signaling.


Thanks for the quick info. That makes sense. So looking back at my Networking Essesntials courses, I can’t remember or find which port RTP uses? So, where do I find which ports I need to open for RTP. DOes Asterix using a specific port range for RTP?


I think I answered every question you just asked in my reply. Did you actually read it?

Oh, Ok. I thought you have to specify the port numbers for RTP within your conf file. If the ports are already specified, than I’ll just look there and report back. So, I forgot to mention that when calling internally, I get two way voice just fine.

You don’t need to report back you will find by default it will be between 10000 and 20000, probably larger than you need, adjust it to suit. Of course it works on your LAN, there is no PNAT going on.

Thanks for the help, man. Appreciate it.