I’m setting up a new install.
Using FreePBX 18.104.22.168
And I’m in the trial of SIP STATION (first time I use it).
I went though the setup and everything is fine except that when I receive a call, the other end doesn’t hear my voice.
This is only occurring on inbound calls.
Outbound calls are fine!
My firewall is set to forward the following ports (UDP) to FreePBX : 5060, 5061, 5160, 5161 & 10000 to 20000.
Any idea what could be causing the outgoing voice to not be sent in inbound calls?
Have you confirmed the ports are actually open? Specifically the 10000-20000 ports?? SIP ALG turned off???
Yes the ports are open!
What is SIP ALG?
I’m using pfSense as a firewall / gateway
In Asterisk SIP Settings, confirm that External Address and Local Networks are correctly set.
If you change these, after Submit and Apply Config you must restart Asterisk.
I dont use pfsense but after a quick search it doesnt appear you have to worry about SIP ALG on pfsense…
Yes external address and local networks addresses are correctly set!
in pfSense, I followed these instructions:
Still won’t work.
It states that “With a minority of providers, rewriting the source port of RTP can cause one way audio.”
I did set my RTP ports to STATIC.
I don’t understand why it still does not work
Definitely sounds like a pfsense/network configuration issue…
Usually making RTP ports static works on pfSense. I would look at a pcap (you can use
sngrep on the PBX’s CLI) to see what’s the address you are sending out as your external address, also the contact information.