SIP to SIP call's when both phones are outside the firewall

I’m currently running Trixbox 2.8.0 and I have the following issue.

  1. I have two soft phones outside my firewall. They can successfully register to the PBX. I can even make external call’s and call’s to phones registered inside the firewall and vice verse. The problem appears when I try to make a phone call’s between the two soft phones outside the firewall, there is dial tone and ring tone but no audio after I pick up the phone.

Does anyone has clue on how to solve this?


Do you have NAT=YES for your extensions and does your firewall pass port 5060-5080 and 10000-20000?

I got NAT set to yes on both extension’s and the mentioned port’s forwarded to my Trixbox machine.

As I mentioned before, there is no issue making call’s from a client located outside the firewall to a client inside the firewall and vice verse. The problem appears when I try to make a call from a phone located outside the firewall to another phone outside the firewall. Both phones are local phones registered to the PBX.

Since this is a test/evaluation the WAN port of my fire wall has a 192.168.. address. and the phones on the WAN side of the firewall also has 192.168.. adsresses.


Do you have “can reinvite” set to “no” for the stations outside the firewall? Again, check to insure your router is passing the correct ports both directions.

There issue has nothing to do with FreePBX or Asterisk. You’ll find the trouble in the setting of the SIP clients, the router, firewall or extension setup.

Device option

dtmfmode: rfc2833
canreinvite: no
context: from-internal
host: dynamic
type: friend
nat: yes
port: 5060
qualify: yes

I can give out the monowall setup if u want.


It was a firewall issue.

The source port range was also set to 5060-5082 same as destination port range.

I removed the source port range and set it to *. Then it worked.

Thanx for the input.