One way audio on incoming calls - Local Network w/ double NAT

I setup freepbx 12 on a physical machine.

it is placed on a local network which is double NATed.

local calls to extensions work fine and outbound calls work fine.

Inbound calls have 1 way audio.

I already tried it with DMZ and set NAT rules to point to the device.

It is on a Zyxel USG40 behind a Comcast SMC Busines Gateway

the PBX machine is on the same local network as the phones.
when i setup a remote phone to test i got no audio at all and the call disconnected after 6 seconds.
(but i dont need this working any way)

any ideas?

Is the trunk configured to use NAT ie nat=yes? In the Asterisk Sip settings you should have the local and external IP setup there also. In some cases your VOIP trunk provider may also give you and option for NAT translation.

that’s about all i can say without knowing what your configuration is.

You shouldn’t need to use DMZ. that is generally a bad idea anyway. the UDP ports may need forwarding (10000-20000 as a rule), but you want to shield your PBX from the outside world as much as possible.

well i would have just forwarded 10000-20000 but the SMC gateway said sorry, you can only forward 5000 ports total. :frowning:

I am now switching to dedicated IP which is the only way they said i can get rid of the double NAT and bridge the Comcast SMC box. hopefully this will solve my problem.

I am using flowroute as my SIP trunk and i don’t see anything about NAT in their PEER Details.

They may not have Proper nat support thier end, i’ve never used flowroute (i’m in the UK),

bot some others like sipgate who had no nat settings, still worked when i put nat=yes into the peer configuration details.

I hope you get it sorted :slight_smile:

Also… you dont need to foward them all. You could set your port range in asterisk to just 2000 or so ports and then forwarded them. Each call normally takes up about 4 ports, so unless you’re planning on having it really really busy 2k should be fine.

asterisk sip settings -> external and internal ip’s need to be correct (and NAT)
RTP port range 10000 - 12000 should do you fine.

then in peer add

see if that improves things

just found this:

any help?

Also… i noticed they use G729 audio… without a licence that can hurt… in each Extension and sip trunk try adding this:


that will force the 64Kbit codec that needs no licence on all calls.
You can also disable certain codecs from asterisk Sip settings.

@miribis Thanks for your help!.

I called up flowroute (that’s why i use them - they provide phone support unlike cheaper trunks.) they said they do support that nat=yes in peer so i added that.

They also noticed i was registering to them with a local IP 192.168…

I went to the Settings>Asterisk Sip Settings and added External Address and Local Networks using the detect button.
Flowroute then confirmed i was registering correctly with external IP and calls now have 2 way audio.

no dedicated IP necessary.


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Glad to have helped :slight_smile: