Sonicwall pbx route issue

I have a site to site vpn set up between site b connecting to main site a

Site a hosts the pbx server on a lan 192.168.30.X gateway is

Pbx server has address of it has a separate 3rd party gateway of

I can’t ping the box server from site b I can ping all other devices on the lan at site a from site b but not the pbx server.

I assume this is due to the difference in gateway which I can’t change but how can I resolve this some sort of static route?

I don’t have a specific solution for you, but this thread might at least get you connected to the right sub-culture. Happiness With Sonicwalls - It can happen!

1 Like

That’s not a SonicWALL issue - that is a simple Static Route:

Add a route specifying the gateway that allows access to the remote subnet, and you are good.

1 Like

Thanks Greg

So how would that look in this case

So my users are at Site B on range 192.168.33.x

Site A is on 192.168.30.x with Gateway of PBX server is on routing out via gateway of

The two are connected by site to site vpn

What route would I add would it look something like this?

ip route add via dev eth0

Other way around:

ip route add via dev eth0

So you are telling the Asterisk how to get to the complete 192.168.33.x subnet via the router that allows access to that subnet.

ahhh sorry i’ve just realised i may have omitted som relevant information,

the pbx system is running in a vm on nethserver pbx vm ip nethserver
would i run the above command on the nethserver or on the Asterisk .252 or both

also its running as bridged connection so the adapter is br0 not eth - just edit to say ip route add via dev br0?

i’m about to make the change, quick what is the command to back it out if any problems?

From the article above:

Add a permanent static route

To make the route permament, you need to create a static route configuration file. Create a file with the name route-interface in /etc/sysconfig/network-scripts, such as:

nano /etc/sysconfig/network-scripts/route-eth0

Or whatever interface you are using - routing for Asterisk has to be done on Asterisk.

ok i have added the above route Greg, thank you i can now ping the server from site b.and the softphone is connecting

the issue i now have is when a call is made there is no audio on either end?

if i test using vpn from site b (open vpn not site to site) it works fine, but obviously i’m trying to use with site to site so we don’t have to use open vpn.

Under SIP Settings on the PBX, you have to add the remote Subnet as a local Subnet - otherwise, the Asterisk tries to do NAT translation - and you get no audio.

1 Like

Greg… Once again you have saved the day, a true hero of the forum, a million thank you’s to you sir.

No problem.

As you get more experience with the various issues with FreePBX (and really, Asterisk in general) and you see questions here that you can answer, go for it - Everyone is at some stage in the process of learning all the various things that go into getting an install working - help where you can and the whole community benefits.


I will do Greg, be great to give something back to the community.

This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.