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 192.168.30.252 nethserver 192.168.30.19
would i run the above command on the nethserver 192.168.30.19? 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 192.168.33.0/24 via 192.168.30.1 dev br0?
i’m about to make the change, quick what is the command to back it out if any problems?
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.
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.