No voice when calling on FreePBX?

Question

Asterisk

There is a local network No. 1: 192.168.0.0/24
It has an IP PBX on FreePBX and ip phones. Phones ring among themselves, everything works fine.
There is a local network No. 2: 192.168.4.0/24
It has ip phones that are connected to FreePBX - Lan No. 1 and Lan No. 2 are interconnected by a provider VPN channel.
FreePBX is configured extremely minimally - installed, phones and trunk added.
The problem is this: Phones with Lan#2 dial up phones in Lan#1, but no sound is transmitted. The reverse is the same. What could be the problem, where to dig?
I read the Internet, with a similar problem they often write about NAT, but it is not used in this scheme, there is no address translation. They thought maybe the provider does not let some ports through, they called - they let them through. How to be?

Answer 1

in the main sip settings

localnet 192.168.0.0/24
localnet 192.168.4.0/24
localnet какие_ещё_сети_без_NAT
directmedia=no

Answer 2

In such cases, you first need to look where it goes.
ngrep and tcpdump will help you.
You can start with set sip debug on in the Asterisk console.