Voip no one way voice, how to set up sip?

Question

Asterisk

Hello everyone, please help with solving the problem, there is an organization network, the network has asterisk (freepbx) and a branch network, all this is combined into l2tp. There are also yealink sip phones that work in the main office network, and in the branch network, these phones are registered on the asterisk server, the call goes on, but the voice is not heard. Voices are not heard only from the one who calls from the branch

Answer 1

As a rule, such a problem is either a crookedly configured routing, or there is nat somewhere on the route.
As a result, SIP traffic runs normally, and RTP is lost.

Answer 2

The simplest crutch is to run RTP through an asterisk - set directmedia=no

Answer 3

Have you tried looking at port forwarding? Check filtering? If a voice is not heard in the branch, then packets on ports 10XXX do not run there.
It's also worth googling the relationship between SIP and encryption. Although it works in one direction.
Without configs, local telepaths will not see anything. I always forget to post them myself.

Answer 4

I thank everyone for their participation in solving my issue, the solution was found, and as always it was the simplest solution. It was only necessary to add the subnet of the branch to the list of trusted networks (like the localnet section) and that's it, after that everything worked