Asterisk NAT voice?

Question

Asterisk

Colleagues asteriskovody, help me out)
Given:
Server with a public address (NOT behind NAT), on board asterisk 13.6.0.
Use ONLY pjsip as the channel driver.
A bed of clients sitting behind NAT.
We have two trunks: A and B.
When calling through trunk A, all clients always receive a voice from a remote subscriber, in short, everything works.
With incoming calls through trunk B, again, everything works.
For outgoingcalls through Trunk B, 100% of clients do not receive a remote voice. On the asterisk, the conversation is written, the voice reaches it normally. On the router, behind which the clients are sitting, I monitor the connection and see that voice packets are coming, traffic is going in both directions. However, clients receive silence. In the SIP debug, the whole process from INVITE to ACK after the voice session is established, the ports to which asterisk sends the voice match the ports of the device behind nat, I see this on the router. But NO voice!
There are no messages about codec mismatch, for example, or any other problems in the asterisk console.
I would be grateful for any ideas / tips, I've been fighting for a very long time, googling does not give anything worthwhile.

Answer 1

The problem in the end was the incompatibility of the codecs of my PBX and the provider's equipment. I don’t know how it is, but everything was decided after contacting the support.

Answer 2

I had a somewhat similar question - No voice when calling on FreePBX?
Read it, it will help you too.