Outgoing call audibility problem

I use FreePBX and Asterisk 15.7.4
My server is in a LAN with NAT and correctly works on the LAN
I need to connect one my user from home. The user is behind the NAT.
I’ve opened port 5060 and 10000-20000 for the user white IP on the firewall.
A SIP phone is correctly connecting, but I cannot be heard during an outgoing call.
When I connect the user from remote network without NAТ Incoming and outgoing calls work correctly.
On logs i see only:

[2020-03-26 12:28:27] WARNING[13857][C-000008a0]: translate.c:405 framein: no samples for ulawtolin

Where can be the problem?

Hi, The above one is the warning message and not the error.
The PCAP trace will help here to find out the problem.

Odds are high that the SIP phone may not be using these ports.

Linphone uses 7200-7299 by default.
Zoiper uses 32000-65535

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