Outgoing call audibility problem

Tags: #<Tag:0x00007fafc3afaf58>

(Perrfect) #1

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?

(P Ramarajan) #2

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

(Jared Busch) #3

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

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