We have FreePBX 15.0.16.42 (Current Asterisk Version: 16.6.2).
Extension numbers are configured as SIP.
When trying to make an internal call, there is no voice. The call ends after 31 seconds and we see in the logs
chan_sip.c: 29861 check_rtp_timeout: Disconnecting call for lack of RTP activity in 31 seconds
Phones sets are located in a separate VLAN (172.21.104.0/24), the FreePBX server is located in another VLAN (172.21.100.0/24) via ipsec connection.
We tried to enable/disable NAT. The result does not changes.
The identical configuration in the network topology without ipsec works fine.
Where to look for the answer? Thanks!