No sound in asterisk

sachin

My Asterisk is set up on the local network and registered a SIP trunk. On very call, my media IP is going with the INVITE is local due to which I'm unable to get sound in my call

sachin

In your pjsip.conf, configure the transport with NATing.

[transport-udp]                           ;default
type=transport                            ;default
protocol=udp    ;udp,tcp,tls,ws,wss,flow  ;default
bind=0.0.0.0                              ;default

local_net=192.168.2.1/24                  ;NAT
external_media_address=119.21.17.152      ;NAT
external_signaling_address=119.21.17.152  ;NAT