Called party cannot hear me

[SHussainAkbar]

I am running Asterisk 16.9.0 on a CentOS server at a client’s location. That server connects via a SIP trunk provided by the telco.

I configured Zoiper on a couple of PC’s there and connecting using IAX works for inbound & outbound calls. For some reason, connecting via SIP was giving audio problems. Now, calls from ext to ext, ext to outside and incoming calls are working fine without issues.

Now I am trying to connect to this server from my PC at my office. No VPN; port 5060 has been forwarded from the client’s firewall.

From Zoiper, I connect using IAX, console shows the client registered. I can dial outside number and hear the called party fine. The called party hears a highly distorted version of my voice.

Using a Grandstream GXP1625, the console shows that a SIP client has connected. Dialing out, I can hear the called party, but he hears no sound and the call disconnects after 60 seconds.

What should I check / change?

[EkFudrek]

Nowadays you usually do not need to forward any ports. Many service providers simply reuse the connection that exists when you register, especially when using TCP. Your route only has to maintain the state of the connection sufficiently long. The standard recommended NAT settings have always worked for me so far.