Inbound Call - We hear caller, caller can't hear us

Ive gone over the forum for hours trying to figure this out, with tons of google searches. All seem to point me to a NAT / Firewall issue. But my settings their have not changed that I know of.

Everything was working a few days ago, then today it started on all inbound routes except for the primary. Now all the inbound routes have the same issue. When we receive an inbound call, we hear the caller and they can’t hear us. There is no issue with outbound, audio works both ways.

NAT is enabled, external and internal IP address our correct in the Asterisk Sip Settings.

Nothing has changed on the fire wall, SIP ALG disabled.

Have nat translation and port forwarding setup for 10000-20000.

Have reboot the freepbx / asterisk server, no change.

Any help would be greatly appreciated I am running FreePBX version FreePBX 13.0.192.19

If you need any other info from me please let me know.

Thank you

Nathaniel

Update, after setting all the inbound routes to ring groups, all work but one. It still has the same issue, no ringing from the caller inbound to freepbx, and once we answer the call we can hear the caller they can’t hear us. But 2 way audio on the other inbound routes, also they all run off the same Flowroute trunk.

Make sure have possible Flowroute servers as inbound trunks and none are being blocked

https://support.flowroute.com/SIP_Trunking_and_Voice/Networking_Guides/Set_Firewall_Policies_for_Flowroute’s_SIP_Signaling_and_RTP_Media

When you say as inbound trucks am I setting these settings with those IP addresses you sent in the link here, creating a new trunk for each? This is under inbound of the sip settings for the trunk?

Now I have the issue back of inbound calls dropping after 30 seconds. This has been coming back randomly and usually a reboot fixes it.

This seems like there’s something wrong with the RTP settings.
Make sure you enable NAT and have 10000-20000 fwded correctly to the PBX.

Also, do you have any “SIP Helper” on your router?

Check the /var/log/asterisk/full log and you’ll probably see that inbound calls drop after 30 seconds because of RTP failures.

One part of that is that your register string points to a network and not a specific server. I doubt that you are using 16 addresses on this server for your VOIP system. Set that register string so that it works with your specific server and you might see some relief.

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