Audio cuts out on outbound calls as of first half of first ring

Audio issues with outbound calls only. There is “almost” no audio. When I make outbound calls, I hear the first part of the first ring, then that cuts out and there is nothing but silence, the call completes but there’s silence on both ends. Inbound and intra PBX calls work great.

Being very new at this, I’m not sure what further info to provide about my system except that the biggest issue seems to have been NAT so far. I have added the following three lines to /etc/asterisk/sip_nat.conf:


Also, all extensions are on the same subnet as the FreePBX server.

In one of the topics in this forum, I found the following that was said to someone with a similar (but not identical) problem than mine but that discussion is closed and I do not understand it - maybe this is relevant to what’s going on here?

“Ok, sounds like a router problem to me. You may need to open RTP ports to your carrier. Default is 10000-2000. You can edit in /etc/asterisk/rtp.conf You need 2 per concurrent call leg and 20% headroom.”

rtp.conf does not have any settings that I can change - there’s a terse warning about making any changes to this file at the top. Also, aren’t RTP ports opened only on incoming as all outgoing ports are open by default?

Thank you.