Asterisk 13.15.1: NAT enabled, SIP ports opened in firewall.
- calls can be established but no audio on both sides
- calls auto drop after 30 seconds
- firewall stopped RTP packets as RTP ports not opened: audio transmission needs UDP ports (10000-20000) opened;
- call will be cut if no RTP packets received in 30 seconds(this timeout can be customized in Chan SIP settings).
I Googled a lot and didn't get any clear guides or posts providing a solution. I solved it by adding a firewall rule when I found asterisk calls are over RTP ports(range is 10000-20000).
#firewall-cmd --zone=public --add-port=10000-20000/udp --permanent