hi all, im using astpp community version 6. at the sip profile set “inbound-bypass-media=true” to bypass media direct from telco provider to end user. the RTP manage to pass each other. soft phone call to mobile with answer call, mobile hang up call 1st but soft phone still on call. soft phone look like unable to receive bye event. do you all facing same issue?
My NIC is direct configure public IP. Just using debian os(same server astpp) firewall to control policy. My environment should be behind nat right?
If your NIC has a public IP address assigned to it, then it should not be NAT