- I have a successful FreePBX 13 installation on Debian as per the steps in FreePBX wiki, I configured 2 extensions, where both can connect and place a call but can not hear each other, also if one party end the call then the call is not ended on the other party.
- I want the server and the clients to use TLS and SRTP only using non standard TLS and RTP ports, also I created FreePBX self-signed-certificate and applied in FreePBX settings.
- My FreePBX PC connect to a wireless router (LAN 192.168.1.xx) that is connected to internet (WAN) via static IP (so I believe I am behind NAT)
- I went through FreePBX SIP audio issues wiki with no luck to resolve it
- Here are all of the configuration I have edited where I left the rest to their default values:
— SIP nat = yes
— SIP encryption = yes
Asterisk SIP Settings >> General SIP Settings
— External Address = (My Public IP)
— Local Networks = 192.168.1.0 / 255.255.255.0
— RTP Port Ranges = start/end = custom range but not (10000-20000)
Asterisk SIP Settings >> Chan SIP Settings
— NAT = Yes
— Enable TLS = yes
— Certificate Manager = default (self-signed generated by FreePBX certificate management)
— SSL Method = sslv2
Advanced General Settings
— Bind port = custom port but not 5060
— TLS bind port = custom port but not 5061
Extensions >> Advanced
— NAT Mode = yes (force_rport,comedia)
— Port = custom port which is the same as default TLS port
— Transport = TLS only
— Enable Encryption = yes (SRTP only)
I have forwarded all the above custom ports from my router to the FreePBX PC.
I allowed all traffic in debian firewall.
As I used custom RTP ports, I have un-comment the line “include rtp_additional.conf” from /etc/asterisk/rtp.conf and I assured that rtp_additional.conf contains my custom RTP start/end ports.
I restarted asterisk after every change.
I configured 2 android phones using Zoiper to connect to FreePBX externally over 3g/4g network
Please let me know if there are extra settings to do.