I have installed FreePBX 15 and Asterisk 16 on one of my virtual servers running CentOS 7 64-bit at Linode. This server has a public static IP; therefore I can access both FreePBX and Asterisk on that IP with no problem. I have 2 extensions for testing 112 and 113. I’ve configured them on softphones (3CX Phone on PC and GS Wave on Android) and they register successfully.
The problem is that, when either extension calls the other, there is no audio and the call is dropped on the 30th second with the following message:
NOTICE: chan_sip.c:29981 check_rtp_timeout: Disconnecting call 'SIP/100-00000001' for lack of RTP activity in 31 seconds
Since last week, I have been attempting different configuration suggestions found online but nothing worked. Tried enabling and disabling NAT with no luck.
What is the best way to configure FreePBX 15 and Asterisk 16 so that it works when the box is connected directly to the Internet?
I’m posting here because I have tried everything, now I just don’t know what else to do. The same installation and configuration works perfectly on the server in the office. The box in the office connects to a 4G router, to which my PC and mobile also connect. This is the same router through which all these devices connect to the Internet. When I change the IP of the SIP Server, extensions are registered successfull, but calls only last 30 seconds with no audio both ways.
I have tried allowing all necessary ports through firewalld and even disabling the freiwall altogether with no luck. Selinux and IPTables are also disabled.
I look forward to your assistance. Thanks.
As for the current configuration:
sip_general_additional contains the following information (IP is not the real one):
All extensions contains the following:
canreinvite=yes session-timers=accept nat=force_rport,comedia