Just replaced an existing FreePBX 2.6/Asterisk 1.4 system with a new FreePBX (6.12.65-26 distro) running Asterisk 11.16.0
We had a number of remote phones (behind NAT) and they were all working fine until the new PBX was installed…
Now the remote phones are registered, however there is audio in only one direction.
The PBX is behind the same (unchanged) firewall (ports forwarded: 5060, 10000-20000). I have added the external IP and internal networks ( dual NIC’s, so both 192.168.2.0/25 and 192.168.115.0/24 added).
NAT is set to yes in the SIP configuration
NAT Mode is set to Yes - force_rport,comedia in the extension settings
In fact I have tried every Nat mode for the extension and they do not appear to make any difference.
Running rtp debug, I see the PBX sending the RTP to the wrong IP address (internal IP of remote phone) instead of the IP address of their firewall.
I have also checked and confirmed the settings in the various sip_* files and they match what FreePBX is telling me, but Asterisk just seems to ignore the NAT settings completely.
I’ve tested with a different brand phone (Yealink) and the issue still persists.
I have discovered though that the issue is intermittent and not constant at all. Make a call, no audio, hangup, make another call, audio… this with absolutely no changes between calls.
I have compared sip debug with a working and non-working audio call without seeing any differences.
Using rtp debug, the only difference I see on a call with audio is that there is a line with
Sent RTP packet to ext.ip.addr:51956 (type 00, seq 007971, ts 000160, len 000160)
> 0xb4e5a990 -- Probation passed - setting RTP source address to ext.ip.addr:51956
When there is no audio, there is no line like this and the rtp debug just starts with
Sent RTP packet to int.ip.addr:56818 (type 00, seq 042536, ts 000160, len 000160)
where int.ip.addr is the local internal address of the outside phone.
Who is your DID provider? The carrier in Canada (CanadaDIDs & Prodosec) are experiencing a long term 1-way audio issue on their end, but it happens fairly rarely, about 1 several hundred incoming calls.
Thanks for the response - it has nothing to do with the service provider - these are internal (extension to extension) or external calls.
We actually use basic POTS lines for service here.
Good morning all, i am sorry but i do not reply on this thread to give a correct answer it’s rather to get help because i am very interesting with what you are discussing here.
The following lines is are my case :
Asterisk server has a fix IP @ on the network : 192.168.1.110 so i use FreePBX to configure these different files , SIP / Extension … An employee is in vacation so he wants to use his extension even when home, in SIP configuration i made a bit modification and put NAT set to Yes for this feature, the public IP @ is configure in the router’s interface and on NAT Firewall these ports are forwarded 5060 ; 10002 - 10006 pointed to the IP @ address of the Asterisk server which is 192.168.1.110 but the SIP Remote phone does not work . I know i miss a lot of things !!! If someone has already experience this feature please help me…
One last question,does only the router that hosted the PBX server i need to configure for NAT firewall,what about to the other , the home router ?
Thanks !