Hi All ,
We have a freepbx box with SIP trunk terminated. When calling to the DID number, from the logs we can see that the call is landed as per the Incoming route and directed to a SIP extension. The problem is there is NO AUDIO. We cannot hear any IVR or the reponse from the SIP extension. But calling between SIP protocol (Internal) there is clear voice. What could be the problem ??
Asterisk Version 184.108.40.206
Freepbx Version 220.127.116.11
Thanks in Advance.
Most likely your firewall. Do you have port 6050 and RTP 10,000 - 20,000 forwarded from your router to the FreePBX box. Is your external IP setup in FreePBX?
Port 5060 is visible and RTP is also available. There is another Box running on same subnet with same firewall rules. But having a E1 line connected using Digium Card.
Have you configured your externip and localnet in sip settings?
You should search the topic of NAT and Asterisk/FreePBX, no audio is one of the most popular threads.
Also, do you think it is wise to have 10,000 RTP ports open to the world. Perhaps taking a look at the documentation and adjusting /etc/asterisk/rtp.conf would be a good idea.
I also suggest you use the permit/deny in the extensions or an access list in your router to only except traffic from your users.
If they don’t have fixed IP’s and you have to open box to the world, block countries that your users are not in and make sure you have hardened your box.
My Box is not under any NAT. I hope sip_nat.conf file is not relevant in this case.
RTP ports are open to the world. This System is located in the public. What are the possible reasons ??