FreePBX 16– No Audio 2 way and Call Pickup Issues

FreePBX Networking
1

Hi All,

I’m currently working on a POC (Proof of Concept) setup for a VoIP solution using FreePBX and Asterisk:

  • FreePBX Version: 16
  • Asterisk Version: 18

I’ve purchased 2 DID numbers and a 2-channel SIP trunk from a provider in Singapore for testing.

Setup Details:

  • Asterisk/FreePBX is running on an internal network.
  • Port forwarding is configured for UDP 5060 and UDP 10000–20000 to the FreePBX server.
  • NAT settings are configured with the public IP and internal IP under SIP settings.

I’m new to FreePBX but very interested in VoIP solutions and learning more.

So far:

  • Trunk is configured for both inbound and outbound.
  • Inbound and outbound routes are set up.
  • Extension 200 is registered and connected to the FreePBX server via a WAN IP.

Issues:

  • Outbound calls to mobile numbers connect, but there is no audio in either direction.
  • Inbound calls ring extension 200, but I can’t pick up the call.

Goal:
To successfully make and receive calls from the extension with working audio in both directions.

Any guidance or suggestions on how to resolve these issues would be greatly appreciated!

Thank you!

2

Have you confirmed SIP ALG is turned of on your router?

You’d also need to provide logs/redacted pcaps

3

You can check what @airsay suggested, and you can also go to Settings > Asterisk SIP Settings > Detect Network Settings and check if the External Address is the correct one.

4

Hello, we are facing a very similar issue. We are using a SIP trunk from the Brazilian provider Oi UC4X, which requires SRTP and TLS transport via PJSIP.

Even after correctly setting the external_media_address, external_signaling_address, and media_address parameters, and ensuring NAT is properly configured on FreePBX/Asterisk, we are still experiencing no audio during calls.

When analyzing SIP sessions with sngrep, we observed that the c= field in the SDP (responsible for indicating the media IP) is showing the server’s local IP or the extension’s internal IP, instead of the server’s public IP, which is the actual address used for communication with the provider.

This behavior breaks RTP/SRTP media exchange, preventing proper audio transmission.

Important: When the SIP account is registered on MicroSIP, within the same network, everything works perfectly — with two-way audio. The issue only occurs when using FreePBX/Asterisk with TLS and SRTP.

We would like to know if anyone has faced this situation and if there is a reliable way to force Asterisk to use the public IP address in the c= field of the SDP when making external calls using TLS and SRTP.

5

Is the local network configured properly in the transport?

6

Correctly set the local networks and the public media address.

Or, configure the remote side to do so.

7

Yes, the network is properly configured. NAT is set to ‘yes’, I filled in the static IP field with the public IP and set the LAN network to ‘192.168.0.0/16’ as required in the Asterisk SIP settings. I also confirmed in the pjsip.transports.conf file that the external_media_address and external_signaling_address fields are correctly set with the public IP, and the local_net is set with the LAN subnet.

8

Although not going to be the problem here, nat=yes is a chan_sip thing and relates to cases where Asterisk is outside NAT relative to the endpoint, not to the case here.