EN admin_guide troubleshooting_no_audio

Troubleshooting calls with no audio

When the call connects but no sound/audio passes, in most cases the issue is related to RTP/NAT/firewall configuration, not the billing itself.

Please check the following:

1. Open the RTP ports on the server firewall MagnusBilling by default has these ports open, but if you have a custom firewall configuration, please make sure the following ports are open:

10000:60000/udp

Check if they are open in your firewall, cloud provider firewall, router, or security group.

2. Check Asterisk NAT settings

Edit:

/etc/asterisk/sip.conf

Confirm that the server has the correct public IP configured, for example:

• externip=YOUR_SERVER_PUBLIC_IP
• localnet=YOUR_LOCAL_NETWORK/NETMASK
• nat=force_rport,comedia

Then reload Asterisk:

asterisk -rx "sip reload"

3. Check if RTP packets are arriving

Run this command during a test call:

tcpdump -n udp portrange 10000-60000

If you do not see RTP packets, the audio is being blocked before reaching the server.

4. Disable direct media if needed

If direct media is enabled, audio may try to pass directly between caller and provider, causing one-way audio or no audio depending on NAT.

In MagnusBilling / Asterisk options, test with directmedia disabled.

Documentation: https://wiki.magnusbilling.org/en/source/get_started/first_call.html#directmedia

5. Test with another provider or SIP account

This helps confirm if the problem is on:

• the customer device/softphone,
• the provider trunk,
• firewall/NAT,
• or server configuration.

Useful documentation: https://wiki.magnusbilling.org/en/source/get_started/first_call.html https://wiki.magnusbilling.org/en/source/get_started/quick_install.html

If after checking this the issue continues, please send us:

• one example call date/time,
• caller number,
• destination number,
• provider used,
• and whether the issue is no audio both ways or only one-way audio.

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