When setting up SIP trunks with 3CX, most common issues fall into a few predictable categories: registration failures, one-way or no audio, call drops, and problems with outbound or inbound routing. Troubleshooting effectively begins with a structured approach. First, confirm that your SIP trunk details exactly match those provided by your service provider, including SIP server address, authentication ID, password, and any required outbound proxy. A single typo or incorrect transport type (UDP/TCP/TLS) can prevent the trunk from registering. Next, verify that your firewall and router are correctly configured: SIP and RTP ports must be open, forwarded to the 3CX system where appropriate, and any SIP ALG features on routers or firewalls should usually be disabled, as they often interfere with VoIP signalling.
Once registration and connectivity are confirmed, attention should turn to call quality and routing. Audio issues typically relate to codec mismatches, NAT configuration, or blocked RTP ports. Ensuring that the correct codecs (such as G.711) are enabled and that 3CX is aware of its public IP address will often resolve one-way audio or no audio problems. For routing issues, carefully review inbound and outbound rules to ensure numbers are formatted as your provider expects, and that calls are being directed to the correct extensions, ring groups, or queues. Using the 3CX activity log and built-in firewall checker provides detailed diagnostics, allowing you to pinpoint where calls are failing. By following these steps methodically, most SIP trunk issues with 3CX can be identified and resolved quickly, restoring reliable voice services for your organisation.
Begin by checking SIP trunk registration status in the 3CX management console. If the trunk shows as “Not Registered” or “Rejected”, re-enter the SIP credentials, confirm the SIP server/host name, and ensure the correct port and transport are selected. Run the 3CX firewall checker to verify that required ports are open and that no misconfiguration is blocking SIP or RTP traffic.
For audio-related issues, confirm that 3CX is using the correct public IP or STUN settings, especially if it is behind a NAT router. Disable SIP ALG on the router, as it often causes one-way audio. Check that appropriate codecs are enabled and prioritised, and that the RTP port range configured in 3CX is allowed through the firewall.
If calls fail in one direction only, focus on dial plans and number formatting. Make sure outbound rules match the dialled numbers and prepend or strip digits as required. For inbound issues, verify DID numbers, inbound rules, and that the SIP provider is sending numbers in the expected format.
