Start with your UKDDI activation email, choose registration or direct-IP delivery, then follow the guide for your PBX, SIP proxy or handset.
These are the shared settings for a standard registered connection. Use the values in your activation email if they differ, and use the separate direct-IP guide for an IP-authenticated trunk.
Open the complete client SIP reference →Credentials: supplied securely for the individual service. The DDI and the authentication username are not necessarily the same value.
Use these pages only when UKDDI has issued a direct handset account. Phones behind a PBX normally register to the PBX instead.
On a compatible Yealink SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Cisco SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Grandstream SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Poly SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Snom SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Fanvil SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Gigaset SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Panasonic SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Mitel SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible AudioCodes SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Spectralink SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Akuvox SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Htek SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Yealink SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Cisco SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Grandstream SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Poly SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Snom SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Fanvil SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Gigaset SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Panasonic SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Mitel SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible AudioCodes SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Spectralink SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Akuvox SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →On a compatible Htek SIP device, create one line with sip.39networks.net on port 5061 over TLS, enter the customer-specific identity and password, apply SRTP and confirm a stable registration before testing calls.
Open guide →Seven guides cover DrayTek, BT Business Hubs, FortiGate, pfSense, Cisco, Ubiquiti EdgeRouter and MikroTik while preserving normal access controls.
DrayTek documents SIP ALG as disabled by default on current supported configurations. First prove whether it is active; where the model exposes NAT > ALG, leave SIP ALG disabled, then clear only the affected sessions or restart the test endpoint and retest.
Open guide →On a supported BT Business Hub, open 192.168.1.254 or businesshub.home, choose Advanced Settings, authenticate, open Firewall > Configuration and set Disable SIP ALG to Yes under Application Layer Gateway, then Apply and retest.
Open guide →FortiGate can process SIP through the SIP ALG or SIP session helper, and the exact disable procedure varies by FortiOS version and inspection design. Back up first, identify which mechanism owns the session, follow the matching Fortinet release instructions and clear only affected sessions before retesting.
Open guide →A standard pfSense installation does not provide a conventional SIP ALG toggle. Check that an added SIP proxy package such as siproxd is not intercepting traffic, then diagnose NAT state; if a phone or PBX genuinely needs source-port preservation, use a narrow Hybrid Outbound NAT static-port rule rather than a network-wide change.
Open guide →On Cisco ASA, SIP ALG behaviour is provided by SIP inspection in a service policy; find the applied policy and class, remove only inspect sip, save and establish fresh sessions. Cisco Secure Firewall Threat Defense is managed differently, so use the matching FMC or Device Manager procedure rather than pasting ASA CLI.
Open guide →On an EdgeRouter running EdgeOS, disable the SIP conntrack module, review the pending change, commit and save. This guide is for EdgeRouter, not UniFi gateways or Dream Machines.
Open guide →MikroTik exposes its SIP NAT helper under IP > Firewall > Service Ports. Disable only the sip entry, verify disabled=yes and test fresh SIP sessions while keeping connection tracking and normal firewall protections enabled.
Open guide →Five practical articles explain quality, bandwidth and interoperability, including which codec to prefer for a UK business trunk.
For most UK PSTN-facing SIP services, prefer G.711 A-law when it is enabled on the UKDDI account because it is widely interoperable and avoids unnecessary low-bitrate transcoding. Use G.722 for end-to-end HD voice and Opus for compatible WebRTC/softphone paths; keep G.729 for constrained links only when every required leg supports it.
Open guide →G.711 A-law encodes narrowband voice at 64 kbit/s and is commonly aligned with UK and European PSTN interworking. When UKDDI enables it and the network has capacity, it is normally the first codec to test because it avoids the extra quality loss of low-bitrate compression.
Open guide →G.722 carries wideband speech with audio bandwidth up to 7 kHz at a 64 kbit/s encoded rate. It can make compatible internal or IP-to-IP calls clearer, but a narrowband PSTN leg or incompatible endpoint will constrain or transcode the call.
Open guide →G.729 compresses speech to an 8 kbit/s payload, but RTP/IP overhead remains and compression can reduce fidelity. Use it only when UKDDI and every necessary call leg support it and a measured bandwidth constraint outweighs transcoding and quality costs.
Open guide →Opus is a flexible interactive speech and audio codec with a wide operating range defined by RFC 6716. It is an excellent fit for compatible WebRTC and softphone paths, but a UK PSTN leg commonly needs G.711 or transcoding, so use Opus only where the full route and UKDDI profile support it.
Open guide →Complete the checks in order so registration, inbound routing and media are tested separately.
Check the registrar, transport, port and account-specific credentials.
Add and match the complete E.164 number, including the leading +44.
Place inbound and outbound calls and confirm the approved caller ID.
Check two-way speech, DTMF, hold, transfer and a call longer than five minutes.
UKDDI support can confirm the settings issued for your number without asking you to disclose the SIP password.