SIP Trunks¶
A SIP trunk connects your NexusUC instance to the public telephone network so you can make and receive external calls.
Default setup: Localcom SIP trunks (Australia)¶
New NexusUC instances are provisioned with Localcom SIP trunks and Australian outbound routes added automatically. For most Australian deployments no trunk configuration is needed — order numbers (DIDs) through the Partner Portal, then route them under inbound destinations (see Call Routing).
Localcom trunks are the recommended option for Australian deployments: they are pre-tested with the platform, support fax (T.38/G.711), and are covered by Localcom support.
Connecting a third-party SIP trunk¶
For deployments outside Australia, or where you have an existing carrier relationship, NexusUC can connect to any standards-compliant SIP trunk provider.
1. Gather these details from your provider¶
| Setting | Typical values / notes |
|---|---|
| SIP server / registrar address | Hostname or IP, and port (usually 5060) |
| Authentication method | Registration (username + password) or IP authentication |
| Username / Auth ID and password | Registration-based trunks only |
| Your DID numbers | In E.164 format where possible |
| Supported codecs | G.711a/G.711u are safest; confirm G.722 if you want HD audio |
| DTMF method | RFC 2833 is the most widely compatible |
| Fax support | T.38 or G.711 passthrough required for faxing |
2. Create the trunk (gateway) on the PBX¶
Add the provider as a gateway/trunk in the PBX admin portal using the details above.
3. Create outbound routes¶
Add outbound routes that match your country's dialling patterns (national, mobile, international, emergency) and point them at the new gateway. Include the correct emergency number for your country (e.g. 000 in Australia, 911 in North America, 112 in the EU) and test it with your provider's guidance — emergency calling obligations vary by country.
4. Route your DIDs¶
Point each DID at the appropriate destination (extension, ring group, IVR, or time condition). See Call Routing.
5. Test before going live¶
- Outbound call to a mobile and a landline (check audio both ways and caller ID)
- Inbound call to each DID
- Long-duration call (15+ minutes, checks session timers)
- Fax send/receive if used (see Fax & Contacts)
- If you get one-way or no audio, see Troubleshooting — NAT and SIP ALG are the usual causes.
Provider-specific guides¶
Step-by-step guides for popular providers will be added here as they are validated. If you would like a specific provider documented, request it through the Partner Portal.