IP PBX software
Resonance Telcom SIP trunks for FreePBX
Add Resonance Telcom as a SIP trunk in FreePBX and keep managing your phone system from the web interface you already know.
About FreePBX
FreePBX is an open-source web interface for building and managing phone systems on Asterisk. It is maintained by Sangoma and widely deployed on dedicated hardware and virtual servers.
FreePBX handles outside lines through SIP trunks configured in its trunk module, so Resonance Telcom fits in alongside your existing extensions, ring groups and IVRs.
Why Resonance Telcom for FreePBX
- Configure Resonance Telcom as a standard PJSIP trunk using the settings below.
- Route inbound numbers to the extensions, ring groups and IVRs you have already built.
- Engineers familiar with Asterisk and FreePBX trunking, NAT and codec settings.
Connection settings for FreePBX
Use these values when you add Resonance Telcom as a SIP trunk. Your account-specific credentials and number assignments are provided when your account is activated.
| SIP server / registrar | sip.resonancetelcom.example |
|---|---|
| Port | 5060 (UDP/TCP), 5061 (TLS) |
| Transport | UDP, TCP or TLS |
| Codecs | G.711u, G.711a, G.729 |
| Authentication | SIP registration or IP authentication |
How to configure FreePBX
FreePBX setup guide in progress
Step-by-step instructions for FreePBX are being finalized. Contact us and an engineer will walk you through setup.
FreePBX FAQs
Do I configure the trunk as chan_pjsip?
Yes. Current FreePBX versions use PJSIP by default, and that is the driver we recommend for new trunks.
Does FreePBX need a public IP?
Not necessarily. A registration-based trunk can work behind NAT if the external address and local networks are set correctly in FreePBX. We can help you check them.
Ready to move your phones to SIP?
Tell us about your phone system and how many calls you handle. We will put together a plan that fits.