IP PBX software
Resonance Telcom SIP trunks for Asterisk
Point your Asterisk server at Resonance Telcom SIP trunks and keep the dialplan you have already built.
About Asterisk
Asterisk is an open-source telephony framework sponsored by Sangoma. It is the engine behind many commercial and do-it-yourself phone systems, and it can be configured through text files or a graphical front end.
Asterisk speaks SIP natively, so connecting it to Resonance Telcom is a matter of defining a trunk endpoint and routing calls to it in your dialplan.
Why Resonance Telcom for Asterisk
- Standard SIP trunk settings that drop into your existing PJSIP configuration.
- Your dialplan, IVRs and queues stay exactly as they are.
- Engineers who can read an Asterisk log and help you troubleshoot quickly.
Connection settings for Asterisk
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 Asterisk
Asterisk setup guide in progress
Step-by-step instructions for Asterisk are being finalized. Contact us and an engineer will walk you through setup.
Asterisk FAQs
Which SIP channel driver should I use?
Use chan_pjsip. The older chan_sip driver is deprecated and has been removed from recent Asterisk releases.
Can I use registration or IP authentication?
Both approaches can work with Asterisk. Which one your trunk uses is confirmed when your account is set up.
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.