Carrier-Grade Voice for Any PBX
Already have a PBX you love? Don't replace it — upgrade its connectivity. IntelliVoice SIP Trunking delivers enterprise-grade PSTN connectivity to your existing phone system, whether it's on-premise or in the cloud.
Our network spans three geo-redundant datacenters with automatic failover. If one path goes down, your calls route through another — seamlessly and instantly.
Works With Your Existing PBX
FreePBX, PBXact, and custom Asterisk builds
On-premise and cloud-hosted 3CX systems
CUCM, CME, and Webex Calling
Avaya, Mitel, Grandstream, and more
Technical Specifications
| Feature | Specification |
|---|---|
| Codecs Supported | G.711u, G.711a, G.729, Opus |
| Signaling | SIP over UDP/TCP/TLS |
| Media | RTP/SRTP |
| DTMF | RFC 2833, SIP INFO, In-band |
| Fax Support | T.38, G.711 passthrough |
| Max Concurrent Calls | Unlimited (per account) |
Why Choose Our SIP Trunks?
Multi-Carrier Network
We're not a single-carrier reseller. Our network aggregates multiple Tier-1 carriers for redundancy and quality.
Expert Support
Our team has configured thousands of SIP trunks. We'll help you get connected — even with tricky firewall setups.
How SIP Trunking Works
A SIP trunk replaces your old analog lines, T1s, or PRI circuits with a virtual connection that carries calls over the internet or a private data circuit. Your phone system stays exactly where it is — the trunk simply gives it a faster, cheaper, more resilient path to the outside world.
Point your existing PBX or SBC at our network using registration or IP authentication. No hardware to replace.
We move your local and toll-free numbers over at no cost and add new DIDs in any US area code.
Inbound and outbound calls travel as encrypted SIP/RTP across three geo-redundant POPs with automatic failover.
Add or remove channels instantly as your call volume changes — no truck rolls, no new circuits.
SIP Trunking vs. PRI vs. Hosted PBX
| SIP Trunking | Legacy PRI / Analog | Hosted PBX (FusionUC) | |
|---|---|---|---|
| Keep your existing PBX | ✓ Yes | ✓ Yes | ✗ Replaced by cloud PBX |
| Who manages the phone system | You | You | IntelliVoice |
| Add / remove capacity | Instant, per channel | Provision in blocks of 23 (T1) | Instant, per user |
| Typical monthly cost | $$ — lowest for existing PBX | $$$ — loop + line charges | $$ — per user, all-inclusive |
| Geo-redundant failover | ✓ Built in | ✗ Single circuit | ✓ Built in |
| Best for | Businesses with a PBX they want to keep | Legacy sites not yet migrated | Teams that want IntelliVoice to run everything |
Not sure which fits? If you love your current phone system, choose SIP trunking. If you'd rather hand the whole platform to us, see FusionUC Cloud PBX.
Backed by the Same Network as FusionUC
IntelliVoice SIP trunks ride the exact carrier-grade voice network that powers our FusionUC cloud PBX — the same Tier-1 carrier aggregation, the same three geo-redundant points of presence, the same 99.99% uptime SLA and US-based support.
That means you can start with trunks for the PBX you own today and migrate to a fully hosted FusionUC platform later — same numbers, same network, no second porting project. It also means you can pair trunks with branded caller ID and AI voice agents when you're ready.
SIP Trunking Questions
SIP trunking connects your existing phone system (PBX) to the public telephone network over the internet instead of physical phone lines. Calls travel as data, so you keep the PBX you already own and replace old analog lines, T1s, or PRIs with flexible, lower-cost digital channels.
With a hosted PBX like FusionUC, the phone system lives in our cloud and you just use the phones. With SIP trunking, you keep and manage your own PBX and we supply the dial tone and numbers. Choose trunks when you want to keep an existing Asterisk, 3CX, Cisco, Avaya, or Mitel system.
Yes. We port your existing local and toll-free numbers to our network at no cost, and can add new local DIDs in any US area code or toll-free numbers on demand. Your numbers stay the same for your customers.
Trunks are billed per concurrent channel (a flat monthly rate per simultaneous call) or metered per minute for lower-volume sites. Most businesses save 30–60% versus legacy PRI or analog lines — no per-line loop charges and no porting fees. Contact us for a quote based on your channel count.
Our standards-based trunks work with virtually any SIP-capable PBX — Asterisk/FreePBX/PBXact, 3CX, Cisco CUCM/CME and Webex Calling, Avaya, Mitel, Grandstream UCM, and SBCs. We help with registration or IP-authenticated trunks and firewall/NAT setup.
Yes — T.38 fax over IP and G.711 passthrough for analog fax machines behind an ATA, plus Enhanced 911 (E911) with a registered dispatchable location for compliance.