Concurrent-call channels, from your PBX to the PSTN.
A SIP trunk is a pool of concurrent-call channels that carries voice between your PBX — on-prem Asterisk or FreePBX, or a hosted platform — and the carrier network that reaches the PSTN. We size the trunk to your real call volume, provision and route your DIDs, and stand up a redundant failover path so a single carrier or circuit issue doesn’t drop your phones.
The trunk itself is delivered through partner carriers — we’re able to quote and integrate with the platforms we’re partnered with — but the engineering around it is ours: least-cost routing, E911, codec selection, and toll-fraud controls. One team owns the trunk, so when a call fails there’s someone who reads the SIP trace.
The engineering around the trunk.
Channel & trunk sizing
We size concurrent-call channels to your real peak — not a guess — so you're not paying for idle capacity or blocking calls at busy hour.
DID management
We provision, group, and route your direct-inward-dial numbers — mains, departments, and DID blocks mapped to the right endpoints.
Redundancy & failover
A standby trunk and failover routing so a carrier or circuit blip reroutes calls instead of dropping them — tested, not assumed.
Least-cost routing
Outbound routing rules that pick the sensible path per destination, with the carrier relationship handled on your behalf.
E911
Emergency calling configured with the right dispatchable location, so a 911 call lands where it should.
Fraud controls
Toll-fraud guardrails — international dialing limits, rate anomalies, and registration hardening — to keep a compromised extension from running up a bill.
Codec selection
Codec choices matched to your bandwidth and quality needs, so voice stays clean across the trunk without wasting transport.