Channels you control
Set the concurrent-call ceiling to suit your peak and change it between months. No engineer visit, no new physical line.
Overview
The same phone system, a cheaper and more flexible line into it
A SIP trunk is simply the path your calls take between your phone system and the public telephone network, carried over your internet connection instead of dedicated copper. Everything your staff see stays the same: the same handsets, the same extensions, the same numbers.
What changes is the economics and the flexibility. ISDN is sold in fixed blocks of channels with line rental attached to each one, and adding capacity means an engineer visit. A SIP trunk can be resized between months, and with Pay As You Go billing you are charged for the calls you make rather than for capacity sitting idle.
Most systems installed in the last fifteen years already speak SIP, and Asterisk is SIP-native. Where a PBX genuinely cannot, we will tell you before you spend anything — and moving that system into the cloud is usually cheaper than replacing it under time pressure.
Same-day activationon most accounts
Per-second billingno rounding to the minute
Fraud caps includedspend limits and alerting
What's included
What comes with a VOIPNOX trunk
Capacity, security and routing control, without a separate line for each one.
Per-second billing
Calls are billed by the second from a topped-up balance. A 20-second call costs 20 seconds, not a full minute.
Keep your numbers
Existing numbers port across and stay yours. New geographic and mobile DIDs can be added to the same trunk.
Fraud protection
Spend caps, destination blocking and alerting on the account, so a compromised handset cannot run up thousands overnight.
IP authentication
Trunks are locked to your addresses with ACLs rather than left open to registration attempts from anywhere.
Failover routing
If your connection drops, inbound calls divert to a mobile or a second site instead of ringing into nothing.
FAQ
Questions about SIP trunking
Almost certainly, if it was installed in the last fifteen years or so. Tell us the make, model and firmware and we will confirm before you order anything. Asterisk and FreePBX systems are SIP-native and need only configuration.
Channels are concurrent calls, not staff — most offices need roughly one per four or five people. We size it from your own call records rather than guessing; there is a fuller explanation in our guide to sizing SIP channels.
About 100 kbps per concurrent call on G.711, so twenty simultaneous calls is roughly 2 Mbps each way. That is trivial on a modern connection; what matters more is that voice traffic is prioritised over large uploads.
Inbound calls follow the failover destination you set — a mobile, another site, or voicemail. The trunk itself does not need to be reconfigured; the diversion is held on our side, not yours.
Related
Other services we provide
UK Tier-1 Voice Routes
Premium UK termination on Pay As You Go terms, billed per second.
UK DIDs — Landline & Mobile
Geographic and mobile numbers delivered over SIP to your PBX.
Asterisk PBX Installation
On-premises or cloud builds, fully configured and documented.
ISDN Switch-Off Migration
Move off retiring analogue and ISDN lines without losing a number.
Firewall & ACL Configuration
Tight access control so only your traffic ever reaches your PBX.
Still paying ISDN line rental?
Send us a recent bill and we will show you what the same capacity costs as a SIP trunk.