SIP trunks UK, keep your PBX.
Why replace a working phone system when you can replace just the line into it? SIP trunks for UK businesses — when they beat hosted VoIP, integration with all major UK PBX brands, real costs, PSTN switch-off migration.
- 📞 What SIP trunks actually are
- ⚖️ SIP vs hosted VoIP decision
- 🔌 PBX integration (major UK brands)
- 💷 Real costs and channel sizing
- 🔄 ISDN-to-SIP migration process
- 📡 Network requirements
- 📠 Fax, modem, PSTN device migration
- 🚨 PSTN switch-off context
What SIP trunks are, in plain English.
A SIP trunk is the modern replacement for ISDN/PSTN voice lines into a business. Instead of physical copper phone lines or ISDN circuits coming into your building, the voice service arrives over your internet connection. Your existing on-premises PBX connects to the SIP trunk service via Ethernet rather than via specialised telephony hardware.
From the PBX's perspective, calls still come in and go out. Extensions still dial each other. Numbers still ring. Call recording, call routing, IVR menus — all the PBX's existing functionality continues working exactly as before. Only the delivery technology of the external lines has changed.
This matters because of the UK PSTN switch-off in January 2027. Every business currently on PSTN or ISDN must migrate to an IP-based voice service before then. SIP trunks are one of three migration paths (the others being hosted VoIP and Microsoft Teams Phone). The right path depends on whether your existing PBX has useful life remaining.
SIP trunks vs hosted VoIP — which is right.
Choose SIP trunks when:
- You have an existing on-premises PBX that's recent (under 10 years old) and still has useful operational life
- The PBX has custom configuration, call routing, or integrations that would be expensive to recreate on a hosted platform
- Your existing handsets are recent and work with the PBX — replacing them would add significant cost
- Staff are trained on the existing handsets and don't want disruption
- The per-user cost of hosted VoIP × user count is significantly higher than SIP trunk channels × channel count
Choose hosted VoIP when:
- Your existing PBX is old (10+ years), nearing end of life, or has been having reliability issues
- You don't have an existing PBX (greenfield deployment)
- You want modern features (mobile apps, AI receptionist, advanced analytics) that on-premises PBXs handle awkwardly
- Hybrid working is important — hosted VoIP's mobile apps are usually better than on-premises PBX mobile clients
- You want to eliminate on-premises hardware and the maintenance overhead
The honest test: if your PBX has been working reliably for years and you'd be replacing it just because of the PSTN deadline rather than because you actually want a new system — SIP trunks are right. If you'd be migrating away from the PBX anyway because it's outdated — hosted VoIP is right and you're getting the migration over with at the same time as the PSTN switch.
SIP trunks with the major UK PBX brands.
Most modern PBX systems support SIP trunking natively. Older systems may need a SIP gateway. Telexico has integrated SIP trunks with all the major UK brands.
Mitel
3300 ICP, MiVoice Business, MiVoice MX-One. Native SIP trunk support across modern Mitel platforms. Configuration via Mitel's system administration. Common UK enterprise install base.
Avaya
IP Office (popular UK SME platform), Aura platforms. Native SIP support. IP Office in particular is widely deployed in UK 20-200 user businesses and SIP-trunk-friendly.
Cisco
Unified Communications Manager (CUCM), BroadWorks platforms. Enterprise-grade SIP trunk integration. Used in larger UK organisations.
Panasonic
KX-NS, KX-NSX, KX-NCP series. SIP trunk support on modern models. Older KX-TDA series may need a SIP gateway.
Yeastar
S-series, P-series. Native SIP trunk support, popular in UK SMEs as a more affordable PBX alternative. Web-based configuration.
3CX
Software-based PBX running on Linux or Windows. Designed around SIP trunks natively. Common in UK businesses preferring software-defined infrastructure.
Asterisk-based
FreePBX, Issabel, Sangoma PBXact, AsteriskNow. Open source family with strong SIP trunk support. Common in UK IT-led organisations.
Grandstream
UCM6200/6300 series. Affordable SIP-native PBX option. Common in small UK businesses with technical IT staff.
Older PBX (pre-2010)
Legacy systems with only analogue or ISDN interfaces need a SIP gateway device to bridge to SIP trunks. Typically £200-500 one-off. Not pretty but works as a bridge before full PBX replacement.
How many SIP channels you actually need.
A SIP channel is one concurrent call capacity — the ability for one call to be in progress through the trunk. Channel sizing matters because you pay per channel; over-sizing wastes money, under-sizing means callers get busy signals.
Typical sizing ratios (for inbound/outbound mixed business):
- Office staff: 1 channel per 3-5 employees (most people aren't on calls at the same time)
- Sales / customer service teams: 1 channel per 1-2 employees (high concurrent call usage)
- Call centre / contact centre: typically 1 channel per agent (close to 100% utilisation)
Worked examples:
- 30-person professional services office (mixed roles): 8-10 channels typical
- 15-person sales team: 10-12 channels typical
- 50-seat call centre: 50 channels recommended
- 5-person small business: 3-4 channels (small reserve for inbound concurrent during busy periods)
Monitoring after deployment: good SIP trunk providers give you visibility into channel utilisation. Look at peak concurrent call counts over a month. If you're hitting your channel limit during peaks (callers getting busy signals), add channels. If you're chronically under 50% utilisation, you bought too many. Channels are usually adjustable mid-contract — review periodically.
How an ISDN-to-SIP cutover actually works.
Week 1 — Pre-migration audit. Engineer reviews your existing PBX: model, firmware, current configuration, integrations. Confirms SIP trunk compatibility. Identifies any required configuration changes or SIP gateway hardware. Calculates correct channel count. Audits non-voice PSTN dependencies (fax, alarms, EPOS) for parallel migration planning.
Week 1-2 — Number porting initiation. Submit number porting requests to existing PSTN/ISDN carrier. Standard UK port time: 7-14 working days. Multiple number ranges can port in parallel.
Week 2 — PBX configuration. SIP trunk endpoints configured on the PBX. Dial plans set up to route outbound calls via the SIP trunk. Authentication credentials configured. Test extensions created. Most modern PBX configuration is done remotely; older systems may need brief on-site engineer visit.
Pre-cutover — Test calls. Make and receive test calls through the SIP trunk on test numbers (separate from the production numbers being ported). Verify call quality, two-way audio, DTMF (touch tones for IVR menus), call transfer, call recording if used. Iron out any configuration issues.
Cutover day. Numbers port at the agreed time slot (usually morning, 8am-9am UK). PBX is switched to route all calls via the SIP trunk. Brief observation period (1-2 hours) with engineer support on standby. Old ISDN lines remain physically connected but no longer carry traffic.
Post-cutover (7-14 days). Continue monitoring call quality, channel utilisation, any specific issue patterns. After stability is confirmed, old ISDN lines are formally cancelled with the previous carrier.
Ongoing. Standard support relationship for the SIP trunk service. Channel additions/changes as needed. Monitoring of trunk health.
What your internet connection needs to support SIP.
SIP trunks deliver voice over your internet connection, so the connection quality determines voice quality. The bandwidth requirement is modest (~80-100 Kbps per concurrent call, so 10 channels = ~1 Mbps) — but the engineering layer matters more than the raw speed.
QoS prioritising voice traffic. The router needs to prioritise SIP signalling and RTP voice packets over other traffic. Without QoS, voice quality degrades during peak network usage. See the VoIP quality guide for the engineering detail.
SIP ALG disabled. Most consumer routers have buggy SIP ALG implementations that interfere with SIP traffic. Disable it. Modern SIP trunk providers handle NAT properly at their end via Session Border Controllers.
Static IP recommended. Many SIP trunk providers authenticate by IP address rather than (or as well as) username/password. Static IP at your office simplifies configuration. Typically £5-10/month from most business broadband providers.
Adequate upload bandwidth during peak. Each call uses ~80-100 Kbps upload. Even small offices with 10 concurrent calls need ~1 Mbps available upload. Standard business FTTP provides 100+ Mbps upload — typically not a constraint, but worth confirming if you have lower-tier service.
Failover recommended. 4G/5G failover keeps SIP trunks operational during primary internet outages. Critical for businesses where voice service uptime matters. See the failover guide for details.
What else needs migrating alongside.
SIP trunks handle voice calls cleanly. Several other things commonly running over the same PSTN/ISDN infrastructure need parallel migration planning:
Fax machines. Fax over IP (T.38 protocol) works but quality is variable, particularly for high-volume fax users. Most businesses migrate to fax-to-email gateways instead — incoming faxes arrive as PDF attachments, outgoing faxes sent via email gateway. Eliminates the physical fax machine.
Alarm panel dial-out. Burglar alarms, fire alarms, intruder alarms with PSTN dial-out monitoring don't work over SIP. Migration to GSM-based alarm modules required. Typically £150-400 per panel for the module, plus monitoring contract update with the alarm provider.
Lift emergency phones. Statutory requirement under EN 81-28. GSM-based lift phone units replace the PSTN versions. Typically £200-500 per lift including installation.
Modem-based EPOS terminals. Older card payment terminals using PSTN dial-up modems for authorisation don't work over SIP. Migration to IP-based card terminals — usually handled by the card acquirer (Worldpay, Stripe, etc.) at low/no cost.
Door entry / gate intercoms. PSTN dial-out door entry systems need GSM or IP-based replacements. Particularly common in apartment blocks and gated commercial premises.
All of these are typically caught in Telexico's PSTN audit alongside the SIP trunk migration so everything cuts over together rather than discovering surprises after the SIP cutover.
Review my current setup.
Not ready to switch yet? Send us your current contracts, bills, or photos of your existing equipment. We'll review what you have, what you're paying, and where you could simplify, consolidate or improve — without any pressure to buy anything from us.
We benchmark your existing broadband, phones, mobile and IT against current UK market pricing and what your business actually needs.
Real engineer review of your current connectivity, voice setup, WiFi, security and continuity — strengths, gaps, and where you're overpaying or underprotected.
We tell you honestly what's available at your postcode — FTTP, leased line, alt-net carriers — and which makes commercial sense for your operation.
For businesses still on ISDN or aging on-premises PBX — an honest cost-and-feature comparison before the 2027 BT switch-off forces a rushed decision.
No hard sell. No fixed package pressure. If we're not a better fit, we'll tell you straight — and recommend what is.
Frequently asked questions
What is a SIP trunk?
A SIP trunk is the modern replacement for ISDN/PSTN voice lines. Instead of physical copper telephone lines coming into your building, the voice service is delivered over your internet connection using the SIP (Session Initiation Protocol) signalling standard. Your existing on-premises PBX (phone system) connects to the SIP trunk via an Ethernet cable instead of an ISDN PRI cable. From the PBX's perspective, calls still come in and go out — only the delivery technology has changed.
Why would I want SIP trunks instead of hosted VoIP?
If you have an existing on-premises PBX that's working well and isn't ready to retire, SIP trunks let you keep that PBX and just replace the line into it. You preserve any existing PBX investment, custom call routing rules, integrations with door entry systems or paging systems, and the staff training around the existing handsets. The alternative — moving to hosted VoIP — replaces the whole phone system, which is the right answer when the PBX is genuinely past its useful life but unnecessary when the existing PBX is recent and capable.
How much do SIP trunks cost?
Typical UK SIP trunk pricing: £8-15 per channel per month for a standard business package including UK domestic minutes. A 'channel' is a simultaneous call capacity — a business with 30 employees rarely needs more than 8-10 concurrent channels because not everyone is on the phone simultaneously. So a 30-person office might need 8 channels × £10 = £80/month total. Compare with hosted VoIP at £15-25 per user — for 30 users, ~£500-750/month. SIP trunks are dramatically cheaper if you already have a functioning PBX.
Which PBX systems work with SIP trunks?
Most modern on-premises PBX systems support SIP trunking natively. Telexico has integrated SIP trunks with: Mitel (3300, MiVoice, MX-One), Avaya (IP Office, Aura), Cisco (Unified Communications Manager, BroadWorks), Panasonic (KX-NS, KX-NSX series), Yeastar (S-series, P-series), 3CX, Asterisk-based systems (FreePBX, Issabel, AsteriskNow), Grandstream UCM series, and various others. Very old PBX systems (pre-2010) may need a SIP gateway device to bridge between their analogue/ISDN interfaces and SIP — typically £200-500 one-off hardware.
What's involved in migrating from ISDN to SIP trunks?
Standard migration timeline: 7-14 working days. Steps: (1) Pre-migration audit — confirm PBX supports SIP, identify any required configuration changes, calculate channel count needed. (2) Number porting initiation — submit port requests to existing carrier. Takes 7-14 days. (3) PBX configuration — set up SIP trunk endpoints, dial plans, authentication. Usually done remotely or via brief on-site visit. (4) Test call setup — make and receive test calls via the new SIP path to verify configuration. (5) Cutover day — numbers port at agreed time, PBX switches from ISDN to SIP. Old ISDN lines disconnected after stability confirmed (typically 7-14 days post-cutover). UK businesses on ISDN have been migrating throughout 2024-2026 ahead of the January 2027 PSTN switch-off deadline.
Do SIP trunks need special internet connectivity?
Standard business broadband is usually sufficient for SIP trunks at the bandwidth level — each call uses ~80-100 Kbps so even 10 concurrent calls only need ~1Mbps. However, the connection needs to be configured properly: QoS prioritising voice traffic, SIP ALG disabled on the router (it usually breaks SIP), no double-NAT, reasonable upload speed during peak usage. A poorly configured 100Mbps connection can deliver worse SIP call quality than a properly configured 50Mbps one. The 'business broadband' part isn't really about speed; it's about the engineering layer underneath.
Can SIP trunks support fax and other PSTN-dependent equipment?
Voice — yes, perfectly. Fax — works via T.38 fax-over-IP protocol but quality and reliability are variable; many businesses migrate fax to fax-to-email gateways instead. Modems for ATM/EPOS/alarm dial-out — generally don't work over SIP/IP because they expect analogue line characteristics. Migration plan for these devices: alarm panels need GSM-based replacements, ATMs typically managed by the operator on IP networks, modem-based EPOS gets replaced with IP-based card terminals (your acquirer usually handles this). These non-voice migrations are typically caught in the pre-migration audit and planned alongside the SIP trunk cutover.
Tailored around your business.
Send us your current setup. We'll review what you have, what you pay, and where we can simplify, consolidate or improve it — no hard sell, no fixed-package pressure.