Capacity by design
Channels are sized to the number of simultaneous calls your busiest hour requires — and resized as you grow.
Connect your PBX or contact center to the global phone network over SIP. Direct routes to 206 destinations, numbers in 120+ countries and per-minute pricing scale from one office to an enterprise fleet.
SIP trunking replaces physical phone lines with a virtual connection over the internet. The trunk is also called a SIP line or SIP circuit.
Your existing hardware or software PBX connects to KataTelecom using SIP, and inbound and outbound calls travel as data. No PRI circuits and no per-line rental are required: capacity grows through concurrent channels on the trunk.
The connection brings PBX capacity, international numbers and destination-based routing together. Codec, transport and caller ID settings are confirmed for the selected account and routes.
Channels are sized to the number of simultaneous calls your busiest hour requires — and resized as you grow.
Attach eligible local, toll-free or multichannel numbers from KataTelecom’s international inventory.
Use destination-based international tariffs from the same controlled PBX connection.
Present approved numbers where the destination, regulation and account configuration permit.
Agree the compatible codec, signalling and transport profile before production traffic is enabled.
Standards-based platforms including Asterisk, 3CX and Cisco can connect over SIP after configuration review.
A business SIP trunk can support a single office, while enterprise SIP trunking consolidates external routes across larger estates and international sites.
Keep extensions and call flows in your chosen PBX while KataTelecom supplies numbers and external routes.
Replace physical carrier circuits without forcing an immediate change to the internal phone platform.
Consolidate external voice routing while preserving local teams, extensions and country number requirements.
Global SIP trunking connects a standards-based PBX to international numbers and outbound destinations through one managed voice edge.
outbound destinations in the current tariff catalog
countries represented across KataTelecom number services
Local, toll-free and mobile numbers attachable to the trunk
Popular trunk destinations include the UK (+44), US (+1) and Germany (+49) — with number documentation handled per country. Our SIP trunk provider UK coverage includes +44 local and national numbers.
SIP trunk pricing has two parts: monthly numbers and per-minute outbound rates. Channel capacity is confirmed separately for the simultaneous-call profile on each account.
Concurrent-call profile confirmed per account.
Country, number type, documents and inventory determine the final monthly price.
Compare number prices →Destination-level rates apply to outgoing calls and are detailed in the route sheet.
Open calling rates →| Destination | Outbound from | Unit |
|---|---|---|
| 🇺🇸United States (USA) | $0.0255 | per minute |
| 🇬🇧United Kingdom | $0.143 | per minute |
| 🇩🇪Germany | $0.0251 | per minute |
| 🇰🇿Kazakhstan | $0.0744 | per minute |
| 🇷🇺Russian Federation | $0.0727 | per minute |
| 🇨🇦Canada | $0.0255 | per minute |
| 🇫🇷France | $0.0554 | per minute |
| 🇪🇸Spain | $0.187 | per minute |
| 🇵🇱Poland | $0.0248 | per minute |
| 🇮🇹Italy | $0.0247 | per minute |
A SIP trunk is one practical form of VoIP for a business PBX. PRI is a physical carrier circuit; VoIP is the wider technology category that also includes apps, phones and hosted PBX services.
| Factor | SIP trunk | PRI | Other VoIP service |
|---|---|---|---|
| Capacity | Concurrent SIP channels | Fixed circuit channels | Depends on the service |
| Scaling | Capacity adjusted by configuration | Additional physical capacity | Plan or platform dependent |
| Cost model | Channels, numbers and usage | Circuit rental and usage | Subscription or usage |
| Geography | International routes through one edge | Local carrier footprint | Provider dependent |
| Setup | Credentials, PBX configuration and test | Physical installation and provisioning | Account or application setup |
When migrating from PRI or changing a SIP trunk provider, document current numbers, peak concurrent calls, caller ID rules and failover before moving production traffic.
VoIP describes voice carried over IP networks. SIP trunking is the PBX-to-provider connection that carries multiple VoIP calls to and from the public phone network.
The sequence stays the same whether you start with a new PBX or migrate an existing business phone system.
Share PBX platform, countries, inbound numbers and expected simultaneous calls.
Agree the technical profile and obtain the server, authentication and routing settings.
Enter the settings, validate inbound and outbound routes, then move approved traffic to production.
Billing, compatibility, porting and emergency-calling responsibilities are stated directly before activation.
A SIP trunk is a virtual voice connection between your PBX and the public phone network. It replaces a physical PRI or analogue line with an internet-based SIP connection.
SIP line, SIP circuit and SIP trunk are commonly used for the same type of virtual PBX connection. Capacity is measured by the number of calls that can run at the same time.
Allow one channel for every simultaneous inbound or outbound call. Use your busiest hour and expected growth to size the trunk, then confirm the final capacity with our team.
UK number availability and documentation vary by number type and locality. Our team can check the current inventory and attach eligible KataTelecom numbers to your configuration.
Yes — KataTelecom operates as a SIP trunk provider in the UK with +44 local and national numbers. A UK SIP trunk uses the same per-minute rates and channel model; number documentation requirements are confirmed before activation.
VoIP is the broad technology for voice over internet networks. A SIP trunk is a business connection that carries multiple VoIP calls between a PBX and the phone network.
Billing combines trunk capacity, any monthly DID number fees and per-minute outbound rates by destination. The detailed rate sheet confirms route-specific billing increments and current prices. Payment methods & refund policy →
Standards-based PBX platforms such as Asterisk, 3CX and Cisco can connect over SIP. Share the product and version with our team so the exact signalling, codec and authentication profile can be confirmed.
Number porting may be available depending on the country, number type and current provider. KataTelecom checks eligibility and documentation before a port is scheduled.
KataTelecom SIP trunks do not include emergency calling (E911/999/112). Customers must maintain alternative means of reaching emergency services. See the service agreement for details.
Send us your PBX platform, destination mix, number requirements and expected concurrent calls. We will confirm the technical and commercial profile.