E.164 phone numbers: the international numbering standard explained

Who this guide is for

Audience: VoIP administrators and developers configuring number formats.

Understand how VoIP calling works — SIP, PBX, codecs, trunking, DID numbers, STIR/SHAKEN, and the protocols behind business phone systems.

Use this guide when you want the setup to be correct the first time and easy for another admin, manager, or supervisor to verify later.

What this workflow helps you accomplish

E.164 is the international numbering standard maintained by the ITU that defines the format and structure of telephone numbers worldwide. Every public telephone number in the world follows E.164, and VoIP systems require numbers in this format for proper routing between carriers.

This workflow matters because numbers, routing, access, and reporting in CallOrbit are connected. Skipping one setup detail usually creates avoidable support work later.

• Step 1: Use the correct E.164 format: a full E.164 number starts with a plus sign (+), followed by the country code (1-3 digits), the national destination code or area code (variable length), and the subscriber number. Example: +14155551234 (US), +442071234567 (UK), +27123456789 (South Africa).
• Step 2: Strip formatting for system storage: store phone numbers as pure E.164 strings without spaces, dashes, parentheses, or dots. Formatting characters add parsing complexity for SIP routing, CTI integrations, and API lookups. Display formatting should only be applied at the UI layer.
• Step 3: Check country code assignment by region: +1 covers the US and Canada (NANP), +44 is the UK, +61 is Australia, +27 is South Africa, +49 is Germany, +33 is France, +81 is Japan, and +86 is China. Each country code prefix determines how the rest of the number is structured.
• Step 4: Validate E.164 in your VoIP setup: most SIP providers and PBX systems reject numbers that are not in proper E.164 format. Configure outbound dial plans to normalise dialled numbers — strip trunk prefixes, add country codes, and prefix with + before routing to the SIP trunk.
• Step 5: Avoid common E.164 mistakes: storing numbers without the country code (so they cannot be dialled internationally), omitting the + prefix (some platforms treat it as optional but others reject it), and confusing country code with area code (area codes are part of the national number, not the country prefix).

Setup checklist

• Use the correct E.164 format: a full E.164 number starts with a plus sign (+), followed by the country code (1-3 digits), the national destination code or area code (variable length), and the subscriber number. Example: +14155551234 (US), +442071234567 (UK), +27123456789 (South Africa).
• Strip formatting for system storage: store phone numbers as pure E.164 strings without spaces, dashes, parentheses, or dots. Formatting characters add parsing complexity for SIP routing, CTI integrations, and API lookups. Display formatting should only be applied at the UI layer.
• Check country code assignment by region: +1 covers the US and Canada (NANP), +44 is the UK, +61 is Australia, +27 is South Africa, +49 is Germany, +33 is France, +81 is Japan, and +86 is China. Each country code prefix determines how the rest of the number is structured.
• Validate E.164 in your VoIP setup: most SIP providers and PBX systems reject numbers that are not in proper E.164 format. Configure outbound dial plans to normalise dialled numbers — strip trunk prefixes, add country codes, and prefix with + before routing to the SIP trunk.
• Avoid common E.164 mistakes: storing numbers without the country code (so they cannot be dialled internationally), omitting the + prefix (some platforms treat it as optional but others reject it), and confusing country code with area code (area codes are part of the national number, not the country prefix).

Operational follow-up

After you complete this flow, confirm the live experience from both the agent and customer side so ownership, routing, permissions, and reporting all match what the workspace expects.

If your team is rolling this out across multiple users, queues, or phone numbers, pair this article with the broader knowledge base and the relevant routing or numbers guides to keep deployment consistent.

• What is the CallOrbit Knowledge Base for? — It is the public help hub for how CallOrbit works, covering numbers, webphone setup, SIP, extensions, routing, users, roles, and billing basics.
• Can customers read this without signing in? — Yes. The Knowledge Base now lives on a public route so customers can read setup guidance before or after they enter the portal.
• Does the portal still have its own Knowledge Base page? — No. The signed-in portal navigation no longer carries a separate Knowledge Base page, and the old portal path now redirects to this public version.
• What is VoIP and how does it work? — VoIP (Voice over Internet Protocol) converts analogue voice signals into digital packets and transmits them over IP networks. Unlike traditional PSTN phone lines that require dedicated copper wiring per line, VoIP calls use your existing internet connection, which makes them cheaper, more flexible, and easier to scale.
• What is SIP trunking? — SIP trunking is a virtual connection that replaces traditional analogue phone lines or PRI circuits. A SIP trunk carries multiple concurrent voice channels over a single IP connection to your PBX or phone system, eliminating per-line hardware costs and monthly line rental fees.
• What is the difference between hosted PBX and cloud PBX? — Hosted PBX runs on dedicated virtual infrastructure managed by a provider, while cloud PBX uses shared multi-tenant cloud infrastructure. Hosted PBX suits organisations needing custom configuration and predictable pricing. Cloud PBX is better for instant scalability and per-user monthly billing.
• What is a DID number? — A DID (Direct Inward Dialling) number is a virtual phone number that routes directly to a specific extension, IVR menu, queue, or user within a phone system without an operator. DIDs decouple the phone number from the physical phone line, so you can have hundreds of numbers routed through a single SIP trunk.
• What are G.711, Opus, and G.729 codecs used for? — These are VoIP codecs that convert voice into digital data. G.711 uses 64 Kbps for toll-grade quality and is the PSTN standard. Opus uses 6-510 Kbps and adjusts to network conditions. G.729 uses 8 Kbps for bandwidth-constrained links. The right codec depends on your available bandwidth and call quality requirements.

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