Getting Started • Updated April 11, 2026

Track calls, recordings, and performance inside CallOrbit

Use CallOrbit reporting and call history to review outcomes without sending teams into separate tools.

Audience: Customer admins, analysts, and QA users. This guide focuses on operational setup inside the CallOrbit platform.

Launch a new CallOrbit workspace with the right customer, users, and voice basics.

  • Search call logs for recordings, transcripts, and follow-up history.
  • Open analytics to watch service levels, outcomes, and operational trends.
  • Use QA and analyst roles when someone needs read-focused access instead of editing rights.

Who this guide is for

Audience: Customer admins, analysts, and QA users.

Launch a new CallOrbit workspace with the right customer, users, and voice basics.

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

Use CallOrbit reporting and call history to review outcomes without sending teams into separate tools.

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: Search call logs for recordings, transcripts, and follow-up history.
  • Step 2: Open analytics to watch service levels, outcomes, and operational trends.
  • Step 3: Use QA and analyst roles when someone needs read-focused access instead of editing rights.

Setup checklist

  • Search call logs for recordings, transcripts, and follow-up history.
  • Open analytics to watch service levels, outcomes, and operational trends.
  • Use QA and analyst roles when someone needs read-focused access instead of editing rights.

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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