Making your first call on CallOrbit

Who this guide is for

Audience: New users testing their first outbound call on CallOrbit.

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

Place your first outbound call using the CallOrbit webphone or a connected SIP device, verify audio is working, and confirm your number and routing are set up correctly.

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: Open the Webphone from the portal sidebar and check that your microphone and speaker are detected. The webphone will display your assigned caller ID number at the top of the dialler panel.
• Step 2: Enter a phone number in the dialler using full international format — for example +14155551234 for US numbers or +442071234567 for UK numbers — then press the call button to place the call.
• Step 3: Listen for the ringtone and confirm the recipient hears your voice and their caller ID shows your CallOrbit number. If there is no audio, check your browser permissions for microphone access and try again.
• Step 4: End the call and open the Call Logs page from the portal sidebar to verify the call record appears with the correct duration, caller ID, destination number, and call outcome.
• Step 5: Test inbound calling by dialling your CallOrbit number from another phone and confirm the call reaches the destination you configured — IVR, queue, extension, or forwarding target.

Setup checklist

• Open the Webphone from the portal sidebar and check that your microphone and speaker are detected. The webphone will display your assigned caller ID number at the top of the dialler panel.
• Enter a phone number in the dialler using full international format — for example +14155551234 for US numbers or +442071234567 for UK numbers — then press the call button to place the call.
• Listen for the ringtone and confirm the recipient hears your voice and their caller ID shows your CallOrbit number. If there is no audio, check your browser permissions for microphone access and try again.
• End the call and open the Call Logs page from the portal sidebar to verify the call record appears with the correct duration, caller ID, destination number, and call outcome.
• Test inbound calling by dialling your CallOrbit number from another phone and confirm the call reaches the destination you configured — IVR, queue, extension, or forwarding target.

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.

Related CallOrbit guides