Softphone setup for CallOrbit

Who this guide is for

Audience: Agents and remote workers setting up desktop softphones.

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

Install, configure, and connect a desktop softphone application to CallOrbit using SIP credentials so you can make and receive calls from your computer without opening a browser.

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: Choose a SIP-compatible softphone application — for example MicroSIP, Zoiper, Linphone, or X-Lite — and download the version that matches your operating system. These applications work with standard SIP credentials from CallOrbit.
• Step 2: Open the softphone settings panel and enter the CallOrbit SIP domain as the registrar server, your assigned SIP username as the authentication ID, your SIP password, and leave the transport protocol set to UDP or TCP as shown in the portal SIP setup page.
• Step 3: Configure the softphone audio settings so the correct microphone, speaker, and ringer devices are selected and the codec list is prioritised for G.711 or Opus, which are the codecs that CallOrbit supports for clear voice quality.
• Step 4: Register the softphone by clicking the Connect or Register button in the application and confirm the status indicator changes to a green Registered state. This means the softphone is authenticated and ready for calls.
• Step 5: Place a test call using the softphone keypad or contacts list and verify that the recipient hears your voice clearly and that their caller ID displays your assigned CallOrbit number. Check the softphone call log for duration and quality metrics.
• Step 6: Repeat the configuration steps for each remote worker or agent who needs a desktop softphone. Each person needs their own SIP credentials from the CallOrbit Voice Setup page and should use their unique extension for internal calling.

Setup checklist

• Choose a SIP-compatible softphone application — for example MicroSIP, Zoiper, Linphone, or X-Lite — and download the version that matches your operating system. These applications work with standard SIP credentials from CallOrbit.
• Open the softphone settings panel and enter the CallOrbit SIP domain as the registrar server, your assigned SIP username as the authentication ID, your SIP password, and leave the transport protocol set to UDP or TCP as shown in the portal SIP setup page.
• Configure the softphone audio settings so the correct microphone, speaker, and ringer devices are selected and the codec list is prioritised for G.711 or Opus, which are the codecs that CallOrbit supports for clear voice quality.
• Register the softphone by clicking the Connect or Register button in the application and confirm the status indicator changes to a green Registered state. This means the softphone is authenticated and ready for calls.
• Place a test call using the softphone keypad or contacts list and verify that the recipient hears your voice clearly and that their caller ID displays your assigned CallOrbit number. Check the softphone call log for duration and quality metrics.
• Repeat the configuration steps for each remote worker or agent who needs a desktop softphone. Each person needs their own SIP credentials from the CallOrbit Voice Setup page and should use their unique extension for internal calling.

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