SIP setup on CallOrbit

Who this guide is for

Audience: IT admins and technical users configuring SIP endpoints.

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

Configure SIP trunks, SIP users, and SIP endpoints on CallOrbit so your existing desk phones, PBX systems, or softphones can register and place and receive calls through the platform.

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 SIP section in Voice Setup and review the SIP domain, authentication credentials, and port settings that CallOrbit provides for your workspace. These values are used in every SIP device configuration.
• Step 2: Decide whether you need a SIP trunk for connecting an existing PBX or a SIP user for each individual desk phone or softphone endpoint. SIP trunks carry multiple channels while SIP users are tied to a single device and extension.
• Step 3: For each SIP endpoint, create a SIP credential set with a username, password, and optional IP access control list so only authorised devices and networks can register against your CallOrbit SIP domain.
• Step 4: Configure each desk phone, softphone, or PBX with the SIP domain, authentication credentials, and port settings from CallOrbit, and make sure the device is set to register with a keep-alive interval that matches the platform recommendation.
• Step 5: Verify registration by checking the SIP status page in the CallOrbit portal — each configured endpoint should display a green Registered status with the device IP and registration expiry timestamp visible.
• Step 6: Place a test call from each SIP endpoint to confirm two-way audio is working and that inbound calls to the assigned numbers reach the correct registered SIP device without delay.

Setup checklist

• Open the SIP section in Voice Setup and review the SIP domain, authentication credentials, and port settings that CallOrbit provides for your workspace. These values are used in every SIP device configuration.
• Decide whether you need a SIP trunk for connecting an existing PBX or a SIP user for each individual desk phone or softphone endpoint. SIP trunks carry multiple channels while SIP users are tied to a single device and extension.
• For each SIP endpoint, create a SIP credential set with a username, password, and optional IP access control list so only authorised devices and networks can register against your CallOrbit SIP domain.
• Configure each desk phone, softphone, or PBX with the SIP domain, authentication credentials, and port settings from CallOrbit, and make sure the device is set to register with a keep-alive interval that matches the platform recommendation.
• Verify registration by checking the SIP status page in the CallOrbit portal — each configured endpoint should display a green Registered status with the device IP and registration expiry timestamp visible.
• Place a test call from each SIP endpoint to confirm two-way audio is working and that inbound calls to the assigned numbers reach the correct registered SIP device without delay.

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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