A SIP connection is the phone line your voice agent talks over. The form is deliberately generic: it asks for the handful of details every SIP provider issues, so you can connect whichever carrier you already work with — keeping your existing numbers, routes, and per-minute rates.
Set one up once and every agent and campaign can place calls through it.
Quick start
Collect the details from your provider — five values, no more.
Create the connection and fill in the Settings tab.
Decide whether registration is needed — it depends on how your carrier authenticates you.
Enable and test the line with Test Call.
How it works
BSG doesn't resell you a line. You bring a SIP trunk from your own carrier, and the connection stores the credentials needed to authenticate against it. When an agent places or answers a call, it does so through your trunk, billed by your carrier at your rates.
Because the form is provider-agnostic, the Provider field reads generic. It's read-only — fixed when the connection is created — and it doesn't limit which carrier you use.
What you manage where
Where |
What you do there |
Your telephony provider |
Buy numbers, set up the trunk, get SIP credentials, manage routes and call rates. |
Store those credentials as a SIP connection, enable it, and test the line. |
Call charges stay with your carrier. BSG stores the credentials and uses them for your agents' calls.
What to collect from your provider
Ask your carrier's support or account manager for the SIP trunk details. You need:
Value |
What to ask for |
Server |
The SIP registrar or proxy hostname used for call setup. |
Username |
The SIP account or authentication user for the trunk. |
Password |
The secret paired with that username. |
Phone |
The number assigned to the trunk — this becomes the caller ID your agent presents. |
Registration |
Whether the trunk expects the connection to register and stay logged in, or authenticates you another way. |
If your carrier gives you a longer configuration sheet, most of it is for PBX setups and isn't needed here.
Creating the SIP connection
Open Connections in the Voice Agent cabinet and click Add SIP connection.
Rename the connection at the top to something you'll recognise, such as the carrier's name. The generated identifier below it stays put, so you can still match the connection against its logs.
On the Settings tab, fill in Server, Username, Password, and Phone from the details you collected.
Set Register as described below.
Tick Enabled.
Click Save.
The password field is masked. Use the eye icon to reveal what you've pasted before saving — a trailing space copied along with a credential is a common cause of an authentication failure.
Registration or not
Register → Enabled when your provider expects the connection to log in and stay logged in. The connection then registers with the SIP server and keeps that registration alive, which is how most carriers handle username-and-password trunks — and it's what allows inbound calls to find you.
Register → off when your provider authenticates a different way, typically by whitelisting the IP address that traffic arrives from. In that setup, registering isn't required and may be rejected outright.
If you're unsure, ask your carrier which model your trunk uses. Getting this wrong is the most common reason a correctly filled-in connection still won't place calls.
Testing the line
Click Test Call at the top of the connection. This checks that the connection can reach your provider and set up a call, without needing an agent or campaign first.
Open the Logs tab to see what happened. Latest Calls lists recent test calls and connection activity, newest first, with a refresh control to pull the latest:
Column |
What it tells you |
Audio |
A play button for the call recording, so you can hear how the line actually sounded. |
Phone |
The number that was dialled. |
Status |
How the call ended overall — COMPLETED for one that connected and finished normally. |
Reason |
The carrier's own explanation for that outcome, such as NORMAL_CLEARING on a healthy call. |
Date and Duration |
When the call ran and how long it lasted. A duration of zero means the call never connected. |
A successful test looks like COMPLETED / NORMAL_CLEARING with audio you can play back. Anything else, read the Reason — it comes from your carrier, not from BSG, so quoting the Status and Reason pair to their support is the fastest way to get an answer. They can match it against their own logs for the same timestamp.
Audio playback is worth using even on a successful call: it catches one-way audio, codec problems, and delay that a green status won't show.
Keeping the line under control
Clearing Enabled takes the connection out of service for every agent using it, without deleting anything. That's the clean way to pause a line during maintenance or while you rotate credentials.
Keeping a second SIP connection for testing is worth doing: point test agents at it and leave production traffic on its own trunk, so an experiment can't disturb live calls.
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