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Most modern business phone systems speak SIP – the same protocol Verbu uses. Whatever the brand, there are two ways to get calls from your PBX to the agent: forward them to a Verbu phone number, or let Verbu sign in to your PBX as a SIP user. This page is the general pattern. For Unitel specifically, see Unitel. For IPNordic, see IPNordic.

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The two ways

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1. Forwarding to a Verbu number

Your account has at least one Verbu phone number from the start. Your PBX forwards the relevant calls to that number – exactly like forwarding to a mobile. Choose this if you want to get going quickly, or if your PBX doesn’t support SIP users.

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2. SIP connection – Verbu signs in to your PBX

Verbu signs in to your phone system as one or more SIP users (extensions) – exactly like a desk phone signing in. Calls you send to those extensions land with the agent. Choose this if your phone system supports SIP users. Most do – both on-premise systems like 3CX, Asterisk and FreePBX and hosted solutions like Unitel/OneConnect and IPNordic.

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Before you start

You need:
  • Admin access to your PBX, or contact with whoever manages it
  • The number you want the agent to answer on
  • For forwarding: your Verbu number – find it under Agent → Phone, in the Phone numbers tab
  • For a SIP connection: a SIP user in your PBX for Verbu with an auth username, password and extension

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Step by step: forwarding (way 1)

1

Find the Verbu number

Select the agent in the dashboard, go to Agent → Phone, and open the Phone numbers tab. Copy the number assigned to the agent.
The list of available phone numbers on the account
2

Create Verbu as a destination in your PBX

In your PBX admin, find the section for external destinations. It goes by different names – for example External numbers, Outbound destinations or simply forwarding. Create the Verbu number as a destination.
3

Point the incoming number at Verbu

Find the routing rule for the number (sometimes called an “inbound route” or “DID route”) and set the Verbu destination.If you only want some calls to go to Verbu – for example after hours – add a time-of-day rule first. If your team should get the first chance, set Verbu as the fallback after unanswered calls.
4

Save and test

Apply the changes and call from your mobile. The agent should greet you within a couple of seconds.

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Step by step: SIP connection (way 2)

1

Create a SIP user for Verbu in your PBX

In your PBX admin, create a new SIP user that Verbu can sign in with. It’s typically called an Extension (3CX, FreePBX), a SIP user or an endpoint (Asterisk-based systems).Note the auth username, password and extension.
2

Add the SIP connection in Verbu

Select the agent, go to Agent → Phone → Phone numbers, and scroll down to the SIP connections section. Click Add SIP connection.Choose the Provider: Uni-tel / OneConnect, IPNordic, or Custom for every other system. Enter your main number. For known providers, the domain, port and transport fill in automatically – if you choose Custom, enter your PBX’s SIP domain, port and transport yourself.
Add SIP connection with provider, main number and domain
3

Add an extension

Click Add extension on the connection, and enter the SIP user’s auth username, password and extension. Add one extension per SIP user.
Add extension with fields for auth username, password and extension
4

Check that Verbu is signed in

When the extension shows the registered badge, Verbu is signed in and ready to take calls.
SIP connections with main number and registered extensions
5

Send the calls to the extension

In your PBX: point the relevant calls at the extension Verbu is signed in on – directly, via a queue, or via a ring group. Time rules and fallback are set up the same place as for any other extension.

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Common gotchas

Almost always wrong credentials or a wrong domain, port or transport. Check the auth username and password against what’s in the PBX, and that the SIP user is active.If your PBX runs on-premise behind a firewall, it may also be blocking registrations from outside. Ask your PBX administrator.
Almost always a NAT or codec problem. Make sure your PBX uses the public IP for SIP signalling, and that the codec list starts with G.711 (PCMU / PCMA).
Some setups overwrite the caller ID with the PBX’s own number. Look for a setting called CLIP or Caller ID and make sure the original caller’s number passes through – both for forwarding and on extensions.
The PBX or a firewall in the middle is closing the SIP session. Look for session timer or re-INVITE settings and disable the aggressive timeout.

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When in doubt

If you’re not sure how to do this in your specific PBX, the people who set it up for you (Telavox, your reseller, an internal IT team) will know. Send them this page – all they need to do is create a SIP user or a forwarding rule. You can also ask Verbu support. Send us the make and model of your PBX and we’ll send back the exact steps.

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Next steps