> ## Documentation Index
> Fetch the complete documentation index at: https://help.verbu.com/llms.txt
> Use this file to discover all available pages before exploring further.

# Most PBX solutions

> The general pattern for connecting Verbu to any SIP-based PBX — 3CX, Asterisk, Telavox, FreePBX, and most others.

The vast majority of business phone systems sold today speak **SIP** — the same protocol Verbu uses. If your PBX speaks SIP (and it almost certainly does), connecting it to Verbu is the same job in every system: forward the right calls to a SIP destination we give you.

This page is the general pattern. For Unitel specifically, see [Unitel](/en/pbx/unitel).

## The shape of the setup

Three things, in this order:

1. **Verbu gives you a SIP endpoint** — a small piece of text that looks something like `sip:agent-XXXX@sip.verbu.com`
2. **You add it to your PBX as a destination** — under whatever the PBX calls "external SIP", "SIP trunk", or "SIP URI forward"
3. **You point an incoming number at that destination** — so calls to that number end up at your agent

That's the whole job. Everything below is just where to click in different systems.

## Before you start

You need:

* Admin access to your PBX
* The number you want the agent to answer on
* The Verbu SIP endpoint — find it under **Phone numbers → Connect existing number** in the dashboard, or ask your Verbu contact

## The general steps

<Steps>
  <Step title="Add Verbu as a SIP destination">
    In your PBX admin, find the section for outbound or external SIP routing. Depending on the system, this is called:

    * **SIP Trunks** (3CX, FreePBX)
    * **Outbound destinations** (Asterisk-based systems)
    * **External numbers** (Telavox, smaller cloud PBXs)

    Add a new destination with the Verbu SIP endpoint as the URI.
  </Step>

  <Step title="Allow the right audio codecs">
    Make sure the trunk allows `PCMU` (G.711 µ-law) and `PCMA` (G.711 A-law). These are the standard codecs Verbu uses. Most PBXs allow them by default.
  </Step>

  <Step title="Route the incoming number to Verbu">
    Find the routing rule for your incoming number (sometimes called an "inbound route" or "DID route"). Change its destination to the SIP destination you just created.

    If you only want some calls to go to Verbu — for example after hours — add a time-of-day condition first.
  </Step>

  <Step title="Save and test">
    Apply the changes and call the number from your mobile. You should hear your agent greet you within a second or two.
  </Step>
</Steps>

## Common gotchas

<AccordionGroup>
  <Accordion title="Calls connect but there's no audio in one direction">
    Almost always a NAT or codec problem. Make sure your PBX is configured to use the public IP for SIP signalling, and that the codec list starts with G.711 (PCMU / PCMA).
  </Accordion>

  <Accordion title="The PBX strips the caller's number">
    Some setups overwrite the caller ID with the PBX's own number. The agent then can't tell who's calling and won't find them in your CRM. Look for a setting called **CLIP** or **Caller ID** on the outbound SIP trunk and make sure it passes the original caller's number through.
  </Accordion>

  <Accordion title="The PBX won't let me forward outside the system">
    Some on-premise PBXs are locked down to internal numbers only. You need to allow outbound SIP destinations on the trunk. If you're not comfortable changing that, ask your reseller — it's a small one-time tweak.
  </Accordion>

  <Accordion title="Calls drop after a few minutes">
    The PBX or a firewall in the middle is closing the SIP session. Look for **session timer** or **re-INVITE** settings on the trunk and disable the aggressive timeout. Verbu keeps sessions alive on its side.
  </Accordion>
</AccordionGroup>

## 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. Forward them this page and the SIP endpoint Verbu gave you — they'll have it done in under an hour.

You can also ask Verbu support for help. Send us the make and model of your PBX and we'll send back the exact steps.

## Next steps

* For Unitel specifically, see [Unitel](/en/pbx/unitel)
* If you'd rather skip the PBX entirely, see [Verbu as a soft phone](/en/pbx/verbu-as-a-soft-phone)
* Make sure your agent itself is set up: [Building your first agent](/en/getting-started/building-your-first-agent)