Connecting Your Phone Line to the Voice Agent

This guide explains how to route real phone calls to your ChatLab voice agent, whatever telephony you have today - a modern VoIP number, a fresh number, or an existing line on an old PBX. For enabling the feature in ChatLab and getting your number's SIP address, see the Phone Calls article first; this guide focuses on the telephony side: getting a call from the outside world to that SIP address.

How it works

Telephony today splits into two "worlds":

• Legacy telephony - ISDN/analog lines and on-premise PBX systems. Calls travel over the carrier's lines, not the internet.
• VoIP telephony - numbers and calls carried "over SIP" (the internet-telephony protocol). This is where modern services live, including the ChatLab voice agent.

The agent has an address in the VoIP world called a SIP URI (think of it as an "internet address" for the phone line). For a call to reach the agent, the number has to enter the VoIP world at some point. A VoIP provider makes that happen - a company that supplies numbers and telephony over the internet, for example Zadarma (simple web panel, popular choice), or Twilio / Telnyx (international, more technical).

Your job comes down to two things:

1. get the call to a number that a VoIP provider handles, and
2. in that provider's panel, set a forward to the agent's SIP URI.

The agent runs in the cloud - you install nothing on your side or on your PBX.

Before you start: get your SIP URI

Enable voice on the bot and register your number in ChatLab to obtain the exact SIP address to forward to. The full steps (enable ElevenLabs Voice, copy the SIP URI, register the number, set security limits) are in the Phone Calls article. You will end up with a value like:

+1XXXXXXXXXX@sip.rtc.elevenlabs.io:5060;transport=tcp

Host sip.rtc.elevenlabs.io, port 5060 (TCP) or 5061 (TLS), no authentication. That SIP URI is the destination every option below points to.

Choose your setup

Pick the path that matches what you have today.

Option 1 - you already have VoIP telephony

This assumes your telephony already "speaks SIP": a VoIP/cloud PBX (for example 3CX, FreePBX, or a cloud PBX from your provider), or a number hosted directly with a VoIP provider. Then you only set a forward:

1. Log in to your VoIP provider panel (or your VoIP PBX).
2. Find call forwarding for the number or extension.
3. Set the destination to an external SIP URI and paste your agent's SIP URI.
4. Save and place a test call.

Option 2 - you have an old PBX and want to keep your main number

An old PBX (ISDN or analog) does not "speak SIP", so you cannot point it straight at the agent. You do not need to replace it or change your main number, though. The simplest approach: the PBX forwards the call to a regular VoIP number, and that number (at the provider) bridges it to the agent's SIP URI. To the PBX this is just a forward to an external number - it never needs to know about SIP.

How the call travels:

1. The caller dials your main number (unchanged).
2. The PBX forwards the call to the agent's VoIP number (a regular phone number) - immediately, or on a schedule (night mode).
3. The VoIP provider that hosts that number routes the call to the agent's SIP URI.
4. The AI agent answers.

There are three ways to do this (pick one):

A. New VoIP number + forward the PBX to it (recommended).

1. Buy a virtual VoIP number from a provider (for example Zadarma).
2. Point that number at the agent (forward to the SIP URI, as in Option 1).
3. On the PBX, forward your main number to that VoIP number (ideally on a schedule / night mode).

Your main number and PBX stay unchanged. Conditions: the PBX must be able to forward to an external (city) number; a forwarded call uses a channel and may be billed as outbound (call minutes on the PBX owner's side).

B. Port your main number to a VoIP provider.

You move the main number to a VoIP provider, so it becomes a VoIP/SIP number. Note: an old ISDN PBX cannot receive a number delivered over SIP, so this option means retiring the old PBX - replacing it with a VoIP/cloud PBX, or running with no PBX if the number only needs the agent. After porting, set the forward in the provider panel (as in Option 1). Choose this only if you are modernising the phone system anyway.

C. ISDN-to-VoIP gateway.

A device that bridges the ISDN world (your PBX) and the VoIP/SIP world. Here it converts the PBX's outbound traffic from ISDN to SIP toward the agent (direction ISDN to VoIP), so the PBX can push calls to the agent without an intermediate VoIP number. It needs hardware purchase and setup (IT/Telco support) - an option for advanced cases when A and B do not fit.

Option 3 - you want a separate, new number for the agent

1. Buy a new VoIP number from a provider (for example Zadarma).
2. Point it at the agent (forward to the SIP URI, as in Option 1).

Publish that number as an AI line (for example a 24/7 hotline). Your existing telephony and main number stay untouched. This is the quickest start and a good fit for a pilot. Trade-off: the agent only handles people who dial that specific number.

Optional: route only some calls to the AI

If the agent should answer only in certain windows (for example after business hours), use a time rule on the telephony side - in whichever element directs the traffic:

• when the number is already VoIP, in the VoIP provider panel (for example a forwarding schedule in Zadarma);
• when an old PBX forwards your main number (Option 2), in the PBX schedule / night mode.

The telephony decides when a call goes to the AI and when it goes to your existing destination - the agent does not switch calls between numbers.

In the rule you define both directions:

• business hours - forward to your existing number / team;
• after hours - forward to the agent's SIP URI.

Multiple simultaneous calls

• The agent handles several concurrent calls independently (up to 5 by default, can be raised).
• There is no per-channel fee on ChatLab's side - calls are billed per minute, so more concurrent calls just use minutes faster.
• The number of simultaneous channels may, however, be limited or charged by your line provider, depending on their plan (for example a SIP trunk channel bundle).

What the agent does not do

• The agent handles the whole conversation on its own and does not transfer a live call to another number or to a person. Where a human is needed, the agent collects contact details and you get the transcript after the call.
• Deciding which calls reach the AI (for example by schedule) is done by your telephony, not by the agent.

Glossary

• VoIP - telephony over the internet.
• SIP - the protocol that sets up VoIP calls.
• SIP URI - the agent's address in the SIP world, for example number@sip.rtc.elevenlabs.io. Works like an "internet address" for a phone line.
• SIP trunk - a SIP link a provider uses to deliver/receive calls over SIP instead of a classic line; it can carry several simultaneous channels (calls).
• ISDN - older digital carrier telephony (city lines from before the VoIP era).
• PBX - the device or service that distributes calls inside a company (extensions, forwarding, night mode).
• VoIP provider - a company that supplies numbers and telephony over SIP (for example Zadarma, Twilio, Telnyx).
• Virtual VoIP number - a phone number hosted at a VoIP provider with no physical line; it can be forwarded anywhere, including to a SIP URI.
• Porting - moving a number to a different provider while keeping the number itself.
• ISDN-to-VoIP gateway - a device that translates calls between the ISDN world and the VoIP/SIP world.