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How to connect an AI voice bot to your PBX: A1 VATS and Asterisk

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You can connect AI to your telephony without changing your number or carrier. For A1 VATS, an administrator connects the account in the console, after which employees' calls and recordings appear in the call log and are analyzed. For Asterisk, FreeSWITCH and other PBXs, a SIP connection with ready-made settings is used. The voice bot works on your number, and call results go to your CRM.

Below is what happens at each step, what your IT team will need to provide and where the current limits are. The full list of integrations with their status is on the integrations page.

Two modes: analytics and the voice bot

First decide what you are connecting. The steps depend on it.

Mode What the AI does What the PBX must provide
Speech analytics Receives employees' call recordings, transcribes and scores them Access to calls and recordings
Voice bot Talks to customers itself: answers or places calls A line or SIP trunk for the bot

Many teams start with analytics: it does not change a single conversation but immediately shows what is happening on the line. The bot comes second. More on both modes on the speech analytics and AI calls pages.

Connecting A1 VATS

A1 VATS is a cloud PBX used by many companies in Belarus. The company administrator connects it in the console, without contacting support.

  1. Connect the account. In telephony settings, choose A1 VATS and enter your account access details.
  2. Check the employees. The employee list is pulled from the PBX automatically.
  3. Enable recording and analysis. Transcription and AI scoring are switched on per employee, so you can start with one team.
  4. Import history. Past calls are imported in the background with progress shown on screen. The PBX limits how often requests can be made, so a large history does not load instantly.

From then on new calls appear in the log automatically, and a call in progress is marked "Active". Event intake and transcription have been tested on the real A1 platform; the console itself runs on our test environment as part of the pilot.

Tip: start with one team, such as reception or sales. You will build the checklist faster, compare AI scores with manual ones and learn which alerts you actually need before switching analysis on for the whole company.

Connecting Asterisk, FreeSWITCH and other PBXs

For on-site PBXs there is a SIP connection kit:

  • SIP over TLS and SRTP — signaling and voice are encrypted;
  • ready-made settings for Asterisk, FreeSWITCH and 3CX that your administrator copies into the configuration;
  • automatic recording import from a folder or over SFTP if you only need analytics;
  • a password rotation script for SIP accounts.

The kit has been tested with a real call in our test environment. On each new customer PBX the connection is checked separately at the start of the pilot: different systems have different versions and settings.

What to know about Asterisk

Asterisk connects for receiving and recording calls, but transferring a conversation from the bot to a live agent is not supported on Asterisk at the moment. If handoff is critical for your scenario, we discuss a FreeSWITCH-based setup during the pilot, where transfer has been tested with real calls.

The voice bot on your number

The bot works on the line you assign to it. Your number and carrier stay yours. For outbound campaigns the bot uses your trunk, so customers see your familiar company number. On our test environment, outbound bot calls through the A1 trunk and FreeSWITCH have been tested with real calls.

The console shows all lines on one page: number, owner and three switches — store calls, transcribe, analyze with AI. You can turn on the bot and analytics on one line and leave another untouched.

What your IT team needs to provide

A short list is usually enough:

  1. Administrator access to the PBX or the VATS account.
  2. A decision on which line or number the bot will use.
  3. Network access for the SIP connection, or a folder with recordings for analytics.
  4. Access to the CRM for writing results: Bitrix24 or amoCRM are connected in the console.
  5. If you want events in your own systems, an address for webhooks.

If data must not leave your perimeter, use an installation on your own servers instead of the cloud — see on-premise.

Standby telephony node

If continuity matters, you can specify a standby node in the PBX settings. The system checks the primary node every few seconds and routes new calls through the standby if it fails. Calls in progress on the failed node are not preserved — an honest limit of any failover setup. The scenario has been tested on a local setup with two simulated nodes; on your infrastructure it is worth checking during the pilot.

What goes into the CRM

After each analyzed call, the call summary, outcome and score are written to the customer record. The customer is matched by phone number. In Bitrix24 and amoCRM this is configured in the console; for 1C there is a pilot on a test database where each conversation creates a "Phone call" document. Other systems receive events through signed webhooks.

There is no CRM lookup before the call: results are written after the conversation.

Events in your own systems

If you need more than the CRM, the console sends events via webhooks: call analyzed, lead scored, alert triggered, handoff requested, campaign call completed. Each request is signed with a secret so your system can verify it came from us. If your server does not respond, delivery is retried several times with growing pauses, and a failed delivery can be resent manually. You can receive events in n8n and similar automation tools through a webhook.

Access and security

  • Personal SIP passwords. Each agent has their own account; a password can be changed without touching anyone else's.
  • API keys. Created and revoked by an administrator; the full key is shown only once.
  • Audit log. You can see who changed what in telephony and integration settings.
  • Data masking. Phone numbers, e-mail addresses and card numbers can be hidden in transcripts before they reach the AI.

Pilot sequence

  1. Connect telephony and check a test call.
  2. Turn on analytics for some lines.
  3. Set up the bot scenario and test it in the console.
  4. Switch the bot on for part of your real calls.

Timing depends on your PBX and what needs connecting, so we put together the pilot plan after a demo.

FAQ

Do I need to change my number or carrier?

No. The AI connects to your PBX or SIP trunk; your number and carrier stay the same.

Can I connect A1 VATS myself?

Yes. The company administrator connects the account in the console, employees are pulled in automatically and analysis is enabled per employee.

Does handoff to an agent work on Asterisk?

Not at the moment. Asterisk connects for receiving and recording calls; bot-to-agent transfer has been tested on FreeSWITCH.

What does the IT team need to provide?

PBX access, a line for the bot, network access for SIP or a folder with recordings, and CRM access. We prepare the exact list during the pilot.

Can I receive call events in my own system?

Yes, through signed webhooks. If your server does not respond, delivery is retried automatically.

How long does the connection take?

It depends on your PBX and the set of integrations. We give a timeline after the demo, once the telephony setup is clear.