How to connect your PBX to AI: A1 VATS and Asterisk for speech analytics

What works today. Your PBX connects to speech analytics: we receive call recordings and events. The voice bot and AI chats are soon: get demo access and test them for your business.

You can connect your PBX for speech analytics. 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. Results of analyzed calls go to your CRM. The voice bot connects as the next step on the same PBX.

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

Two modes: analytics today, the voice bot as the next step

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

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

Start with analytics: it does not change a single conversation but immediately shows what is happening on the line. The bot is easy to add as a second step: get demo access to test it on your scenarios. More 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, which is all speech analytics needs. If your voice bot scenario hands calls to a live agent, choose FreeSWITCH: transfer there has been tested with real calls.

The voice bot: the next step

The bot works on the line you assign to it, and outbound calls use your trunk. Want early access to test the bot for your business? Get demo access.

The console already shows all lines on one page: number, owner and switches — store calls, transcribe, analyze with AI. Turn analytics on for the lines you need and leave the others 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. Network access for the SIP connection, or a folder with recordings for analytics.
3. For the voice bot, a decision on which line or number it will use (needed at the bot step).
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.

Results are written to the customer record right 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. Extend analytics to the remaining lines, set up the checklist and alerts.
4. The voice bot is added as a separate step on the same setup.

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

FAQ

Will I keep my number when connecting the PBX to speech analytics?

Yes. Analytics connects to your PBX or SIP trunk.

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.

How do I connect a voice bot to my PBX?

The bot gets a line or number on your PBX or SIP trunk, and outbound calls go through your trunk. Get demo access and test the bot on your scenarios.

Which PBX should I choose if the bot hands calls to an agent?

FreeSWITCH: bot-to-agent transfer has been tested on it with real calls. Asterisk works well for receiving and recording calls for analytics.

What does the IT team need to provide?

PBX access, network access for SIP or a folder with recordings, and CRM access. A line for the bot is needed at the bot step. 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 put together the exact plan and timeline after the demo, once we see your telephony setup.