Turnkey setup of call transcription in Bitrix24

We often encounter the situation where call recording is enabled for everyone, but no one manually listens to hour-long conversations. That's where our transcription setup turns audio into searchable, analyzable text and pushes it into CRM without listening. We offer a turnkey solution: from cloud activation to on-premise integration via external Speech-to-Text services. We assess your project in one day and complete the setup within 2-5 business days. Our extensive experience with Bitrix24 and dozens of implemented projects ensure stable recognition even with suboptimal recording quality.

How call transcription actually works

Bitrix24 uses the built-in Bitrix24 Contact Center module together with a cloud speech recognition service. After a call ends, the recording is sent via the telephony module to the recognition server — the result is returned as text and saved to the CRM entity card (lead, deal, contact).

Technically, the process looks like this:

1. The call ends, the recording is saved in b_voip_call linked to CALL_ID.
2. The transcription task enters the handler queue.
3. The Speech-to-Text service (Yandex SpeechKit or the built-in Bitrix engine) receives the audio file.
4. The text is returned and written to the TRANSCRIPT field of the call record.
5. A "Call transcription" block with speaker separation appears in the CRM card.

What is required for activation

Transcription is available only on Professional and Enterprise plans (Bitrix24 cloud). For on-premise, you need an external Speech-to-Text provider and the REST API method telephony.externalcall.finish.

Activation steps:

1. CRM → Telephony → Settings → "Transcription" tab.
2. Enable the toggle "Transcribe calls".
3. Select recognition language (Russian, English — list depends on plan).
4. Specify which phone numbers to enable transcription for.
5. Set up access rights: who sees transcriptions (manager, employee themselves, everyone).

After activation, every new call longer than 10 seconds will be automatically transcribed. Old recordings — only via manual trigger or REST script.

Limitations and common issues

Recognition quality strongly depends on recording quality. If the line has noise, echo, or low bitrate, accuracy drops to 60-70%. Minimum recommended bitrate: 8 kHz mono, preferably 16 kHz.

Speaker separation (diarization) works only with stereo recordings where operator and client are on separate channels. Mono recordings produce a single text block without speaker labels.

Quotas. Each plan has a monthly limit of transcription minutes. When exceeded, transcription stops until the next billing period. Limits are visible under "Tariffs" → "Usage".

On-premise version. The built-in engine is not available. Integration with Yandex SpeechKit or Google Speech-to-Text is implemented via the OnVoipCallEnd event handler, calling an external API and writing the result back to Bitrix via CRest::call('telephony.externalcall.finish', [...]).

What to do with the transcribed text

Transcripts can be used in robots and business processes: trigger "Call completed" → condition on transcript text → action (change stage, assign task, notify). The transcript field is available as the {=Document.TRANSCRIPT} variable in the business process designer.

Search by transcript texts works via standard CRM search — the transcript is indexed in b_search_content along with other fields. Time savings for call analysis reach 70%.

How to improve recognition accuracy to 95%?

Accuracy directly depends on three factors: recording bitrate (recommended 16 kHz mono or stereo), noise level (noise suppression on telephony), and speech clarity. With stereo recording on separate channels and 16 kHz bitrate, accuracy exceeds 90%. Mono recordings with noise drop to 60-70%. We recommend auditing your current telephony before setup. Read more about quality parameters in Bitrix24 documentation.

What is included in transcription setup

Have questions? Contact us — we will assess your project for free and recommend the optimal solution. Order call transcription setup in Bitrix24 and get a fully automated process for conversation analysis. With guaranteed recognition quality and certified expertise, you save up to 40 hours per month per manager — equivalent to $800+ in productivity gains.

Turnkey Telephony Setup: From SIP to CRM

SIP trunk not registering — the first problem when connecting telephony to Bitrix24. The client sends provider data, you enter it in "Telephony → Telephony Settings → SIP Connector", and the status hangs "Waiting for registration". Ports 5060/5061 for SIP and range 10000–20000 for RTP are blocked on the firewall — classic. Or codec G.729 is not supported, and G.711 consumes 87 kbit/s per line. For 20 simultaneous conversations — count yourself. We connect and configure telephony so that calls are not lost, routing works according to business logic, and CRM receives data automatically.

SIP (Session Initiation Protocol) — standard for voice communication over the internet. Our specialists have 5+ years of experience integrating with Bitrix24. Warranty on configured solutions — 1 year. Contact us for a free audit of your current infrastructure — we often find blind spots: missed calls without tasks, managers with status "On line" and zero calls, IVR branches that no one reaches.

How to Avoid Failures in SIP Registration and Connection?

Improper NAT, closed ports, or incompatible codecs — 90% of failures. We check codec support on the operator's side, configure keepalive and backup SIP servers. After setup, registration stays stable for months.

You already have a contract with an operator, and changing numbers is not an option. We connect to Bitrix24 via SIP connector. For stable operation, you need a channel: at least 100 kbit/s per line for G.729, 87 kbit/s for G.711a.

Typical tasks:

• City numbers from any operator — preserved when switching to B24
• 8-800 numbers for inbound — routing via CRM
• Multi-line numbers with department distribution via IVR
• Analog lines via Grandstream or Linksys VoIP gateways

Where it usually gets stuck:

• SIP/RTP ports are closed at the provider or router level
• NAT not forwarded correctly — one-way audio
• Different codecs on the PBX and Bitrix side — no audio at all

We solve these problems in 1–3 days, not weeks.

Step-by-step instructions for connecting via SIP trunk:

1. Check SIP (5060/5061) and RTP (10000–20000) ports are open on the firewall.
2. Agree on codecs: G.711 or G.729 — test both options.
3. Configure SIP connector in Bitrix24 admin: login, password, provider domain/IP.
4. Enable keepalive (30–60 second interval) for stable registration.
5. Test two-way audio with a test call.
6. Configure backup SIP server in case of primary failure.

Cloud PBX or SIP Trunk: Which to Choose?

Cloud PBX requires no hardware — set up in 1–2 days. SIP trunk retains your numbers and allows a hybrid scheme with an on-premise PBX. Cloud PBX is configured 2-3 times faster and saves up to 40% of budget compared to maintaining your own infrastructure. Compare providers based on real cases.

Why Do You Need Call Recording and How to Set It Up?

All incoming and outgoing calls are recorded automatically. Storage — Bitrix24 cloud (tariff limited) or your own server via the voximplant module. Each recording is attached to a CRM card: contact, deal, lead. Listening — from the deal timeline.

Control tools:

• Recording notification — mandatory under 152-FZ, configured in IVR
• AI transcription — speech recognition via built-in CoPilot module or external service. Converts calls to text, searchable by keywords
• Sentiment analysis — system flags negative calls. Not a replacement for listening, but filters 90% of routine
• Selective recording — only incoming, only specific departments
• Automatic duration control — calls shorter than 15 seconds almost always indicate a problem: dropped, not reached, manager didn't pick up

Manager sees summary reports: number of calls per manager, average duration, missed calls without callbacks. The last point is the most painful: a missed call without a callback task equals a lost client.

Typical Routing Mistakes

Routing is the most fragile part. Set up "in order", but the manager logged in and went to lunch — the call hangs for 40 seconds and drops. Or IVR with five nesting levels — the client hangs up on the third. We use working distribution scenarios:

• By CRM responsible — number found in database, call goes directly to manager
• In order — even load among operators with "On line" status
• By departments via IVR — voice menu, max two levels. More means lost calls
• By time — working/non-working hours, weekends
• By region — detected by number code, distributed to local managers

On incoming call, the system performs: Number found in CRM:

• Route to responsible manager
• Pop-up card: name, company, open deals, contact history
• If manager doesn't answer within 15 seconds — forward to colleague, then to supervisor

Number not found:

• Automatically creates a lead in CRM (setting Telephony → CRM Integration → Create lead for unknown number)
• Call distributed by queue
• All information recorded for later processing

How to Automate Work with CRM Integration?

Pop-up card on incoming call shows:

• Name, company, position
• Current deals with statuses
• Last 5 interactions — calls, emails, chats
• Customer comments
• Buttons: "Create deal", "Set task", "Transfer call"

Automation via business processes is configured in "CRM → Robots and triggers":

• Missed call → lead + callback task for responsible
• Call end → record activity in deal timeline
• Missed without callback within 30 minutes → notify supervisor
• Incoming from client with overdue deal → escalation

Analytics available in "Telephony → Balance and statistics" and CRM reports:

• Number of calls by manager, department, direction
• Average duration and hold time
• Missed call percentage with weekly trend
• Call-to-deal conversion — most important metric for sales

How to Increase Conversion with Callback, Auto-dialer, and Voice Menu?

"Call me back" widget on website — one field (phone), nothing extra. The system calls the manager first, then the client. Each request automatically creates a lead in CRM. Anti-spam: limit requests from one number. A/B testing placement — widget bottom right converts better than left, but depends on design. Callback reduces missed calls by up to 30% compared to manual dialing.

Auto-dialer:

• Dial segment from CRM — manager connects after client answers
• Robotic dial with voice message — order confirmations, appointment reminders
• IVR scenarios: "press 1 to confirm"
• Scheduled with time zone awareness — module checks region by number code

Building a voice menu:

• Welcome message — company name, recording notice
• Department selection via DTMF — max 4–5 items
• Music on hold with queue position
• Non-working hours — voicemail or informational message

Recording by professional voice actor — worthwhile for greeting and main menu. Deeper levels — speech synthesis, easier to update.

What's Included in Telephony Setup?

Deliverables after project completion:

• Configured SIP registration (cloud PBX or SIP trunk turnkey)
• Working call routing (by CRM, queue, IVR)
• Call recording with cloud or server storage
• CRM integration: pop-up cards, auto lead creation, business processes
• Callback widget on website
• Voice menu (IVR) with voice actor recording or synthesis
• Configuration documentation and staff training (2 sessions)
• Technical support for one month after launch

Everything included in the comprehensive setup — no hidden extras. Order telephony setup and get a ready-made infrastructure for working with clients.

Setup Timeline and Pricing

Pricing is calculated individually — depends on number of lines, routing complexity, and required integrations. Average budget for setup ranges from several thousand to tens of thousands in local currency depending on scope. We'll evaluate your project in 1 day. Start with a free audit of your current infrastructure — we often find blind spots: missed calls without tasks, managers with "On line" status and zero calls, IVR branches that no one reaches. Contact us for a consultation — we'll offer the optimal solution for your budget and business processes.