Bitrix24-MTT Integration: VPBX Setup and Auto-Callback

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Bitrix24-MTT Integration: VPBX Setup and Auto-Callback
Medium
~1-2 weeks
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Are you losing incoming calls without CRM integration? Our Bitrix24 MTT integration includes professional MTT VPBX setup, call logging, recording, and auto-callback for missed calls. With over 10 years of Bitrix24 experience, we guarantee stable integration even under high load (up to 10,000 calls per day). The handler is written in PHP 8.1+ using Bitrix ORM, ensuring performance. Minimal project cost starts at $500. Get a consultation on your project — we will assess complexity and timeline.

MTT VPBX API: What's Available

MTT provides an HTTP API for managing virtual PBX and receiving call events. According to official MTT API documentation, key events for Bitrix24 integration:

MTT Event Description Action in Bitrix24
incoming_call Incoming call telephony.externalcall.register
answer_call Operator answered Update call status
end_call Call ended telephony.externalcall.finish
miss_call Missed call telephony.externalcall.finish (STATUS_CODE=304)

MTT sends notifications to a configured URL as POST JSON. Setup in MTT control panel: PBX Settings → Integrations → HTTP Notifications.

Handler Scheme

The handler for MTT works similarly to other VPBX integrations with one important detail: MTT delivers events in real time without delivery guarantees. Idempotency must be implemented — reprocessing an event should not create duplicates in Bitrix24.

PHP idempotency example
$cacheKey = 'mtt_call_' . $event['call_id'] . '_' . $event['type'];
if ($cache->exists($cacheKey)) {
    // Already processed this event
    http_response_code(200);
    exit;
}
$cache->set($cacheKey, 1, 3600); // TTL 1 hour
// Process event

Extension Mapping

MTT VPBX identifies employees by internal numbers (extension). In Bitrix24 — by USER_ID. The mapping table is stored in the handler configuration or database.

A special case is group calling, where MTT dispatches the call to multiple operators simultaneously. In the answer_call event, MTT indicates who actually answered. Use this extension for USER_ID in Bitrix24, not the initially called one.

Call Recording: Download via API

MTT stores recordings on its servers. Access via API:

GET https://api.mtt.ru/v1/records/{call_id}
Authorization: Bearer {api_key}

Response contains a URL to download the file. The file is available for 90 days by default (configurable in the plan).

After receiving the file — standard upload to Bitrix24 via telephony.externalCall.attachRecord.

Setting Up Multiple MTT Numbers

Companies with multiple local numbers on MTT can route all numbers to one operator group or split by departments. In the Bitrix24 integration, correct identification of LINE_NUMBER is important — Bitrix24 uses it to determine which line the call belongs to.

MTT passes the to_number field in the event — the number that was called. The handler determines LINE_NUMBER based on this field:

$lineMap = [
    '+74991234567' => '100', // Sales department line
    '+74997654321' => '101', // Support line
];
$lineNumber = $lineMap[$event['to_number']] ?? '100';

How to Set Up MTT Integration with Bitrix24 in 5 Steps?

  1. Analyze current architecture — study number scheme, load, auto-callback requirements.
  2. Design the handler — define number mapping, event processing scenarios, idempotency.
  3. Develop in PHP 8.1+ with Bitrix ORM, covered by unit tests.
  4. Test — simulate calls, verify data correctness, load test up to 100 concurrent calls.
  5. Deploy to production and train administrators.

Case Study: Trade Automation Company

We had a client with 15 managers and two groups: inbound sales and tech support. MTT was the primary operator with two local numbers. Our Bitrix24 MTT integration requirement: calls to the sales number go to sales managers, calls to support go to tech specialists. Missed calls — automatic callback via Bitrix24 robots.

Challenge: MTT reported a missed call with the miss_call event only if no operator answered. If an operator picked up and immediately hung up (less than 3 seconds), the event came as end_call with duration < 3. We implemented duration check: if duration < 3 — mark as missed in Bitrix24.

For auto-callback on missed calls, we configured a robot in Bitrix24: create a new lead and run a chain with a callback task in 10 minutes. The entire project took 5 business days.

Why MTT Integration Pays Off Faster Than Alternatives?

MTT offers competitive rates and wide number geography, but event delivery reliability is lower than Mango Telecom, which uses queues with acknowledgment. Compare:

Operator Delivery Guarantee API for Recording Recording Storage Cost per Minute
MTT No guarantee HTTP API 90 days Low
Mango Telecom Yes (queues) REST + Websocket 180 days Medium

With MTT, we always implement idempotency to prevent duplicates even with repeated sending. This reduces CRM load by 30% and cuts call processing time in half compared to solutions without idempotency. The integration pays for itself in 2-3 months on average by reducing missed calls and automation. Average telecom savings are 20,000-30,000 rubles per month for companies with 10+ employees.

What's Included

  • Audit of current MTT and Bitrix24 configuration
  • Development of handler with idempotency and number mapping
  • Setup of auto-callback via Bitrix24 robots
  • Documentation of integration scheme and code
  • Load testing up to 100 concurrent calls
  • Administrator training on the new system
  • One month of technical support after deployment

Contact us to discuss your project — we will assess complexity within one business day. Get a consultation on MTT integration with Bitrix24 right now.

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.

Provider Real Benefit Pitfalls
Bitrix24.Telephony Built-in, setup via portal UI Tied to B24 tariff, more expensive at scale
Sipuni Native module for B24, quick integration Limited number geography
Mango Office Powerful analytics, speech analytics included Above-market price
UIS (CoMagic) Call tracking + PBX all-in-one Complex admin panel
Zadarma 70+ countries, budget-friendly API sometimes slow, support slow
MegaFon PBX FMC — mobile as office SIP setup via support, not UI
MTT Flexible tariffs, good API No native module for B24
Rostelecom Stability, government sector Slow setup, bureaucracy

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.

Which codecs to choose for stable operation? G.711 (A-law/µ-law) — standard codec, bandwidth 87 kbit/s per line. G.729 — compressed codec, 31 kbit/s. If bandwidth is limited, G.729 is preferred, but not all providers support it. Also possible G.722 (HD voice) — requires 64 kbit/s and PBX support.

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

Task Timeframe
Cloud PBX + basic routing 1–2 days
SIP trunk (if provider releases data quickly) 1–3 days
Routing + IVR 2–5 days
CRM integration: robots, triggers, cards 3–5 days
Callback widget 1 day
Comprehensive setup with audit 2–3 weeks

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.