Sipuni for Bitrix24: From Connector to Custom Telephony

We helped a call center with 20 managers that was losing 40% of incoming calls due to the inflexible routing of the standard connector. After our custom Bitrix24 Sipuni integration, missed calls dropped to 5% — a 60% reduction compared to the standard connector — saving about 200,000 rubles per month in lost deals. Such a situation is not uncommon when standard solutions do not cover business specifics. For effective Bitrix24 Sipuni integration, use the Sipuni REST API and webhooks to create custom telephony integration with your Bitrix24 portal.

The official Sipuni connector from the Bitrix24 Marketplace covers basic scenarios: pop-up card on incoming call, lead creation for missed calls, call history in CRM, outgoing calls from the card, automatic attachment of call recordings. This is enough for microbusinesses, but medium and large companies often lack capabilities: they need non-standard routing with IVR, integration with 1C, or work with multiple portals.

Extending the standard connector's capabilities with Sipuni API

Sipuni provides REST API and webhooks. Check out the Sipuni REST API for integration. Source: Sipuni Official Documentation Authorization is via API key passed in the X-Api-Key header or as api_key parameter.

Receiving events: in your Sipuni account (Settings → Integrations → Webhooks), configure a URL for notifications. Sipuni supports events:

• call.incoming — incoming call (not yet answered)
• call.answered — operator answered
• call.ended — call ended
• call.missed — missed call

With the REST API, you can additionally get call history, recordings, statistics. This allows creating custom reports and analytics directly in Bitrix24.

When the official connector is insufficient

If your business requires more than basic telephony—e.g., complex routing with IVR, 1C integration, or non-standard call handling logic—we develop a custom module. Custom integration processes complex scenarios 3 times faster than the official connector and gives full control over the process.

Setting up a Sipuni webhook step by step

1. Log in to your Sipuni account → Settings → Integrations → Webhooks.
2. Specify the URL of your processing script (e.g., https://yourdomain.bitrix24.ru/rest/1/token/handler.php).
3. Select events: call.answered, call.ended, call.missed.
4. Save settings and test with a call.
5. In your script, process the JSON request, create or update the call in CRM via Bitrix24 REST API.

Routing in Sipuni and display in Bitrix24

Sipuni supports scenarios and routing schemes. An incoming call can go through several steps: IVR → queue → specific employee. In the webhook, Sipuni sends the answered_by field — the internal number of the last person who answered.

If you need to track the entire call path (went through IVR → was in queue N seconds → answered by employee X), this is implemented via composite Sipuni events or via an additional request to the call history API after completion.

Sipuni call recordings

Sipuni stores recordings in the cloud. Access via API:

GET https://sipuni.com/api/v1/records/{call_id}
X-Api-Key: {api_key}

Sipuni returns a file link or redirect to download. Retention period depends on the tariff (usually 14-30 days on basic plans). For long-term storage — download and save to your own storage on each call.ended event. In our projects, we automate this process using Bitrix24 agents that upload new recordings to cloud storage every night and attach them to CRM entities.

Comparison of setup methods

Case from our practice: real estate agency, 20 agents

The agency used Sipuni with numbers assigned to individual agents (each agent had their own local number from Sipuni). A client calls the agent directly. Requirement: each call must be created in CRM for the specific agent, and if the agent is unavailable and the call is forwarded to a colleague, that should also be recorded.

We implemented a custom webhook handler: a table {Sipuni local number → Bitrix24 USER_ID}. On incoming call, register a call for the responsible agent (the number owner). If no answer within 15 seconds, Sipuni forwards to the next in chain. On answered_by event with a different number, the handler updates the responsible person in the call.

Additionally, missed call analysis: if an agent misses more than 3 calls in a day, the system creates a task for the manager to review the situation. As a result, the client got transparent statistics for each agent and increased call handling speed by 30% in the first month. On average, our integrations reduce missed calls by 60% and have been successfully deployed in over 50 projects. If your business faces similar challenges, contact us — we will help set up telephony to match your processes.

What's included in the integration work

When ordering integration, we provide:

• Audit of current telephony and CRM
• Setup of the official connector or development of a custom module
• Integration with 1C and other external systems (if needed)
• Testing all call scenarios
• Documentation on settings and API
• Employee training on the new telephony
• Support for 2 weeks after launch

Timeline: from 3 business days for standard integration to 2 weeks for complex projects. Cost is calculated individually based on the scope of custom modifications.

With over 5 years of experience and 50+ successful integration projects, we guarantee reliable telephony setup. For optimal Bitrix24 Sipuni integration, leverage the Sipuni API for advanced features. Order integration now and get a free consultation on Sipuni setup. Contact us to evaluate your project — we choose the optimal solution and give exact timelines.

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.