Setting up SIP telephony in Bitrix24

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Configuring SIP Telephony in Bitrix24

Bitrix24 can make and receive calls without installing a separate PBX — telephony is embedded in the platform via the voip module. But "embedded" doesn't mean "works out of the box": you need to connect a provider, configure call routing, and distribute numbers among employees.

How SIP telephony works in Bitrix24

Bitrix24 acts as a SIP client or manages its own cloud PBX. An incoming call follows this path:

  1. The caller dials a number.
  2. The telecom operator (SIP provider) receives the call and transmits it via SIP protocol to Bitrix24's PBX.
  3. Bitrix24 determines where to route the call (by queue, by responsible manager in CRM, by schedule).
  4. The manager answers via browser phone (WebRTC), desktop app, or physical SIP phone.

The entire process occurs in Bitrix24's telephony module. Settings are stored in b_voip_* tables and managed via CRM → Telephony section.

What is needed for setup

1. Phone number. Either rented directly from Bitrix24 (via "Telephony" section → "Connect number"), or an existing operator's number is transferred via SIP trunk connection.

2. Choosing a connection scheme:

Scheme Description When suitable
Bitrix24 cloud PBX Number at Bitrix, routing inside platform No own operator
SIP trunk Own operator number, forwarding to Bitrix Existing number
SIP connection with external PBX Asterisk/FreePBX as intermediary Complex phone infrastructure

3. Configuring incoming routing. In "Telephony" → "Settings" section, configure call processing scheme:

  • Queue: call goes round-robin among available employees.
  • Simultaneous ring to all.
  • Redirect to specific employee.
  • Voice menu (IVR): press 1 — sales, press 2 — support.

4. Business hours. Configured in number parameters: calls outside business hours go to answering machine or redirect to another number.

Integrating calls with CRM

On incoming call, Bitrix24 searches the number in CRM database. If contact or company found — call is linked to it. If not — new lead is created (if auto-creation is enabled in telephony settings).

After call completion, a "Call" type task is created in the CRM card. Call recording is attached to the task — provided recording is enabled in settings.

For outgoing calls via CRM, use the phone button in contact or deal card. Bitrix24 initiates the call via SIP.

Common setup issues

One-way audio — usually NAT problem. SIP signaling passes but RTP media stream is blocked by router. Solved by configuring STUN server in SIP account parameters or opening UDP port range 10000-20000 on router.

Calls not arriving — check SIP account registration status in Bitrix24 ("Telephony" → "SIP connector" section). Often the cause is incorrect password or IP blocking by provider.

Echo in handset — acoustic suppression problem. In browser client solved by WebRTC settings; on physical phone — by phone's hardware settings.

Basic setup of telephony with one number and simple routing takes 2-4 hours.