When an incoming call to Bitrix24 doesn't recognize the client's number, or drops after 30 seconds — that's a typical story from the first SIP telephony setup. If managers hear themselves with a delay, it's one-way audio due to incorrect NAT. A company connects a SIP trunk, but without proper routing, calls fall into a black hole. We break down such cases every month. For a chain of 15 coffee shops, we set up an IVR with zone-based distribution, cutting phone order processing time by 30%. Our team has completed over 120 SIP connections for retail, logistics, and services. Proper SIP telephony setup in Bitrix24 reduces communication costs by up to 40% and boosts call conversion. Contact us for an audit of your infrastructure.
How SIP Telephony Works in Bitrix24
Bitrix24 acts as a SIP client or manages a cloud PBX. The SIP protocol transmits voice over IP. An incoming call follows this path:
- Subscriber dials the number.
- The carrier (SIP provider) accepts the call and forwards it to the Bitrix24 PBX.
- Bitrix24 determines the route: by queue, responsible person in CRM, or schedule.
- The manager answers via a browser phone (WebRTC), desktop app, or physical SIP phone.
The entire process is managed by the telephony module. Settings are stored in the b_voip_* tables and edited through CRM → Telephony.
What Is Needed to Connect SIP Telephony?
A phone number. You can either rent one directly from Bitrix24 (via Telephony → Connect number) or port an existing operator number by connecting a SIP trunk.
Choice of connection scheme. Compare three options:
| Parameter | Bitrix24 Cloud PBX | SIP Trunk | External PBX (Asterisk) |
|---|---|---|---|
| Number | Rent from Bitrix24 | Your existing | Your existing |
| Routing | Within the platform | Through Bitrix24 settings | Configured on PBX |
| Connection speed | 1-2 hours | 2-4 hours | 4-8 hours |
| Flexibility | Limited | High | Maximum |
| Cost | Fixed rental | Provider fee + integration | PBX cost + integration |
For most companies, the SIP trunk is optimal: it processes high call volumes faster than a cloud PBX and doesn't lock you into a specific provider.
Inbound routing setup. In Telephony → Settings, configure call handling: queue, simultaneous ring to all, forwarding to a specific employee, or IVR (voice menu).
How to Connect a SIP Trunk to Bitrix24?
Connecting a SIP trunk is one of the most common requests. Step-by-step guide:
- Obtain login, password, and server address from your SIP provider.
- In Telephony → SIP Connector, select 'Connect SIP trunk'.
- Enter the account details. Specify NAT parameters if employees work remotely.
- Configure inbound call routing (IVR, queue, business hours).
- Verify registration: status should change to 'Registered'.
According to official Bitrix24 documentation, for stable operation you need to open UDP ports 10000-20000 for RTP and port 5060 for SIP. After successful registration, make a test call. If audio is one-way, check the STUN server settings.
Why Calls Don't Go Through After Setup?
Three most common reasons:
- One-way audio — usually a NAT issue. SIP signaling passes, but the RTP media stream is blocked by the router. Solved by configuring a STUN server in the SIP account settings or opening the UDP 10000-20000 port range on the router.
- Calls don't arrive — check the SIP account registration status in Telephony → SIP Connector. Often the cause is an incorrect password or IP block on the provider's side.
- Echo on the line — an acoustic suppression issue. In the browser client, it's resolved by WebRTC settings; on a physical phone, by hardware handset settings.
Resolving one-way audio: verify that the SIP account settings specify a correct STUN server (e.g., stun.l.google.com). Ensure the firewall isn't blocking UDP 10000-20000. If the issue persists, configure port forwarding on the router or use a SIP provider that supports ICE.
What Problems Arise When Setting Up SIP?
- One-way audio: open UDP ports 10000-20000, configure STUN server. Check that the firewall isn't blocking the media stream.
- Calls don't reach: ensure the SIP account is registered. Often the cause is an incorrect password or IP block by the provider.
- Echo on the line: in the browser client, check WebRTC settings; on a physical phone, handset quality.
Integrating Calls with CRM
On an incoming call, Bitrix24 looks up the number in the CRM database. If a contact or company is found, the call is attached to it. If not, a new lead is created (if auto-lead creation is enabled in telephony settings). After the call ends, a 'Call' activity is created in the CRM record. The call recording is attached to the activity if recording is enabled. For outgoing calls via CRM, use the phone icon in the contact or deal card. Bitrix24 initiates the call through SIP.
What Does a Turnkey Setup Include?
The SIP telephony setup process includes five stages:
| Stage | Duration | Description |
|---|---|---|
| Analytics | 1-2 hours | Audit current infrastructure, select provider |
| Connection | 1-3 hours | Register SIP trunk, configure basic parameters |
| Routing | 1-2 hours | Set up IVR, queues, business hours |
| CRM Integration | 1-2 hours | Link calls to contacts, auto lead creation |
| Testing & Training | 1-2 hours | Verify call quality, resolve NAT issues |
Our offering includes audit, connection, routing, CRM integration, and testing. Basic setup with one number and simple routing takes 2-4 hours. If IVR or custom logic is needed, the timeframe extends to 8-12 hours. Cost is calculated individually based on complexity. Communication savings after proper integration can reach 40%. Order SIP telephony setup — get stable communication and automation within 1 day. The investment typically pays for itself in 2-3 months. Get a consultation for your project — we'll assess the possibilities.