Open-Source Licensing Model
The open-source components of FreePBX/Asterisk provide licensing flexibility; however, commercial modules, support, hosting, SIP carrier services and third-party product costs are evaluated separately for each project.
Why Choose Linux-Based FreePBX?
FreePBX/Asterisk can provide a flexible architecture for different enterprise telephony scenarios through its open-source approach and SIP-based integration options.
Linux Kernel Stability
For Linux-based systems, stability and security should be considered together with the correct version, updates, access policies, backup, monitoring and resource sizing.
Capacity & Concurrent Call Planning
Concurrent call capacity is sized according to CPU/RAM, codec/transcoding, recording, network bandwidth, SIP Trunk channel count and call scenarios.
IVR, Queue & Call Features
The scope of features such as IVR, call queues, business-hour rules, recording and reporting is determined by the FreePBX/Asterisk version, modules and project requirements.
SIP-Based Hardware Integration
SIP-compatible IP phones, FXS/FXO gateways and softphones can be integrated; actual compatibility should be verified according to device model, firmware, codec and vendor-specific features.
Fail2Ban & Cybersecurity
Controls such as Fail2Ban, firewall rules, strong credentials, management-access restrictions and, where possible, TLS/SRTP should be evaluated together to reduce the VoIP attack surface.
Our Professional IP PBX Solutions
We provide end-to-end expertise when building your organization’s communications backbone.
FreePBX (Asterisk) Kurulumu
Open-source FreePBX/Asterisk-based IP PBX installation on your physical server (on-premises) or virtual environment.
- Linux Infrastructure Configuration
- Extension Planning Based on Requirements
Cloud PBX
A FreePBX/Asterisk solution that can be hosted on a suitable cloud infrastructure instead of a physical PBX server. Availability depends on the selected provider, architecture and SLA.
- Hosting on Cloud Resources
- Redundant Internet & Business Continuity Planning
SIP Trunk & Number Porting
Migration of legacy analog lines to IP infrastructure and integration with Telekom/Turkcell SIP Trunk services.
- 0850 and Geographic Number Integration
- Multiple Concurrent Calls (Channels)
IVR & Professional Greeting
Multi-level touch-tone voice-response scenarios that greet customers with studio-quality recordings.
- Automatic Business-Hours / After-Hours Switching
- Keypad Routing to Departments
Call Center & Call Queue
Queueing high volumes of incoming calls, hold music and position announcements for callers.
- Live Call Monitoring & Whisper
- Automatic Recording of All Calls
Softphone (Evrensel Uyum)
Configuration of compatibility, codec and security settings for SIP-based mobile and desktop clients such as Zoiper, MicroSIP, Bria and Linphone.
- Mobile Application Setup
- Internal Calling Between Branches / Staff
IP Phones & FXS/FXO
SIP phones such as Cisco, Poly, Snom, Yealink and Gigaset, and FXS/FXO gateways such as AudioCodes, Dinstar and beroNet, can be integrated after project-specific compatibility checks.
- Hardware Supply & Installation
- Wireless Solutions for Factories
PBX Upgrade & Maintenance
Assessment of unreliable or noisy analog systems and migration to next-generation IP (VoIP) infrastructure.
- On-Site Network & PBX Assessment
- VoIP Gateway Integrations
Related Next Steps
Continue with the relevant technical guide, product/licensing page or project scenario.
Plan the IP PBX infrastructure together with firewall / cybersecurity, enterprise Wi-Fi, structured cabling and cloud infrastructure.
What should you define before deploying an IP PBX?
Define extensions, concurrent external calls, carrier connectivity, number porting and existing IP phone compatibility. Plan recording retention, access permissions, backups and communications during an outage separately. Assess the existing PBX and network before choosing a deployment or migration scope.