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.
FreePBX/Asterisk can provide a flexible architecture for different enterprise telephony scenarios through its open-source approach and SIP-based integration options.
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.
For Linux-based systems, stability and security should be considered together with the correct version, updates, access policies, backup, monitoring and resource sizing.
Concurrent call capacity is sized according to CPU/RAM, codec/transcoding, recording, network bandwidth, SIP Trunk channel count and call scenarios.
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-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.
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.
We provide end-to-end expertise when building your organization’s communications backbone.
Open-source FreePBX/Asterisk-based IP PBX installation on your physical server (on-premises) or virtual environment.
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.
Migration of legacy analog lines to IP infrastructure and integration with Telekom/Turkcell SIP Trunk services.
Multi-level touch-tone voice-response scenarios that greet customers with studio-quality recordings.
Queueing high volumes of incoming calls, hold music and position announcements for callers.
Configuration of compatibility, codec and security settings for SIP-based mobile and desktop clients such as Zoiper, MicroSIP, Bria and Linphone.
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.
Assessment of unreliable or noisy analog systems and migration to next-generation IP (VoIP) infrastructure.
Continue with the relevant technical guide, product/licensing page or project scenario.