FusionPBX is a full-featured, web-managed PBX built on FreeSWITCH. This Azure image arrives preinstalled and tuned for cloud workloads so you can register phones, connect SIP trunks, and handle inbound/outbound calling with minimal setup. It’s a clean, repeatable baseline for pilots, labs, and production voice services.

Who it’s for

Telecom/VoIP engineers and MSPs building SIP voice solutions on Azure
Contact centers and UC providers needing IVR, ACD, and conferencing
Developers/DevOps prototyping RTC features with a reliable SIP/WebRTC backend
Enterprises standardizing on a managed PBX with predictable updates and support

Standing up a PBX from scratch is time-consuming and error-prone. Misconfigured codecs, NAT, RTP ranges, firewalls, or TLS cause call failures and poor media quality. This offer removes guesswork with an Azure-ready baseline, sensible defaults, and guidance for secure networking so you can focus on call flows and user experience.

Why choose this image on Azure

Azure-Optimized from First Boot: cloud-init and Azure Linux Agent are preconfigured for VM extensions, diagnostics, and lifecycle automation.
Faster Time to Dial Tone: FusionPBX and FreeSWITCH are preinstalled with sensible defaults, cutting hours of build and hardening.
Built for Scale and Resilience: Works with Azure Load Balancer, Availability Zones, and managed disks.
Security-First Posture: TLS/SRTP capable, OS hardening guidance, automatic security updates available, and CIS-aligned recommendations.
Operational Confidence: Log rotation, system services, and upgrade paths documented to keep you current without surprises.

What’s included

FusionPBX web UI (multi-tenant capable)
FreeSWITCH media engine with common codecs enabled
Azure Linux Agent + cloud-init
Sensible OS/network defaults for cloud environments

Features at a glance

Extensions and Devices: SIP endpoints, provisioning patterns, BLF, voicemail, ring groups
Call Control: IVR, queues, time conditions, call recording, failover routing
Connectivity: SIP trunks, TLS/SRTP support, NAT-friendly defaults
Observability: CDRs, logs, and real-time status via web UI
Multi-Tenant: Optional multi-tenant configuration for MSPs and large orgs

Networking: ports (NSG guidance)

SIP: 5060 UDP/TCP, 5061 TCP for TLS
RTP: UDP 16384–32768 (default FreeSWITCH RTP range)
Web UI: 80/tcp (optional), 443/tcp
SSH: 22/tcp (restrict to admin IPs)

Harden NSGs to only the sources you control. For public services, pair with Azure DDoS Protection and Azure Firewall or an SBC as appropriate.

Getting Started

Deploy the VM

Launch your FusionPBX instance from the Azure Marketplace into your preferred region, VNet, and subnet. Choose a VM size that matches your expected call volume.

Access the VM

Once deployment completes, connect via SSH using the admin account you specified during provisioning:

ssh -i <ssh-key> admin@<public-ip>

Retrieve FusionPBX Credentials

The installer generates a unique username and password at first boot. Run the following command on the VM to display them:

sudo cat /opt/default-fusionpbx-creds.txt

Access the Web UI

Open a browser and navigate to:

https://<public-ip>

(Replace <public-ip> with your VM’s public IP address.) Log in with the admin username and the password retrieved in the previous step.

Complete Initial Setup

Follow the FusionPBX setup wizard to:

Add extensions and register phones
Connect your SIP trunk provider
Configure outbound and inbound routes
Test calls, voicemail, and IVR

Launch your FusionPBX on Azure instance now and move from setup to dial tone in minutes. Empower your teams with a reliable, scalable, and secure PBX solution that’s ready for production.

FusionPBX on Azure - Turn-Key FreeSWITCH PBX for SIP & WebRTCSolve DevOps

FusionPBX on Azure - Turn-Key FreeSWITCH PBX for SIP & WebRTC

Solve DevOps

FusionPBX on Azure - Turn-Key FreeSWITCH PBX for SIP & WebRTC

Solve DevOps

FusionPBX on Azure - Turn-Key FreeSWITCH PBX for SIP & WebRTC
Solve DevOps