Open Source VoIP Development & Support
Installation, customization, security hardening, and ongoing support for FreeSWITCH, Asterisk, Kamailio, OpenSIPS, FusionPBX, VICIdial, and the entire open-source telephony ecosystem.
From fresh installations to complex multi-node clusters - we make open-source telephony production-ready, scalable, and secure.
Core Telephony Engines We Master
The foundational frameworks that power every serious VoIP deployment. We install, configure, customize, and scale these engines for production workloads.
FreeSWITCH
The Enterprise PowerhouseBuilt specifically for high concurrency, stability, and enterprise-grade routing. Designed from the ground up to handle massive call volumes with native multi-tenant architecture support, making it the go-to engine for large contact centers and carrier-grade platforms.
Best for: Contact centers, carrier-grade platforms, multi-tenant PBX systems, large-scale IVR deployments.
Asterisk
The Industry PioneerThe oldest and most famous open-source communication framework. A massive toolkit to build PBX systems, IVRs, and VoIP gateways. Incredibly versatile with a massive community and thousands of available modules for every telephony need.
Best for: Business PBX systems, IVR platforms, VoIP gateways, small-to-medium call centers, unified communications.
YATE
The Lightweight PerformerA lightweight engine focused heavily on routing and scalability. Often used in carrier environments and mobile networks due to its exceptionally high performance and minimal resource footprint. Ideal for signaling-heavy workloads.
Best for: Carrier environments, mobile networks, high-frequency routing, signaling gateways, MVNO platforms.
SIP Proxies & Load Balancers
These platforms do not handle audio - they route millions of SIP signaling messages per second, distribute load across server clusters, and protect your infrastructure from attacks.
Kamailio
The SIP Traffic PowerhouseAn absolute powerhouse for handling massive SIP traffic. Build highly scalable SIP routing engines capable of handling thousands of call setups per second while protecting core PBX servers from SIP attacks and overload.
Best for: SIP load balancing, fraud prevention, high-volume routing, WebRTC gateway, carrier interconnect.
OpenSIPS
Enterprise SIP RouterA high-performance SIP routing and load balancing platform with unique enterprise-focused modules and a highly active community. It excels at complex routing logic, call rate limiting, and seamless integration with external systems.
Best for: Enterprise SIP infrastructure, multi-site routing, carrier peering, call center load distribution.
PBX & Contact Center Platforms
Full-featured graphical interfaces that sit on top of core engines to provide visual dashboards, agent management, campaign builders, and comprehensive reporting.
FusionPBX
FreeSWITCH GUIA highly advanced, full-featured graphical interface built strictly on FreeSWITCH. Natively designed for multi-tenant environments, making it ideal for hosting multiple businesses or contact center clients on a single server cluster.
VICIdial
Outbound Dialer KingThe undisputed king of open-source outbound call center software. An enterprise-grade predictive dialer suite built on Asterisk, capable of handling massive outbound campaigns, agent monitoring, and complex lead routing.
FreePBX
Asterisk Made EasyThe most widely used GUI for Asterisk. Incredibly user-friendly and perfect for setting up standard business phone systems with ring groups, IVRs, conferencing, and voicemail management out of the box.
Issabel
All-in-One UC PlatformA fork of the defunct Elastix project. An all-in-one unified communications platform based on Asterisk that includes PBX, faxing, billing, and basic call center modules out of the box. Ready to deploy in minutes.
Billing & Class 4/5 Softswitches
Open-source billing platforms and real-time rating engines for VoIP providers managing prepaid accounts, wholesale trunking, and enterprise telecom billing.
A2Billing
VoIP Billing PlatformAn open-source telecom billing platform usually paired with Asterisk. Used by VoIP providers worldwide to manage calling cards, wholesale SIP trunking, customer invoicing, and prepaid/postpaid account management.
CGRateS
Real-Time Rating EngineA highly advanced, real-time rating and billing engine that integrates flawlessly with Kamailio, OpenSIPS, and FreeSWITCH. Purpose-built for enterprise telecom billing with support for complex rating plans and real-time balance management.
Our Open Source VoIP Services
End-to-end services for the entire lifecycle - from initial installation and configuration to ongoing support, security patching, and scaling your platform to handle millions of calls.
Installation & Setup
Complete installation and configuration of any open-source telephony platform on your infrastructure - bare metal, VPS, or cloud. We handle OS hardening, dependency management, clustering, and production-ready deployment.
Live Demo & POC
Before committing, see it live. We set up fully functional demos and proofs of concept so you can test call flows, agent desktops, IVRs, and admin dashboards with real traffic before going to production.
Security Patches & Hardening
Keep your platform bulletproof. We provide timely security patches, SIP attack prevention, TLS/SRTP enforcement, intrusion detection, rate limiting, and comprehensive security audits for all open-source telephony systems.
Post-Deployment Support
Round-the-clock support to keep your systems running at peak performance. We monitor, troubleshoot, and resolve issues across all layers - SIP, media, database, and infrastructure - so you never miss a call.
Custom Module Development
Extend any open-source platform with custom modules, dialplan logic, APIs, and integrations. Build custom IVRs, billing connectors, CRM integrations, WebRTC frontends, and reporting dashboards tailored to your workflow.
Scalability & Performance Tuning
Make your platform handle 10x more traffic. We architect high-availability clusters, implement load balancing with Kamailio/OpenSIPS, optimize database queries, tune kernel parameters, and design auto-scaling strategies.
Migration & Upgrades
Seamlessly migrate between platforms or upgrade to the latest versions without downtime. We handle Asterisk-to-FreeSWITCH migrations, version upgrades, database migrations, and configuration porting with zero call drops.
Training & Knowledge Transfer
Empower your team to manage the platform independently. We provide hands-on training for system administrators, developers, and operations teams covering architecture, troubleshooting, and day-to-day management.
Our Open Source Ecosystem
The complete technology stack we work with - from core telephony engines to monitoring and infrastructure automation.
Core Engines
SIP Proxies
PBX GUIs
Billing & Rating
Monitoring
Infrastructure
Our Engagement Process
From discovery call to production go-live and ongoing support - a structured, transparent process that delivers results.
Discovery Call
We learn about your infrastructure, current setup, traffic volume, and goals. Whether it is a fresh installation or scaling an existing system, we map out the exact scope.
Architecture & Proposal
Our engineers design the architecture - platform selection, clustering strategy, network topology, security layers, and monitoring. You receive a detailed proposal with timelines and costs.
Environment Setup
We provision servers, install the OS, harden security, compile the platform from source (or deploy via Docker), and configure all dependencies, codecs, and SIP parameters.
Configuration & Customization
SIP trunk setup, dialplan logic, IVR flows, call routing rules, user/tenant provisioning, billing integration, and any custom module development specific to your workflow.
Testing & Load Validation
End-to-end call flow testing, SIP registration tests, codec quality validation, load testing with simulated traffic (SIPp), security scanning, and failover verification.
Go-Live & Monitoring
Cutover to production with real traffic. We monitor call quality (MOS scores), server health, SIP error rates, and system resources for the first 72 hours to ensure stability.
Post-Launch Support
Ongoing support for patches, upgrades, troubleshooting, and optimization. We become your long-term open-source VoIP partner with SLAs that match your business requirements.
Deployment Models
We deploy on any infrastructure. Choose the model that fits your performance, budget, and operational requirements.
Bare Metal / VPS
Maximum performance for media-heavy workloads. Direct hardware access with optimized kernel parameters for the lowest latency and highest call density.
Cloud (AWS / GCP / Azure)
Elastic scaling, global reach, and managed services. Auto-scaling groups, multi-region failover, and pay-as-you-go economics for dynamic workloads.
Docker / Kubernetes
Containerized deployments with orchestration. Reproducible environments, rolling updates, horizontal pod autoscaling, and immutable infrastructure for CI/CD pipelines.
Why Choose Us for Open Source VoIP
We do not just install software. We architect, optimize, secure, and support production telephony infrastructure at scale.
We Run Our Own Platform
AnrizVoIP is our own FreeSWITCH/OpenSIPS-based multi-tenant contact center serving clients in 12 countries. We do not just install - we build and operate production VoIP at scale.
Deep Source Code Mastery
Our engineers work at the source code level of FreeSWITCH, Asterisk, Kamailio, and OpenSIPS. We patch, customize, and extend these platforms far beyond their stock capabilities.
Performance Obsessed
Every deployment is benchmarked for thousands of concurrent calls. We tune at the kernel, database, and application level to squeeze maximum performance from every server.
Security First Approach
TLS/SRTP encryption, SIP authentication, toll fraud prevention, fail2ban, and intrusion detection are standard in every deployment. No shortcuts on security.
Rapid Deployment
Pre-built Ansible/Docker playbooks for all major platforms. A production-ready FreeSWITCH cluster can be up in 48 hours. FusionPBX multi-tenant in 24 hours.
Vendor-Neutral Expertise
We recommend the best tool for your use case - not the one we are most comfortable with. FreeSWITCH for scale, Asterisk for PBX, Kamailio for routing. Right tool, right job.
How to Work With Us
Choose the engagement model that fits your needs - whether it is a one-time setup or an ongoing partnership.
One-Time Setup
Install & ConfigureWe install, configure, and hand over a production-ready system. Perfect for teams who want expert setup and then manage independently.
Managed Support
Ongoing PartnershipMonthly retainer for continuous support, patching, monitoring, and optimization. Your dedicated open-source VoIP operations team.
Custom Development
Build & ExtendCustom module development, platform customization, and deep integration work. For when stock features are not enough for your business.
Which Platform Is Right for You?
A quick comparison to help you choose the right open-source platform for your specific use case.
| Platform | Best For | Scalability | Multi-Tenant | GUI |
|---|---|---|---|---|
| FreeSWITCH | Enterprise Contact Centers | ★★★★★ | Native | FusionPBX |
| Asterisk | PBX & IVR Systems | ★★★☆☆ | Via Modules | FreePBX / Issabel |
| Kamailio | SIP Load Balancing | ★★★★★ | Routing Level | Siremis |
| OpenSIPS | Enterprise SIP Routing | ★★★★★ | Routing Level | OpenSIPS CP |
| VICIdial | Outbound Campaigns | ★★★★☆ | Campaign Level | Built-in |
| YATE | Carrier Routing | ★★★★☆ | Custom | Custom |
Frequently Asked Questions
Common questions about our open-source VoIP development and support services.
It depends on your requirements. FreeSWITCH excels at high-concurrency, multi-tenant contact centers and carrier-grade platforms. Asterisk is ideal for traditional PBX, IVR systems, and unified communications. Kamailio/OpenSIPS are essential for SIP load balancing and routing at scale. YATE is perfect for carrier-grade routing with minimal resource usage. We analyze your traffic patterns, scale requirements, and use case to recommend the right combination.
Yes, this is one of our most common deployments. We install FreeSWITCH with FusionPBX from source, configure multi-tenant domains, set up SIP trunks, provision tenants with isolated configurations, configure call center queues, set up IVRs, and deploy the full admin/tenant portal. The typical setup takes 2-5 days depending on complexity.
Yes. We monitor CVE databases, platform mailing lists, and security advisories for all major telephony platforms. When a critical vulnerability is disclosed, we test the patch in a staging environment and deploy it to your production systems within our SLA window - typically within 24 hours for critical patches. We also implement proactive security measures like SIP attack prevention, TLS/SRTP enforcement, and intrusion detection.
Yes. We handle full platform migrations including dialplan conversion, IVR recreation, SIP trunk re-provisioning, CDR data migration, user/extension porting, and integration re-wiring. We run both systems in parallel during migration and cut over with zero downtime. Most migrations are completed within 1-3 weeks depending on complexity.
We implement multi-layer HA: Kamailio/OpenSIPS as SIP proxies with health checks and automatic failover between FreeSWITCH/Asterisk nodes, database replication (PostgreSQL streaming replication or Galera for MySQL), shared storage for recordings, and keepalived/VRRP for IP failover. This ensures zero single points of failure and sub-second failover.
One-time installations start from $500-$1,500 depending on the platform and complexity. Managed support retainers start from $500/month for basic monitoring and patching, scaling up based on infrastructure size and SLA requirements. Custom development is quoted per project. We provide transparent, detailed estimates after the discovery call.
Yes. We deploy VICIdial with full Asterisk backend including predictive dialer configuration, agent provisioning, campaign setup, lead import pipelines, DNC list management, recording storage, and reporting dashboards. We also customize VICIdial with additional modules for CRM integration, custom dispositions, and advanced reporting.
Yes. We build Docker images for FreeSWITCH, Kamailio, OpenSIPS, and related services with Kubernetes manifests (Helm charts) for orchestration. This includes persistent volume management for recordings, ConfigMap-based configuration, health check probes, horizontal pod autoscaling, and CI/CD pipeline integration for automated deployments.
Yes. We integrate A2Billing with Asterisk for calling card and wholesale billing, and CGRateS with FreeSWITCH, Kamailio, or OpenSIPS for real-time rating and billing. This includes rate plan configuration, CDR mediation, balance management, invoice generation, and payment gateway integration (Stripe, PayPal, etc.).
Our support tiers include: Basic (business hours, email/ticket support, monthly patching), Standard (extended hours, 4-hour response SLA, weekly monitoring reports), and Premium (24/7 support, 1-hour critical response SLA, proactive monitoring, dedicated engineer). All tiers include security patch management, performance monitoring, and quarterly health checks.
Absolutely. We implement horizontal scaling with Kamailio/OpenSIPS load balancers in front of FreeSWITCH/Asterisk clusters, optimize database schemas and queries, tune OS kernel parameters (network buffers, file descriptors, CPU affinity), implement connection pooling, and design auto-scaling strategies for cloud deployments. We have scaled single-cluster deployments to handle high concurrent call volumes.
Yes. We build modern WebRTC-based agent desktops, softphones, and admin dashboards using React/Next.js with SIP.js or JsSIP connecting to FreeSWITCH or Asterisk backends. This includes call controls, real-time statistics, call recording playback, CRM screen pops, and responsive mobile-friendly designs.
Ready to Leverage Open Source VoIP?
Whether you need a fresh FreeSWITCH installation, Kamailio load balancing, VICIdial campaign setup, or ongoing support - our team of open-source telephony experts is ready.