Hire VoIP Developer | Asterisk, FreeSWITCH & Kamailio Experts
Need to hire a VoIP developer or fix a broken phone system fast? Our Asterisk, FreeSWITCH, Kamailio, and OpenSIPS engineers deliver same-day troubleshooting and production-grade VoIP development across open-source and commercial platforms.
Core Capabilities
Custom Asterisk & FreeSWITCH Development
ARI, ESL, AMI and AGI applications in Python and Node.js — custom IVRs, diallers, bridges and call control logic.
PBX Installation & Migration
Fresh installs, OS hardening, SIP trunk setup, IVR configuration and zero-downtime migrations from legacy systems.
Same-Day Troubleshooting
One-way audio, SIP registration failures, choppy audio, SIP fraud — diagnosed and fixed using sngrep, Wireshark and platform logs. Most critical issues resolved the same business day.
Cloud VoIP Deployments
AWS, GCP, DigitalOcean and OVHcloud VoIP deployments with fail2ban, UFW, STUN/TURN and carrier-grade SIP proxies.
CRM & CTI Integration
Screen-pop, click-to-dial, automatic call logging and post-call disposition for Salesforce, HubSpot, Zoho and GoHighLevel.
Contact Centre Platforms
Vicidial predictive diallers, Asterisk queue management, real-time wallboards and TCPA-compliant DNC scrubbing.
Tools & Platforms
Technical Questions
Answered
Can't find your answer? Talk to our team directly →
What VoIP platforms do you specialise in?
Our VoIP engineers cover the full spectrum: open-source (Asterisk ARI/AMI/AGI, FreeSWITCH ESL, Kamailio, OpenSIPS, RTPEngine, WebRTC) and commercial/cloud (FreePBX, VitalPBX, Vicidial, Issabel, 3CX, Amazon Connect, RingCentral, Five9). Each engineer has a primary speciality and cross-stack experience.
Can you fix our broken VoIP system urgently?
Yes — same-day troubleshooting is one of our core strengths. Our engineers diagnose one-way audio, registration failures, call quality issues, SIP fraud and performance bottlenecks using sngrep, Wireshark and platform-specific logging tools. Most critical issues are resolved the same business day.
How much does VoIP development cost?
A basic custom IVR or Asterisk dialplan project starts from $2,000–$5,000 (1–2 weeks). A full contact centre build with predictive dialler, CRM integration, real-time wallboards, and multi-tenant PBX typically runs $15,000–$50,000. We also offer hourly engagements at $25–$55/hr for ongoing development and support.
Do you handle both development and system administration?
Yes. Our VoIP engineers are full-stack telephony professionals — they write application code (ARI, ESL, AMI in Python/Node.js), configure SIP trunks and dialplans, tune Linux server performance, set up firewalls and fail2ban, and manage cloud deployments on AWS/GCP/DigitalOcean.
Can you integrate our phone system with our CRM?
Absolutely. CTI integration is a core strength — screen-pop from inbound caller ID, click-to-dial from CRM records, automatic call logging, and post-call disposition updates. We have done this with Salesforce, HubSpot, Zoho, GoHighLevel and custom REST APIs across Asterisk, FreePBX, VitalPBX, 3CX and Amazon Connect.
In-Depth Technical Reads
Asterisk vs FreeSWITCH in 2026: Which VoIP Platform Should You Choose?
After 250+ VoIP projects: FreeSWITCH scales higher, Asterisk deploys faster. WebRTC, IVR, AI integration & real hosting costs benchmarked. Our 2026 verdict.
Read Article →Kamailio vs OpenSIPS in 2026: Choosing the Right SIP Proxy for High-Volume Telephony
50+ enterprise SIP deployments: Kamailio vs OpenSIPS on performance, scripting, module ecosystem, load balancing and WebRTC for 2026.
Read Article →Diagnosing One-Way Audio in Asterisk: A Practical Debugging Guide
The definitive NAT traversal and RTP debugging guide for Asterisk. Covers sngrep, Wireshark, rtp direct media, externip, and localnet settings.
Read Article →Get a Free VoIP Consultation
Get a free consultation — real technical answers from our engineering team, no sales pitch.