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I will setup and configure your voip server, sip trunk and pbx system
DevOps and Linux Server Engineer, Nginx, Docker, Migrations
I set up, configure, and fix Linux servers — production experience, not tutorials.
WHAT I DO:
- Linux server setup & administration (Ubuntu/Debian)
- Nginx: reverse proxy, SSL, complex routing, Apach...
About this Gig
VoIP breaks in ways that generic sysadmins can't diagnose: calls connect but nobody hears anything, phones register and then drop, audio works one way only, calls fail behind NAT. These are not server problems they're signaling and media problems.
WHAT I DO:
- Install and configure VoIP servers on Linux
- SIP trunk setup and provider connection
- Extensions, dial plans, call routing rules
- Media handling: RTP proxying, codec negotiation, transcoding
- NAT traversal for phones and softclients behind private networks
- WebRTC and secure media (SRTP/TLS)
- CRM integration: click-to-call, call logs, screen pop
- Troubleshooting existing systems: one-way audio, registration failures, dropped calls, poor call quality
WHAT I WORK WITH:
Kamailio, FreeSWITCH, RTPEngine, and standard SIP infrastructure on Ubuntu/Debian. GSM gateways and hardware SIP endpoints.
EXPERIENCE:
I built a multi-tenant telephony platform from the ground up SIP routing, media layer, NAT handling, and CRM integration, running in production with real traffic. The software is registered in the state copyright registry under my name as co-author.
I read SIP traces. When a call fails, find where in the signaling
Provider:
Asterisk
•
FreePBX
•
Other
Other Support & IT Services I Offer
FAQ
Why is my audio one-way or missing entirely?
Almost always NAT. Signaling reaches both sides, so the call "connects," but the media stream can't find its way back through a router or firewall. Fixing it means correcting how your server advertises media addresses and, usually, putting a media proxy in the path.
Do I need my own server?
For a PBX setup, yes - a VPS with root access is enough for most small and mid-size deployments. If you don't have one, message me and I'll suggest specs based on how many concurrent calls you expect.
Can you work with my existing SIP provider?
Yes. Send me the trunk details your provider gave you (host, credentials, allowed codecs) and I'll configure the connection. I work with standard SIP providers, not proprietary closed platforms.
What access do you need?
Root SSH access to the server, SIP trunk credentials from your provider, and network details — whether the server has a public IP or sits behind NAT. You can change all credentials after delivery.
Can you fix a system someone else set up?
Yes, that's a large part of this work. Order the Basic package for diagnostics: I'll find what's wrong and tell you what it takes to fix. If you then want me to fix it, that cost applies toward the full job.
