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$
USD
I will manage your freeswitch server
VoIP Architect
I build voice infrastructure that carries real traffic. 20+ years in open-source telephony; FreeSWITCH, Asterisk, Kamailio, SIP, WebRTC. I've deployed production systems for banks, financial instituti...
About this Gig
I build production FreeSWITCH systems that don't fall over at 3am.
20+ years in open-source telephony. I've deployed carrier-grade voice infrastructure for banks and contact centers migrating them off Avaya, Cisco, and Genesys onto FreeSWITCH and Asterisk.
Not a copy-paste install script. Every deployment ships hardened, monitored, and documented.
What you get:
- FreeSWITCH compiled and tuned for your traffic profile, not distro defaults
- SIP trunking with any carrier Twilio, Telnyx, Bandwidth, or your local ITSP
- Security hardening: fail2ban, ACLs, TLS/SRTP, toll-fraud protection
- WebRTC/Verto for browser-based calling (browser webphone included)
- Dialplan architecture that scales IVR, queues, hunt groups, time conditions
- T.38 fax, voicemail-to-email, call recording + transcription
- Voice AI: Pipecat, LiveKit, or custom STT/LLM/TTS pipelines at sub-second latency
Stack: FreeSWITCH · Asterisk · Kamailio · SIP/WebRTC · Elixir · FastAPI · Docker/K8s
FAQ
Do I need to provide a server?
Yes. You provide root SSH access to a fresh VPS or bare-metal box (Debian/Ubuntu preferred). I'll spec the sizing for your call volume before you order. AWS, Azure, DigitalOcean, Hetzner, or on-prem all work.
What server specs do I need?
Depends on concurrent calls. Rough guide: 2 vCPU / 4GB handles ~100 concurrent calls without transcoding. Transcoding, recording, and Voice AI change this significantly. Message me with your numbers and I'll size it properly.
Do you provide SIP trunks or phone numbers?
No; I configure whatever carrier you choose. I work with Twilio, Telnyx, Bandwidth, Vonage, and regional ITSPs. If you don't have a carrier yet, I'll recommend one based on your country and traffic type.
Can you migrate my existing Asterisk/FreePBX/3CX setup?
Yes. I start with an audit of your current dialplan, extensions, and trunks, then map it onto FreeSWITCH. Migrations usually need a custom offer -- message me first with your current system details.
Will my calls have good audio quality?
Audio quality is mostly network, not software. I configure codecs, jitter buffers, and NAT traversal correctly, and set up STUN/TURN for WebRTC. If your network has issues, I'll identify them; but I can't fix your ISP.
What does "server hardening" actually include?
Firewall rules and SIP ACLs, fail2ban with SIP-aware filters, TLS + SRTP encryption, disabling anonymous SIP, strong auth policies, and toll-fraud rate limiting. This is included in every package, not an upsell.
Which Voice AI frameworks do you support?
Pipecat and LiveKit Agents primarily. I can also wire custom STT/LLM/TTS pipelines using Deepgram, ElevenLabs, OpenAI, or Anthropic. Premium package covers integration; API keys and usage costs are yours.
Do you provide ongoing support after delivery?
Delivery includes documentation and a handover walkthrough. Revisions cover issues with my work. For ongoing monitoring and maintenance, I offer a separate monthly retainer — ask me for details.
Can you sign an NDA?
Yes. Send it before ordering. I regularly work with banks and financial institutions and I'm comfortable with confidentiality requirements.
Why FreeSWITCH instead of a hosted provider?
Control and cost. Hosted platforms charge per seat and per minute forever, and you can't touch the call logic. FreeSWITCH is a one-time build with unlimited extensions. I'll be honest if hosting is genuinely the better fit for you.
