Zahid H.

I build and fix production calling systems — Asterisk, Kamailio, and the SIP infrastructure underneath them. Most recently I shipped multi-tenant WhatsApp Business Calling on a Wazo/Asterisk platform: a Kamailio → RTPEngine → Asterisk media bridge terminating DTLS-SRTP from Meta's SIP gateway, with ICE negotiation, a WebRTC dialer, per-tenant routing, IVR, call recordings, CDRs, and full REST API control. It went to production and the client kept me on for ongoing work. Not many engineers have taken that particular integration live. What I take on: → WhatsApp Business / Meta Calling — SIP gateway, DTLS-SRTP, ICE, RTP → Asterisk, Wazo, FreePBX, FusionPBX — architecture, build, and production debugging → Kamailio + RTPEngine — SIP routing, media bridging, NAT traversal → Multi-tenant VoIP platforms with REST API management → SIP trunking on any provider — Telnyx, Twilio, Plivo, Bandwidth → WebRTC dialers and softphones — JsSIP, SIP.js, MicroSIP, Zoiper, Linphone → Voice AI and call automation on top of SIP — LiveKit, agent pipelines → One-way audio, codec, and NAT failures diagnosed on live traces (sngrep, Wireshark) → Self-hosted on DigitalOcean, AWS, Azure, or your own Linux host How I work: I scope the build before quoting, deliver in milestones you can verify, and hand over documentation your team can run the system from. No black box left behind. I work from UTC+5, which overlaps a full business day with Europe and the Gulf — so live call debugging happens in real time with you on the call, not overnight in a ticket queue. Best fit: production systems, multi-tenant platforms, and integrations that have to survive real traffic. If you need a one-hour PBX tweak, I'm probably not your cheapest option. Tell me what you're building, or send me the trace of what's breaking. — Zahid