I will fix no audio and call drops on your voip or freeswitch system

Calls dropping after 30 seconds? One-way or no audio? Extensions won't register? SIP trunk rejecting calls with 403, 408 or 488? I find the root cause and fix it not a patch that breaks again next week.

I'm a software engineer at a telecom company in Dhaka, working daily on FreeSWITCH, Asterisk and Kamailio. Most VoIP faults come from a short list of causes (NAT/RTP, codec mismatch, firewall, trunk auth, dialplan logic, TLS/SRTP), and I read SIP traces and logs to pinpoint yours.

What I fix

No audio / one-way audio (NAT, RTP, STUN/TURN)

Call drops, early hangups, timeouts

Registration and SIP trunk failures (Twilio, Telnyx, VoIP.ms, any provider)

Dialplan, IVR and call-routing errors

FreeSWITCH, FusionPBX, Asterisk, FreePBX, Kamailio, OpenSIPS

WebRTC and softphone connection problems (Zoiper, Linphone, SIP.js)

Security: brute-force attacks, Fail2Ban, fraud lockdown

How it works

1. Message me the symptom, your platform, and a log or trace if you have one.

2. I confirm scope and send a custom offer.

3. I fix it over SSH, explain what was wrong, and leave notes so it stays fixed.

Please contact me before ordering so I can confirm your issue is within scope.