I will troubleshoot and fix your freeswitch sip trunk call drop and audio issues

Is your FreeSWITCH server dropping calls, rejecting registrations, or giving you one-way audio? I will diagnose and fix it.

With 13 years of VoIP engineering experience, I have resolved hundreds of FreeSWITCH and SIP issues across production environments serving enterprise clients in 13+ countries.

WHAT I CAN FIX:

One-way audio or no audio (NAT/RTP issues)

403 Forbidden / 404 Not Found registration errors

Calls dropping after 30s or 32s (SIP timer issues)

SIP trunk not connecting or rejecting INVITE

Codec mismatch 488 Not Acceptable errors

IVR or dialplan not routing correctly

FreeSWITCH not starting or crashing on startup

High CPU / memory issues under load

MY PROCESS:

1. You share logs, config files, or sngrep/PCAP traces

2. I identify the root cause

3. I apply the fix and confirm with you

4. Basic package: written summary of what was wrong and what was changed

WHAT I NEED FROM YOU:

- SSH access OR relevant log files and configs

- Description of what is happening vs what you expect

If you are unsure what to share, message me first I will guide you.