Expert FreePBX/Asterisk Engineer Needed to Fix No Audio in Sangoma Talk

I am looking for a highly experienced FreePBX and Asterisk VoIP engineer to diagnose and permanently resolve a complex no-audio issue affecting our Sangoma Talk mobile users.

We have already spent considerable time troubleshooting this problem, including extensive configuration changes, packet captures, and network analysis, but the issue remains unresolved.

Our System:

* FreePBX 17 / Asterisk 22
* Debian 12 VPS
* Sangoma Talk on iOS and Android
* SIP over TLS (port 5061)
* SDES-SRTP for encrypted audio
* G.711 A-law / μ-law
* Yeastar TG400 GSM Gateway

The Problem:

Sangoma Talk works correctly for users located in Saudi Arabia.

However, users located in Egypt can successfully register, make calls, and receive calls, but there is no audio in either direction.

Our investigations revealed that RTP packets from one affected user arrive with sequence number increments of +2 instead of +1, causing Asterisk Strict RTP probation to fail.

A similar sequence issue was also observed during testing with unencrypted RTP, but the actual root cause remains unidentified.

We have already investigated TLS, SRTP, NAT, firewall rules, RTP ports, codecs, and several network configurations without finding a solution.

What I Need:

1. Perform a thorough, hands-on technical investigation of the existing FreePBX/Asterisk system.
2. Identify the actual root cause of the no-audio problem.
3. Implement a reliable, permanent solution.
4. Verify successful two-way audio with affected users in Egypt.
5. Preserve system security, including TLS and SRTP, and avoid disrupting existing services.
6. Provide a technical summary of the root cause and the implemented solution.

Strict Requirements:

* Proven expert-level experience with FreePBX, Asterisk, SIP, RTP/SRTP, NAT, and VoIP troubleshooting.
* Strong experience using Wireshark, tcpdump, Asterisk CLI, and packet-level diagnostics.
* Previous experience resolving complex one-way or no-audio VoIP problems.
* NO AI TOOLS ALLOWED. You must not use ChatGPT, Claude, Copilot, or any other AI tools for troubleshooting, implementing the solution, or preparing the technical report. All work must be performed through your own professional expertise and hands-on technical analysis.
* No major configuration changes, disabling security features, or restarting production services without prior approval.

Detailed technical reports, diagnostic findings, and sanitized logs are available.

Important: I am not looking for basic FreePBX setup assistance or generic troubleshooting suggestions. I need a genuine VoIP specialist who can investigate this specific issue, identify its root cause, and implement a verified solution.

Please apply only if you have strong, relevant experience. Briefly describe similar VoIP audio issues you have personally diagnosed and resolved.

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來源：Freelancer.com | 出價次數：1 | 剩餘時間：6 日 | 專案類型：固定預算