Resolve 3CX WhatsApp Call Audio
Budget: $30 – $250 USD
I already have a cloud-based 3CX instance talking to whatsapp calling API over PJSIP for an outbound WhatsApp voice API. Calls connect fine, but only the caller’s audio reaches the line; the person being called hears nothing.
Here’s what you need to know up front:
• Environment: fully cloud-hosted, no on-prem gear
• Media: SRTP from Asterisk, RTP from 3CX
• Symptom: recipient can hear, but caller cannot
Your task is to trace the SRTP→RTP path, pinpoint why one leg is silent, and adjust the PJSIP, codec, or NAT/STUN settings so we get clean two-way audio. Once it works, I also need a short write-up of the changes so I can replicate them in staging.
Acceptance criteria
1. Two-way audio confirmed on at least five consecutive outbound WhatsApp calls placed from 3CX
2. Updated PJSIP/SRTP configs returned to me (or pushed to my repo)
3. Quick walkthrough or screen-share explaining the fix
Tools you’ll likely touch: Asterisk 20+, PJSIP, Wireshark, sngrep, 3CX console. SSH access is ready whenever you are.
Here’s what you need to know up front:
• Environment: fully cloud-hosted, no on-prem gear
• Media: SRTP from Asterisk, RTP from 3CX
• Symptom: recipient can hear, but caller cannot
Your task is to trace the SRTP→RTP path, pinpoint why one leg is silent, and adjust the PJSIP, codec, or NAT/STUN settings so we get clean two-way audio. Once it works, I also need a short write-up of the changes so I can replicate them in staging.
Acceptance criteria
1. Two-way audio confirmed on at least five consecutive outbound WhatsApp calls placed from 3CX
2. Updated PJSIP/SRTP configs returned to me (or pushed to my repo)
3. Quick walkthrough or screen-share explaining the fix
Tools you’ll likely touch: Asterisk 20+, PJSIP, Wireshark, sngrep, 3CX console. SSH access is ready whenever you are.