Asterisk PBX Call-Drop Fix
Budget: $200 – $300 USD
Calls on my Asterisk PBX drop right at the moment they are answered, and the behaviour is random—some users connect without trouble while others are cut off instantly. This is not an audio-quality or configuration-syntax question; it is a stability bug that only shows itself at the very start of certain calls.
What I need is a swift, well-documented fix that pinpoints the root cause (SIP signalling, codec negotiation, trunk response, or any misbehaving module), applies the corrective configuration or patch, and proves the solution with live test calls. Access to full CLI, logs, and SIP traces can be arranged over SSH or a screen-sharing session at short notice.
Deliverables
• Root-cause analysis summary
• Corrected configuration or code patch applied on the server
• Verification: 10 consecutive test calls placed without the initial-drop issue
• Brief post-fix report outlining the steps taken so I can replicate them in staging
Speed matters here, so familiarity with Asterisk 18+, PJSIP, and typical carrier interop quirks will help you move quickly.
What I need is a swift, well-documented fix that pinpoints the root cause (SIP signalling, codec negotiation, trunk response, or any misbehaving module), applies the corrective configuration or patch, and proves the solution with live test calls. Access to full CLI, logs, and SIP traces can be arranged over SSH or a screen-sharing session at short notice.
Deliverables
• Root-cause analysis summary
• Corrected configuration or code patch applied on the server
• Verification: 10 consecutive test calls placed without the initial-drop issue
• Brief post-fix report outlining the steps taken so I can replicate them in staging
Speed matters here, so familiarity with Asterisk 18+, PJSIP, and typical carrier interop quirks will help you move quickly.