Finalize Python/Asterisk Autodialer Integration
Budget / Salary$30–250
TypeFreelance project
LocationRemote
Posted2 hours ago
I have an existing Python/Asterisk outbound autodialer running on a Debian 12 VPS. The application and dashboard are already built. I need an experienced Asterisk/ARI/AMI/Python developer to finish integrating the existing components, NOT rebuild the project from scratch.
The system currently places outbound calls through a VMGate SIP/mobile gateway. Calling is working. We have also confirmed working two-way media/audio capture, AMI MixMonitor recording, DTMF transmission, and speech-to-text/transcription.
The main remaining task is to connect the existing IVR state-machine (ivr_flow.py) to the working live audio/transcription and DTMF system so the dialer can react to IVR prompts in real time rather than relying on fixed delays or post-call transcription.
Important technical details:
Debian 12 VPS
Asterisk 22
Python application
ARI for call/channel control
AMI for MixMonitor and DTMF
PJSIP/VMGate outbound trunk
Existing web dashboard and CSV workflow
Existing IVR state-machine
Deepgram/live STT code already exists
VMGate requires continuous RTP/silence to maintain the media path
MixMonitor has been verified to capture valid IVR audio
DTMF transmission has been verified
I need someone who can inspect the existing code, preserve the parts that already work, and finish the real-time IVR call-control integration. Please do not propose rebuilding the entire application.
I can provide the existing source code, server access, current configuration, logs, and a detailed technical handoff explaining what has already been tested and what remains.
The system currently places outbound calls through a VMGate SIP/mobile gateway. Calling is working. We have also confirmed working two-way media/audio capture, AMI MixMonitor recording, DTMF transmission, and speech-to-text/transcription.
The main remaining task is to connect the existing IVR state-machine (ivr_flow.py) to the working live audio/transcription and DTMF system so the dialer can react to IVR prompts in real time rather than relying on fixed delays or post-call transcription.
Important technical details:
Debian 12 VPS
Asterisk 22
Python application
ARI for call/channel control
AMI for MixMonitor and DTMF
PJSIP/VMGate outbound trunk
Existing web dashboard and CSV workflow
Existing IVR state-machine
Deepgram/live STT code already exists
VMGate requires continuous RTP/silence to maintain the media path
MixMonitor has been verified to capture valid IVR audio
DTMF transmission has been verified
I need someone who can inspect the existing code, preserve the parts that already work, and finish the real-time IVR call-control integration. Please do not propose rebuilding the entire application.
I can provide the existing source code, server access, current configuration, logs, and a detailed technical handoff explaining what has already been tested and what remains.
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