Asterisk IVR & SIP Integration

Require is a custom Asterisk-based autodialer engineered for high-volume, unregulated outbound traffic. The system must run on a hardened Linux distribution (e.g., Debian or CentOS) with Asterisk 18 or 20 compiled with specific modules: chansip for SIP trunking, appdial for call routing, and app_playback for IVR management. The core architecture includes a predictive dialing engine built in Python or C++ that interacts with Asterisk via AGI (Asterisk Gateway Interface). It must integrate the sofia or chansip channel driver to handle SIP trunks, supporting multiple concurrent legs for rapid call placement. For DTMF handling, the system will utilize the appdial with D option to detect dual-tone multi-frequency signals, routing them to specific IVR branches defined in extensions.conf. The IVR layer requires a multi-level menu structure using appvoicemail or appplayback for prompts, coupled with app_read for user input. Voice recognition must be integrated via a local instance of Vosk or PocketSphinx for offline, latency-free speech-to-text processing, parsing keywords to trigger dynamic call flows without external API dependencies. SIP trunk integration will be configured in [login to view URL] with dynamic IP rotation capabilities to mask the source, allowing the system to pull credentials from a JSON database that updates every 60 seconds to avoid blacklisting. The dialer logic will implement "burner" logic to discard trunks that fail a threshold of successful connections. The final deliverable is a Dockerized stack containing the Asterisk server, the Python dialing script, a MySQL database for lead management, and a lightweight GUI (using Flask) to monitor real-time call status, DTMF logs, and voice recognition transcripts. The system will be pre-configured to bypass standard carrier filters by randomizing jitter and call intervals, ready to deploy on a VPS with a static IP range, fully tuned for immediate, high-speed operation.

Project ID: 40641959