AriLink

<p align="center"> <img src="./assets/header.webp" alt="AriLink" width="100%" /> <br> <em><a href="https://asterisk.org"><img src="https://asterisk.org/wp-content/uploads/asterisk-logo.png" alt="Asterisk" width="50" style="vertical-align: middle;"/></a>-powered telephony platform with real-time transcription, campaigns, and a full web dashboard</em> </p>

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Overview

AriLink is a telephony platform built on Asterisk's ARI (Asterisk REST Interface). It combines a Nuxt 4 web dashboard, a Rust RTP audio pipeline, and local AI transcription into a single system for managing calls, running outbound campaigns, and monitoring everything in real time.

<p align="center"> <img src="./assets/dashboard.png" alt="AriLink Dashboard" width="100%" /> </p>

Key Features

• LLM Voice Agent - Full conversational AI via LlmChatBrain — streaming LLM responses with sentence-by-sentence TTS
• Barge-In - Interrupt bot speech mid-sentence. AEC3 echo cancellation + Silero VAD eliminate false triggers
• Pluggable Brains - Swap call logic via config: LLM Chat, IVR Transfer, Direct Dial, OpenClaw AI
• Local TTS - Text-to-speech via Kokoro (82M params) — zero API cost, ~100ms latency
• Local STT - Real-time transcription via Parakeet TDT (25 languages) or Whisper, with Google Cloud fallback
• Web Dashboard - Full management UI with real-time service status, active calls, logs, and configuration
• Campaign Engine - Automated outbound dialing with configurable concurrency and call scripts
• OpenClaw Integration - Bridge to OpenClaw AI agent ecosystem via Socket.IO
• Call History - SQLite-backed call records with search and transcription playback
• CLI Tool - arilink init/start/stop/status/logs for Docker deployments
• Built-in Softphone - SIP softphone directly in the dashboard for testing
• SSH Terminal & File Manager - Remote PBX management and audio file uploads via SFTP

<p align="center"> <img src="./assets/softphone.png" alt="Built-in Softphone" width="220" /> </p>

Architecture

Built on Nuxt 4 + Nitro (single server for SPA, API, and Socket.IO), with a Rust RTP server for audio and Python services for transcription:

flowchart TD
    subgraph Dashboard["Nuxt 4 Dashboard"]
        UI[Vue 3 SPA]
        API[Nitro API Routes]
        SIO[Socket.IO]
    end

    subgraph Core["Node.js Core"]
        CTRL[ARI Controller]
        HARNESS[BrainHarness]
        BRAINS["Brains<br/><small>LlmChat · IvrTransfer<br/>DirectDial · OpenClaw</small>"]
        CAMP[Campaign Engine]
        HIST[Call History]
    end

    subgraph Audio["Rust RTP Server"]
        RTP[RTP Receiver :8000]
        AEC["AEC3 Echo Cancellation<br/>+ Silero VAD"]
    end

    subgraph AI["AI Services"]
        PKT[Parakeet STT]
        WHSP[Whisper]
        GCS[Google Cloud Speech]
        KOKORO[Kokoro TTS]
    end

    PBX[Asterisk PBX] <-->|ARI WebSocket| CTRL
    PBX -->|RTP Audio| RTP
    RTP --> AEC
    AEC -->|Clean audio| PKT
    AEC -.->|Fallback| WHSP
    AEC -.->|Fallback| GCS
    AEC -->|user_speaking| CTRL
    PKT -->|Transcription| CTRL
    CTRL --> HARNESS --> BRAINS
    BRAINS -->|speak| KOKORO
    KOKORO -->|slin16 audio| PBX
    CTRL --> HIST
    CAMP --> CTRL
    SIO <--> Core
    SIO <-->|Real-time| UI
    API --> HIST

    BRAINS <-.->|Socket.IO| OC[OpenClaw AI]

Core Components

<details> <summary><b>Engine</b></summary>

Orchestrates all backend services on startup:

• Starts the Rust RTP server and Parakeet transcription (in parallel)
• Connects to Asterisk ARI
• Wires up the Dashboard, Call History, and Campaign Engine
• Graceful shutdown of all child processes

</details> <details> <summary><b>ARI Controller</b></summary>

Interfaces with Asterisk PBX via ARI:

• Manages call flows, bridges, and DTMF input
• Delegates call logic to pluggable assistants
• Routes transcription results to active calls
• Handles contact lookups for voice-activated dialing

</details> <details> <summary><b>Rust RTP Server</b></summary>

High-performance audio pipeline with voice intelligence:

• Receives RTP audio from Asterisk ExternalMedia channels
• AEC3 echo cancellation — WebRTC-grade, 30-40dB attenuation with adaptive threshold
• Silero VAD — neural speech detection during TTS playback (barge-in trigger)
• RMS gate — filters noise below threshold before sending to STT
• Per-session audio routing with codec handling (slin16, Opus)
• Forwards clean audio to transcription services via WebSocket

</details> <details> <summary><b>Text-to-Speech (Kokoro)</b></summary>

Local TTS service for zero-latency, zero-cost speech synthesis:

• Kokoro 82M — lightweight neural TTS model
• Outputs 16kHz slin16 PCM directly for Asterisk playback
• Sentence-by-sentence streaming — start speaking before LLM finishes
• Configurable voice, speed, and language
• WebSocket protocol for low-latency communication

</details> <details> <summary><b>Transcription Providers</b></summary>

Multiple backend support with automatic failover:

• Parakeet TDT 0.6B-v3 (recommended) — 25 languages, 2000x+ real-time speed, runs on GPU
• Whisper — Slower but highly accurate
• Google Cloud Speech — Cloud-based fallback (requires API credentials)
• Streaming transcription with interim and final results
• Priority chain configured via TRANSCRIPTION_SERVICES env var

</details> <details> <summary><b>Pluggable Brains</b></summary>

Modular call logic — swap behavior via config.json without changing code:

• LlmChatBrain — Full conversational AI (streaming LLM + sentence-by-sentence TTS + barge-in)
• IvrTransferBrain — DTMF gate → voice → contact match → transfer
• DirectDialBrain — Voice → contact match → direct transfer
• OpenClawBrain — Bridge to OpenClaw AI agent ecosystem
• BrainHarness — shared voice intelligence (filler filtering, turn debouncing, barge-in handling) benefits all brains automatically
• Create custom brains by implementing the IBrain interface

</details> <details> <summary><b>Campaign Engine</b></summary>

Automated outbound dialing:

• Upload phone lists and run campaigns from the dashboard
• Configurable concurrency, retry logic, and call scripts
• Real-time campaign progress via Socket.IO
• Per-call results stored in campaign-results/

</details>

Dashboard Pages

| Page | Description | |------|-------------| | Dashboard | Service status (Asterisk, Rust RTP, Transcription), active calls | | Calls | Call history with transcription, search, and playback | | Assets | Upload/manage Asterisk audio files via SFTP | | Terminal | SSH terminal for remote PBX management | | Campaigns | Create and run outbound call campaigns | | Contacts | Manage contacts for voice-activated dialing | | Assistants | Configure call handling logic per assistant | | Logs | Real-time and historical application logs | | Config | Edit environment variables and server settings | | Docs | Built-in documentation viewer |

Getting Started

Docker (Recommended)

The fastest way to try AriLink — no Node.js, Python, or Asterisk installation needed:

docker compose up -d

This starts 4 services: Asterisk (PBX), Parakeet (STT), Kokoro (TTS), and AriLink (dashboard + engine). Open localhost:3011.

Test with a SIP phone (Zoiper, Linphone): server localhost:5060, extension 1001, password demo1001.

# With GPU acceleration (NVIDIA) — ~1000x faster STT + TTS
docker compose -f docker-compose.yml -f docker-compose.gpu.yml up -d

# With Rust RTP server (AEC3 echo cancellation + Silero VAD for barge-in)
docker compose --profile rust-rtp up -d

# With live code editing (mount local source)
docker compose -f docker-compose.yml -f docker-compose.dev.yml up -d

See docs/docker.md for full Docker guide, configuration, and troubleshooting.

Manual Setup

For development or connecting to an existing FreePBX server:

Prerequisites

1. Set up FreePBX server:
   • New installation? FreePBX Installation Guide
   • Already installed? FreePBX ARI Configuration
2. Node.js 23+ and npm
3. UV (Python package manager) for transcription services:

   # Windows PowerShell
   irm https://astral.sh/uv/install.ps1 | iex
   

Installation

1. Clone and install dependencies (one command installs everything):

   npm install
   

   This automatically installs both the core server and dashboard dependencies.

2. Set up transcription (local speech recognition):

   cd transcription-services/parakeet-service
   python -m venv .venv
   uv pip install -r requirements.txt
   

3. Configure environment:

   cp .env.example .env
   # Edit .env with your FreePBX IP, ARI credentials, etc.
   

Running

# Development (hot reload, auto-starts Parakeet if configured)
npm run dev

# Production (builds if needed, then starts)
npm start

# Build only
npm run build

Set AUTO_START_TRANSCRIPTION=true in .env to have the launcher automatically start Parakeet in parallel with the server. No separate terminal needed.

The dashboard will be available at http://localhost:3011. All settings can be managed from the Config page once the server is running.

Assistant Modes

np