Self-hosted voice AI agent for Asterisk/FreePBX
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Hi HN folks ! I am the author of AVA, a self hosted AI Voice Agent that plugs into Asterisk/Freepbx so you own all the aspects of an AI Voice agent in your own infrastructure. It uses Asterisk native Audiosocket/RTP with python engine to run STT,LLM and TTS loop. The project support several full providers openai, gemini, grok, elevenlabs out of the box and also provides options to build custom pipelines by choosing different stt tts and llm.It also supports full local agent if you have a GPU with 25GB RAM which enables realtime conversation along with tool calling.I started this as a hobby project last year when I started exploring voice agents and every saw every Saas tried to lock you in their eco system. Since then project has taken off and a lot of asterisk people started using it due to ease of setup and wide range of providers and flexibility.Core is MIT: https://github.com/hkjarral/AVA-AI-Voice-Agent-for-Asterisk Free and open source forever.No install Mock UI us hosted at: https://demo.agent6789.comI am also exploring to convert this into a more polished version for Operators to handle multiple customer and multi tenant environment and I would genuinely like to hear feedback on it.Happy to answer any questions and gather any feedback. Thanks.
HN · launched Jul 13, 2026 · 32 upvotes (top 21%) · 0 views
Niche: ai voice agents and platforms · 197 launches, 20 in the last 7 days ▲ +25% · Explore this niche →
Category and tags
- Horizontal
- Agent / copilot
- B2B
- AI-native
- Commercial product
Visit website ↗ Source listing ↗
More like this: all 197 launches in ai voice agents and platforms →
Could you build this?
Partial Interfacing standard STT, LLM, and TTS APIs via Python is straightforward, but integrating tightly with Asterisk PBX via native AudioSocket and RTP streams with low-latency bidirectional voice mechanics is complex.
What it would actually take: The architecture requires an Asterisk/FreePBX deployment paired with a daemon handling AudioSocket or raw RTP/UDP audio streams, transcoding audio (e.g., G.711, Opus) on the fly. It requires voice activity detection (Silero VAD), interruptible duplex audio playback buffers, and streaming WebSocket pipelines linking STT, LLM, and TTS engines under 500ms latency. Deep telephony knowledge (SIP, dialplans, RTP jitter) is required alongside realtime audio streaming infrastructure.
Discussion
11 comments analyzed.
Competitors mentioned: FreePBX, FusionPBX, Freeswitch, Vicidial
Concerns raised: Difficult to justify cost of maintaining self-hosted voice infrastructure vs cheap commercial services, People strongly prefer not to talk to AI voice agents, Spam/scam calls have made phone interactions less desirable
Feature requests: Vicidial integration, System requirements documentation
Competitors
Other products that read as similar to this one : 443 launches clear the similarity bar, closest 8 shown.
Attention rank: #84 of 444 (itself plus its competitors, highest first, normalized so YC and Product Hunt are compared fairly).
Launched 251 days after the earliest competitor.
- Ava · hn · 2026-03-12 · 7 upvotes · similarity 0.83
- openring · ph · 2026-11-10 · 64 upvotes · similarity 0.52
- We open sourced Vapi · hn · 2026-03-12 · 8 upvotes · similarity 0.49
- esp_ai · github · 2026-09-19 · 15 upvotes · similarity 0.49
- Dograh · hn · 2025-12-08 · 16 upvotes · similarity 0.46
- didlogic · ph · 2026-10-28 · 1 upvotes · similarity 0.45
- TTSLab · hn · 2026-02-23 · 5 upvotes · similarity 0.45
- TrueScribe · hn · 2026-10-01 · 5 upvotes · similarity 0.45
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