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Self-hosted voice AI agent for Asterisk/FreePBX

Details

External ID
48887276
Source
HN
Company
—
Product
Self-hosted voice AI agent for Asterisk/FreePBX
Website domain
github.com
Launched
July 13, 2026
Cohort
—
Upvotes
32
Upvotes percentile
0.7939068100358423
Tags
—
Fetched at
Sept. 7, 2026, 9:26 p.m.
Updated at
Sept. 7, 2026, 9:26 p.m.

Description

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.

Enrichment

Theme
voice AI agents and infrastructure
Vertical
Horizontal
Function
Agent / copilot
Audience
B2B
AI stance
AI-native
Project type
Commercial product
Normalized one-liner
voice ai agent for asterisk and freepbx
Manually corrected
False

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 — 157 launches clear the similarity bar, closest 8 shown.

Attention rank: #29 of 158 (itself plus its competitors, highest first — normalized so YC and Product Hunt are compared fairly).

Launched 248 days after the earliest competitor.

Other launches for this product

Same idea, different domain

Nobody's really built a agent / copilot tool for Agriculture yet.