Development of Tool for Healthcare Office / Transcription & Speaker Identification

AbePA · April 11, 2025, 12:04pm

Hi

Im Looking for advice on implementing real-time call analysis with Asterisk/FreePBX: I’m exploring options for a business intelligence solution that can capture and analyze phone conversations in real-time for a healthcare office. Ideally, I need to stream audio to a transcription service while the call is in progress. Has anyone implemented something similar using Asterisk ARI or FreePBX modules? I’m particularly interested in:

Most reliable approach for real-time audio streaming from Asterisk
Recommended integration methods with transcription APIs
Existing modules or solutions that might handle part of this workflow
Potential latency issues and how to minimize them

Any guidance on implementation approaches or developers experienced with similar projects would be greatly appreciated.

Thanks!

Abe

david551 · April 13, 2025, 10:43am

See Asterisk With Ultravox.ai

This is a variation on an FAQ.

nshmyrev · April 13, 2025, 11:32am

ARI is recommended, the integration is like 50 lines of code, you can check example with Vosk here:

the code itself is here:

github.com/alphacep/vosk-server

client-samples/asterisk-ari/vosk_ari.py

master

#!/usr/bin/python3

import anyio
import asyncari
import logging
import aioudp
import os
import vosk
import array

ast_host = os.getenv("AST_HOST", '127.0.0.1')
ast_port = int(os.getenv("AST_ARI_PORT", 8088))
ast_url = os.getenv("AST_URL", 'http://%s:%d/'%(ast_host,ast_port))
ast_username = os.getenv("AST_USER", 'asterisk')
ast_password = os.getenv("AST_PASS", 'asterisk')
ast_app = os.getenv("AST_APP", 'hello-world')


model = vosk.Model(lang='en-us')
channels = {}

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Development of Tool for Healthcare Office / Transcription & Speaker Identification

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