Hello FreePBX Community!
I had been working in a ARI app for pure Asterisk 20 that connects calls from extensions and Sip Trunks to OpenAI RealTime models, that reduce substancially the latency (to 1 sec aprox.) between send and receive answers making felling the conversation more fluid and natural.
Here is the original post where I was building the ARI App:
Now, I had completed the Module for FreePBX 17 that integrates and manages the ARI app:
So I would like to share it with you the Community as Free source. You can see a demo and repo github in the link.
Thanks.
Spam:
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