VOIP Engineer Needed With Experience On Asterisk, FreeSwitch, SIP

I need an experienced VOIP and SIP Engineer. I have developed a custom AI Voice Calling Bot and it is currently connected with Twilio. Whereas my plan is to connect this AI Bot with almost any SIP provider.
When i call from Twilio using my AI Bot, then it works correctly.
Here i am needing you

1. I have installed Asterisk on my AWS. It is fully configured and making outbound calls.
2. My AI Calling Bot works with web hooks, and I have created another instance where I have created an outbound webhook in Node. And the AI Calling bot's webhook is placed in this code
3. When I call from Asterisk, the outbound call works, but there is a sharp noise in the call, and nothing else. My AI Bot should be listening in the call, but i only hear a sharp noise
4. I know this is due to mis-sampling, bit rates, and can be some other issue. I tried a lot, but I am unable to fix the noise.

So, your job is to fix the noise in the call and make my AI BOT listen.

This is not a job for beginners or intermediates. Only Experts needed.
If you know how to fix and can do the job, then write "Asterisk" at the top of your bid, so i can see that you have read the project description.