Welcome! This Node.js application integrates Asterisk 22 with the OpenAI Realtime API to provide a voice-based virtual assistant for SIP calls. It processes audio in real-time and displays user and assistant transcriptions in the console.

• Real-time audio processing with Asterisk and OpenAI.
• Console transcriptions for user and assistant speech.
• Clean resource management (channels, bridges, WebSocket, RTP).
• Configurable via config.conf (e.g., API key, prompt).

1. Install prerequisites:

   sudo apt update
   sudo apt install nodejs npm asterisk

2. Configure Asterisk:
   • Enable HTTP

     sudo nano /etc/asterisk/http.conf

     Add the following lines at the end of the file:

     enabled=yes
     bindaddr=0.0.0.0
     bindport=8088

   • Configure ARI

     sudo nano /etc/asterisk/ari.conf

     Add the following lines at the end of the file:

     [asterisk]
     type=user
     password=asterisk

   • Add dialplan

     sudo nano /etc/asterisk/extensions.conf

     Add the following lines at the end of the file:

     [default]
     exten => 9999,1,Answer()
     same => n,Stasis(asterisk_to_openai_rt)
     same => n,Hangup()

   • Configure SIP Extensions

     sudo nano /etc/asterisk/pjsip.conf

     Add the following lines at the end of the file to configure SIP extension 300 that can call 9999:

     [transport-udp]
     type=transport
     protocol=udp
     bind=0.0.0.0
     external_media_address=3.89.115.249  ; Required: Replace with your EC2 instance's public IP from AWS console
     external_signaling_address=3.89.115.249  ; Required: Replace with your EC2 instance's public IP from AWS console
     local_net=172.31.0.0/16  ; Optional: Adjust to your VPC CIDR if different
     
     [300]
     type=endpoint
     context=default
     disallow=all
     allow=ulaw
     auth=300
     aors=300
     direct_media=no
     media_use_received_transport=yes
     rtp_symmetric=yes
     force_rport=yes
     rewrite_contact=yes
     dtmf_mode=auto
     
     [300]
     type=auth
     auth_type=userpass
     password=pass300
     username=300
     
     [300]
     type=aor
     max_contacts=2

   • Restart Asterisk:

     sudo systemctl restart asterisk

3. Clone the repository and install dependencies:

   git clone https://github.com/infinitocloud/asterisk_to_openai_rt_community.git
   cd asterisk_to_openai_rt_community
   npm install

4. Edit config.conf in the project root and add your OPENAI_API_KEY in the designated field:

   OPENAI_API_KEY=
   

5. Run the application:

   node index.js

1. Make a SIP call to the configured extension (e.g., 9999).
2. Interact with the assistant (e.g., say "Hi", "What is your name?").
3. Check console for transcriptions:

   O-0005 | 2025-06-28T04:15:01.924Z [INFO] [OpenAI] Assistant transcription: Hello! I'm Sofia...
   O-0010 | 2025-06-28T04:15:08.045Z [INFO] [OpenAI] User command transcription: What is your name?
   

4. End the call or press Ctrl+C to stop.

• Error: OPENAI_API_KEY is missing: Verify OPENAI_API_KEY in config.conf.
• Error: ARI connection error: Check Asterisk (sudo systemctl status asterisk, port 8088).
• No transcriptions: Set LOG_LEVEL=debug in config.conf.
• Debug commands:
  • Asterisk logs: tail -f /var/log/asterisk/messages
  • Node.js debug: node --inspect index.js
• Wrong password on SIP registration: Ensure the SIP phone username is 300 and password is pass300. Verify the server IP matches your Asterisk instance.
• No audio: Ensure external_media_address and external_signaling_address in pjsip.conf match your EC2 public IP. Verify RTP ports (12000+) are open in EC2 security group and local firewall. Update asterisk.js external_host to use the server’s IP.

• Report issues with logs and steps to reproduce.
• Submit pull requests via GitHub.
• License: MIT (see LICENSE).