Overview
After creating your Telepath connection, configure your telephony carrier to route calls through Telepath’s infrastructure. All carriers require:- Origination URI:
sip:{your-sip-username}@sip.telepathvoice.com - SIP Username: From your Telepath connection
- SIP Password: From your Telepath connection
- Protocol: UDP or TLS (both supported)
- Codecs: G.711 (PCMU/PCMA) or G.722 (HD audio)
Twilio
Prerequisites
- Active Twilio account with phone number
- Twilio SIP trunk capability
Step-by-Step
- Log into Twilio Console
- Navigate to Voice → Trunks
-
Create a new trunk or edit existing:
- Name: Give your trunk a descriptive name
- Region: Select closest to your users
-
Configure Origination:
- Click Origination in the left sidebar
- Under “Origination SIP URI”, enter:
- Set protocol to TLS (recommended) or UDP
- Save
-
Set SIP Authentication:
- Click Credentials in the left sidebar
- Create or select SIP credentials:
- Username: Your Telepath SIP username
- Password: Your Telepath SIP password
- Save
-
Configure Inbound Dialing:
- In trunk settings, set where inbound calls go
- You can point to TwiML or a webhook
- Or route to a Twilio phone number
-
Test:
- Call your Twilio number
- Monitor in Telepath dashboard
Telnyx
Prerequisites
- Active Telnyx account
- Purchased phone number or DID
Step-by-Step
- Log into Telnyx Portal
- Navigate to SIP Trunks → Outbound Voice
-
Create or Edit Trunk:
- Name: Descriptive trunk name
- Tags: Optional (for organization)
-
Configure Origination:
- Set Outbound Voice URI to:
- Set Origination Server Address to:
- Set Outbound Voice URI to:
-
Add SIP Credentials:
- Click Credentials section
- Add authentication:
- Username: Your Telepath SIP username
- Password: Your Telepath SIP password
-
Enable Codecs:
- Ensure G.711 and G.722 are selected
- Disable any unsupported codecs
-
Assign to Phone Number:
- In Phone Numbers, assign this trunk to your DID
- Inbound calls will route through this trunk
Vonage (Nexmo)
Prerequisites
- Active Vonage account
- Vonage virtual number
Step-by-Step
- Log into Vonage Dashboard
- Navigate to Voice → SIP Trunks
-
Create New Trunk:
- Name: Descriptive name
- Origination Server:
sip.telepathvoice.com
-
Inbound URI:
- Set to:
sip://{your-sip-username}@sip.telepathvoice.com
- Set to:
-
Authentication:
- Username: Your Telepath SIP username
- Password: Your Telepath SIP password
-
Link to Virtual Number:
- In Virtual Numbers section
- Assign trunk to your number
- Confirm routing
Bandwidth
Prerequisites
- Active Bandwidth account
- Configured SIP account
Step-by-Step
- Log into Bandwidth Portal
- Navigate to Account → SIP Peers
-
Create SIP Peer:
- SIP Peer Name: Telepath connection
- Type: Termination
-
Configure Termination:
- Termination URI:
sip://{your-sip-username}:5060@sip.telepathvoice.com
- Termination URI:
-
Authentication:
- Enable SIP Credentials
- Username: Your Telepath SIP username
- Password: Your Telepath SIP password
-
Route Calls:
- Link your phone number to this SIP peer
- Inbound calls will route through Telepath
SignalWire
Prerequisites
- Active SignalWire account
- Purchased phone number
Step-by-Step
- Log into SignalWire Dashboard
- Navigate to Voice → Trunks → Inbound
-
Create Inbound Trunk:
- Name: Telepath trunk
- URI:
sip://{your-sip-username}@sip.telepathvoice.com
-
Set Credentials:
- Username: Your Telepath SIP username
- Password: Your Telepath SIP password
- Enable TLS
-
Configure Endpoints:
- Point to your application endpoint
- Or set up call routing logic
Plivo
Prerequisites
- Active Plivo account
- Plivo phone number
Step-by-Step
- Log into Plivo Console
- Navigate to Voice → SIP Endpoints
-
Create SIP Endpoint:
- Name: Telepath Connection
- SIP URI:
{your-sip-username}@sip.telepathvoice.com
-
Set Trunk:
- In Endpoints, create outbound rule
- Trunk: Create new trunk pointing to
sip.telepathvoice.com - Credentials: Your Telepath SIP username/password
-
Link to Phone Number:
- Assign endpoint to your phone number
- Inbound calls route to Telepath
Codec Configuration
All carriers support both standard and HD audio:G.711 (Narrowband)
- Sample Rate: 8kHz
- Bandwidth: Lower (8kbps)
- Compatibility: Universal support
- Quality: Standard voice quality
G.722 (Wideband/HD)
- Sample Rate: 16kHz
- Bandwidth: Higher (16kbps)
- Compatibility: Most modern systems
- Quality: Enhanced clarity
Verification
Test Your Setup
-
Confirm in Telepath Dashboard:
- Connection shows Active
- Recent calls appear in call log
-
Make a Test Call:
- Call your phone number
- Should connect to your AI agent
- Verify natural conversation
-
Check Metrics:
- View call details in dashboard
- Verify Carrier Lag and AI Latency
Troubleshooting Connections
Calls not routing:- Verify origination URI is correct
- Check SIP username/password match
- Ensure carrier is using correct protocol (UDP/TLS)
- Try switching codecs (G.722 vs G.711)
- Check network quality on carrier side
- Review Carrier Lag metrics in dashboard
- Double-check credentials
- Ensure spaces are trimmed
- Regenerate password if uncertain

