Telepath

Carrier Configuration

Set up inbound SIP routing for Twilio, Telnyx, and other carriers

Overview

After creating your Telepath connection, you need to configure your telephony carrier to route inbound calls to Telepath's SIP endpoint. This page covers setup instructions for the most popular carriers.

All carriers require the following information from your Telepath connection:

  • Origination URI — where to send calls
  • SIP Username — for authentication
  • SIP Password — for authentication
  • Protocol — UDP or TLS
  • Codecs — G.711 (PCMU/PCMA) or G.722

Your Telepath origination URI follows this format:

text
sip:{your-sip-username}@sip.telepathvoice.com

Twilio

Prerequisites

  • A Twilio account with at least one active phone number
  • Access to Twilio Console → Voice → SIP Trunks

Instructions

  1. Log in to the Twilio Console

  2. Navigate to Voice → SIP Trunks and click Create a SIP Trunk

  3. Give your trunk a name (e.g., Telepath Production)

  4. Under Origination, click Add new Origination URI and enter:

    text
    sip:{your-sip-username}@sip.telepathvoice.com

  5. Under Authentication, add a Credential List with your Telepath SIP username and password

  6. Navigate to Phone Numbers → Manage → Active Numbers and click your number

  7. Under Voice & Fax, set Configure with to SIP Trunk and select your new trunk

Tip: Twilio trunk changes typically propagate within 2–5 minutes. If calls are still not routing after configuration, wait a few minutes and test again.

Telnyx

Prerequisites

  • A Telnyx account with at least one active DID (phone number)
  • Access to the Telnyx Mission Control Portal

Instructions

  2. Navigate to SIP Trunking → SIP Connections and click + Add SIP Connection

  3. Enter a connection name and select Credentials as the authentication method

  4. Set the Username and Password to your Telepath SIP credentials

  5. Under Inbound, set the origination URI to sip:{your-sip-username}@sip.telepathvoice.com

  6. Enable G.722 codec in the codec priority list (recommended) alongside G.711 PCMU

  7. Navigate to Numbers, click your DID, and assign it to this SIP connection

Vonage (Nexmo)

Prerequisites

  • A Vonage API account with a virtual number
  • Access to the Vonage API Dashboard

Instructions

  1. Log in to the Vonage API Dashboard

  2. Navigate to Voice → SIP Trunks and create a new SIP trunk

  3. Set the Termination SIP URI to sip:{your-sip-username}@sip.telepathvoice.com

  4. Under Authentication, enter your Telepath SIP username and password

  5. Enable the codecs G.711 PCMU and G.722 in your SIP trunk settings

  6. Link your virtual number to this SIP trunk via Numbers → Your Numbers

Bandwidth

Prerequisites

  • A Bandwidth account with an active phone number
  • Access to the Bandwidth Dashboard

Instructions

  1. Log in to the Bandwidth Dashboard

  2. Navigate to Applications → Voice and create a new application

  3. Set the Call Initiated Callback URL type to SIP

  4. Set the origination URI — Bandwidth requires explicit port 5060:

    text
    sip:{your-sip-username}@sip.telepathvoice.com:5060

  5. Set SIP credentials (username and password) for digest authentication

  6. Assign your phone number to this voice application in Numbers

SignalWire

Prerequisites

  • A SignalWire account with a project and at least one phone number

Instructions

  1. Log in to your SignalWire Space

  2. Navigate to SIP → SIP Endpoints and click New SIP Endpoint

  3. Enter your Telepath SIP username and password as the endpoint credentials

  4. Set the Caller ID and configure the SIP endpoint to forward to sip:{your-sip-username}@sip.telepathvoice.com

  5. Under Phone Numbers, assign your number and point it at this SIP endpoint

Plivo

Prerequisites

  • A Plivo account with an active phone number
  • Access to the Plivo Console

Instructions

  1. Log in to the Plivo Console

  2. Navigate to Voice → Trunk and click Add New Trunk

  3. Set the trunk name and termination URI to sip:{your-sip-username}@sip.telepathvoice.com

  4. Enter your Telepath SIP username and password under Authentication

  5. Go to Phone Numbers, select your number, and point it to this trunk

Codec Configuration

G.711 (Narrowband)

Parameter Value
Sample Rate8 kHz
Bit Rate64 kbps
Bandwidth8 kHz
Latency~1–2 ms
VariantsPCMU (μ-law, North America), PCMA (A-law, Europe/Asia)

G.711 is universally supported by all carriers and SIP equipment. It is the safe default if you are unsure which codec to use, though it provides narrowband audio quality.

G.722 (Wideband/HD)

Parameter Value
Sample Rate16 kHz
Bit Rate64 kbps
Bandwidth16 kHz
Audio QualityHD Voice (wideband)

Tip: Use G.722 when possible. The wider frequency range improves speech intelligibility and significantly increases AI speech recognition accuracy, resulting in better end-to-end conversation quality.

Verification

Test Your Setup

  1. From a regular phone, dial the number associated with your carrier trunk
  2. Listen for your AI agent's greeting message — this confirms audio is flowing end-to-end
  3. Check the Telepath Dashboard Call Log to confirm the call appears with a green status

Troubleshooting Connections

Calls not routing

  • Verify the origination URI exactly matches sip:{username}@sip.telepathvoice.com
  • Confirm the trunk is enabled in your carrier's dashboard
  • Check that the SIP credentials match those stored in Telepath

Poor audio quality

  • Ensure G.722 is in the codec priority list and listed before G.711
  • Check for packet loss on your carrier's network using the Telepath dashboard metrics
  • Consider switching your carrier protocol to TLS + SRTP for encrypted transport

Authentication failures

  • Double-check that the SIP username and password are entered correctly in your carrier — no extra spaces
  • If you recently rotated your Telepath SIP password, make sure you updated the carrier
  • Check the Telepath dashboard for 401 or 403 error codes in the call log

FAQ

Can I use multiple carriers with the same Telepath connection?
Yes. Multiple carriers can point their origination URIs at the same Telepath SIP username. All calls will be routed to the same AI agent connection.

My carrier is not listed here. Can I still use Telepath?
Yes. Any carrier that supports standard SIP trunking with digest authentication can be used with Telepath. Follow your carrier's documentation for configuring a SIP origination/termination endpoint and use the Telepath origination URI format above.

Can I change carriers after going live?
Yes. You can update the origination URI at your carrier or switch carriers entirely without making any changes to Telepath. Your Telepath connection and SIP credentials remain unchanged.

How long does carrier activation typically take?
Most carriers propagate SIP trunk changes within 2–10 minutes. Twilio and Telnyx are typically the fastest. If a trunk is not working after 15 minutes, contact your carrier's support team.