---
title: "Connect via SIP trunking"
description: "Import a carrier number, configure routing and verify signaling and audio."
url: "https://documentation.verbatik.com/deploy/sip/"
---

# Connect via SIP trunking

Import a carrier number, configure routing and verify signaling and audio.

SIP trunking keeps a phone number at its current carrier while routing voice calls to Verbatik. There are two destinations to understand:

| Direction | Destination |
| --- | --- |
| Carrier → Verbatik, for incoming calls | Verbatik **inbound SIP URI** shown in the setup guide |
| Verbatik → carrier, for outbound calls | Your carrier’s **Termination URI** |

Do not enter Verbatik’s inbound URI in the carrier termination field.

## Collect the carrier settings

Ask the carrier for the number in E.164 format, termination URI, trunk username/password or IP-authentication requirements, transport, supported codecs and incoming signaling CIDRs. Ask whether registration is required. Elastic/FQDN trunks generally use routing without SIP registration; enable registration only when your carrier requires it.

## Import the connection

1. Open **Phone numbers**, choose the add menu, then **Connect to your number via SIP trunking**.
2. Select the provider preset: Twilio, Telnyx, Vonage or custom.
3. Enter **Phone Number**. Prefer **Format to E.164**; use **Keep raw input** only when the carrier requires its own number format.
4. Enter the carrier **Termination URI**, **SIP Trunk User Name**, **SIP Trunk Password** and optional **Nickname**. These are trunk credentials, not general account API credentials.
5. Under **Additional settings**, select matching **Codecs** (PCMU, PCMA or G.722), **Outbound Transport** (TCP, UDP or TLS), port, registration/realm where required, and the carrier’s inbound CIDRs. Additional numbers can share the trunk.
6. Confirm that you control the numbers and trunk. Save, then assign published inbound/outbound agents as appropriate.
7. In the carrier console, route the numbers’ origination/inbound destination to Verbatik.

## Production endpoint

Use the endpoint displayed in the app. The current production TCP endpoint is `sip:sip.verbatik.com:5060;transport=tcp`; TLS is `sips:sip.verbatik.com:5061;transport=tls`. RTP audio uses UDP ports 16384–16583. The HTTPS SIP control host is not a carrier destination.

The production gateway currently supports the Mistral voice route. Other managed engines on native SIP need worker configuration and acceptance before use. Twilio, Telnyx and Vonage networks have been reviewed. A custom carrier’s new ranges need Verbatik network admission as well as the saved app configuration; saving CIDRs alone does not open the gateway firewall.

## Verify before routing patients

A successful connection/OPTIONS check proves a signaling response only. Test an inbound call, two-way audio, caller ID, hangup, no-answer fallback and a staff transfer using numbers you control. Test outbound separately if enabled for your account. See the [go-live checklist](/deploy/go-live-checklist).