---
title: "Configuring Your Trunk With FreePBX 17 - IP Trunk"
slug: "standard-article-1"
updated: 2025-03-27T19:39:48Z
published: 2025-03-27T19:39:48Z
canonical: "docs.wiretaptelecom.com/standard-article-1"
---

> ## Documentation Index
> Fetch the complete documentation index at: https://docs.wiretaptelecom.com/llms.txt
> Use this file to discover all available pages before exploring further.

# Configuring Your Trunk With FreePBX 17 - IP Trunk

![image(6)](https://cdn.document360.io/902e9780-a72e-4ea2-92b0-fee80121007e/Images/Documentation/image(6).png)

## **Configuring Your Trunk With FreePBX 17 - IP Trunk (PJSIP)**

---

### **About FreePBX 17**

FreePBX 17 is the latest version of the powerful open-source PBX platform, designed to provide flexible and user-friendly communication management. With improved performance, security enhancements, and new features, FreePBX 17 makes it easier than ever to manage your phone system.

This guide will walk you through setting up your SIP trunk using PJSIP, a modern and versatile SIP channel driver in FreePBX. Compared to the older Chan_SIP driver, PJSIP offers better performance, enhanced NAT handling, and improved support for multiple registrations. By following these steps, you'll ensure a smooth and reliable connection for your inbound and outbound calls.

**Supporting Documentation:**

[FreePBX documentation](https://sangomakb.atlassian.net/wiki/spaces/DocHome/overview?mode=global#all-updates)

---

### **Pre-requisites**

- [Download](https://www.freepbx.org/downloads/) and install FreePBX17
- Configure basic settings for your FreePBX (Administration and Firewall settings)
- [Create a SIP trunk](/docs/creating-a-sip-trunk) on the Wiretap Portal and assign DID(s) to your trunk.

## **Step 1: Configure SIP settings**

Here we’ll add our network settings

1. Navigate to **Settings** —> **Asterisk SIP Settings** to verify your network settings.
2. Make sure to enter the **external** and **local** network addresses in both the **General SIP Settings** and **PJSIP Settings.**
3. Click **Submit** and **Apply Config**.

![](https://cdn.document360.io/902e9780-a72e-4ea2-92b0-fee80121007e/Images/Documentation/Screenshot 2025-03-20 165914.png)

## **Step 2: Create a SIP Trunk**

Next, we’ll configure our SIP Trunk using chan_pjsip.

1. Navigate to **Connectivity** —> **Trunks** and select **Add SIP (chan_pjsip) Trunk**.

![](https://cdn.document360.io/902e9780-a72e-4ea2-92b0-fee80121007e/Images/Documentation/Screenshot 2025-03-21 110457.png)

1. Next, we’ll input our general settings. Here, I’ve given my trunk a name and set the outbound caller ID.

![](https://cdn.document360.io/902e9780-a72e-4ea2-92b0-fee80121007e/Images/Documentation/Screenshot 2025-03-21 111220.png)

> [!NOTE]
> A Note About Caller ID
> 
> If you don't configure an Outbound CID for your trunk, you'll need to set one on each applicable extension instead. Without setting a caller ID at either the trunk or extension level, your calls will reach us lacking a valid caller ID.

1. Then we’ll set up our Dialed Number Manipulation Rules. Select the **pjsip Settings** tab, then the **General** tab.
  1. You can use the **Dial Patterns Wizard** for this step to create custom rules depending on your use case.
  2. Below are some basic dial patterns for US numbers:

![](https://cdn.document360.io/902e9780-a72e-4ea2-92b0-fee80121007e/Images/Documentation/Screenshot 2025-03-21 141017.png)

1. Now we’ll move on to our pjsip settings. Select the **pjsip Settings** tab, then the **General** tab.

![](https://cdn.document360.io/902e9780-a72e-4ea2-92b0-fee80121007e/Images/Documentation/Screenshot 2025-03-21 142810.png)

1. Since we’re using an IP based trunk, we do not need to include registration details.

Be sure to specify:

- **Registration:** None
- **SIP Server:** srv.globalsbc.core-trunk.com
- **Sip Server Port**: Leave blank if using SRV record. The SRV record defaults to **port 5060**.
- **Context:** from-pstn

## **Step 3: Apply Changes**

1. Select **Apply Config**
2. Add Extensions, Inbound and Outbound Routes