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

# SIP extensions

> How SIP extensions work in Astradial — what they are, how to configure them, and how to connect a phone.

## What is a SIP extension?

A SIP extension is a short number (like 1001, 1002, 1003) that identifies a person or device in your phone system. When someone calls extension 1001, it rings the phone app registered to that extension.

**SIP** stands for Session Initiation Protocol. It is the standard technology that makes internet phone calls work. When you register a SIP app (like Zoiper) with your extension credentials, your device connects to the Astradial server and can make and receive calls.

## How extensions are assigned

When you [create a user](/users/add-users), you give them an extension number. Astradial then:

1. Creates a SIP endpoint on the Asterisk PBX server
2. Generates a unique SIP password
3. Deploys the configuration so calls to that extension number are routed correctly

## Extension number conventions

You can use any number for extensions, but here are common patterns:

| Range       | Typical use                |
| ----------- | -------------------------- |
| 1000 - 1099 | Individual user extensions |
| 5000 - 5099 | Call queue extensions      |
| 9000 - 9099 | Conference rooms           |

<Tip>
  Keep your extension numbers consistent. For example, use 1001-1099 for people and 5001-5099 for queues. This makes it easy to remember.
</Tip>

## Connecting a SIP phone

You can use any SIP-compatible phone app or hardware phone. Here are the most popular options:

### Software phones (free)

| App          | Platform                          | QR Support |
| ------------ | --------------------------------- | ---------- |
| **Zoiper**   | iOS, Android, Windows, Mac        | Yes        |
| **Opal**     | iOS                               | Yes        |
| **Linphone** | iOS, Android, Windows, Mac, Linux | No         |
| **MicroSIP** | Windows                           | No         |

### Hardware phones

Any SIP-compatible desk phone works with Astradial. Popular brands include:

* Yealink
* Grandstream
* Polycom
* Cisco SPA series

## Registration settings

When configuring a SIP phone manually, use these settings:

| Setting             | Value                                               |
| ------------------- | --------------------------------------------------- |
| **Protocol**        | SIP                                                 |
| **Server / Domain** | Your Astradial server IP or hostname                |
| **Port**            | 5060                                                |
| **Transport**       | UDP                                                 |
| **Username**        | The Asterisk endpoint name (shown in SIP QR dialog) |
| **Password**        | The SIP password (shown in SIP QR dialog)           |
| **Auth Username**   | Same as Username                                    |

<Note>
  The SIP username is the **Asterisk endpoint name**, not the user's login username. You can find it in the SIP QR Code dialog for each user.
</Note>

## Making your first call

Once your SIP phone is registered:

1. **Call another extension** — Dial the extension number (like 1002) to call a colleague
2. **Call an external number** — Dial the full phone number (with country code if needed). This requires an outbound trunk to be configured.
3. **Receive a call** — When someone calls your phone number (DID) and it is routed to your extension, your phone will ring

## Troubleshooting

<AccordionGroup>
  <Accordion title="My SIP phone won't register">
    * Double-check the server IP and port (5060)
    * Make sure you are using the correct SIP password (not the login password)
    * Check that UDP port 5060 is open on your server firewall
    * Try using the SIP QR code instead of manual entry
  </Accordion>

  <Accordion title="I can register but calls have no audio">
    * Make sure UDP ports 10000-10100 are open on your server
    * Check that your server's public IP is correctly configured in Asterisk
    * If using Docker on Mac, SIP audio does not work locally — use a remote server
  </Accordion>

  <Accordion title="Calls connect but the other person can't hear me">
    * Check your microphone permissions in the SIP app
    * Make sure you are not on mute
    * This is usually a NAT issue — your server may need `external_media_address` configured in Asterisk
  </Accordion>
</AccordionGroup>