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.
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, you give them an extension number. Astradial then:- Creates a SIP endpoint on the Asterisk PBX server
- Generates a unique SIP password
- 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 |
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 |
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.
Making your first call
Once your SIP phone is registered:- Call another extension — Dial the extension number (like 1002) to call a colleague
- Call an external number — Dial the full phone number (with country code if needed). This requires an outbound trunk to be configured.
- Receive a call — When someone calls your phone number (DID) and it is routed to your extension, your phone will ring
Troubleshooting
My SIP phone won't register
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
I can register but calls have no audio
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
Calls connect but the other person can't hear me
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_addressconfigured in Asterisk

