Skip to main content

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:
  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:
RangeTypical use
1000 - 1099Individual user extensions
5000 - 5099Call queue extensions
9000 - 9099Conference rooms
Keep your extension numbers consistent. For example, use 1001-1099 for people and 5001-5099 for queues. This makes it easy to remember.

Connecting a SIP phone

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

Software phones (free)

AppPlatformQR Support
ZoiperiOS, Android, Windows, MacYes
OpaliOSYes
LinphoneiOS, Android, Windows, Mac, LinuxNo
MicroSIPWindowsNo

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:
SettingValue
ProtocolSIP
Server / DomainYour Astradial server IP or hostname
Port5060
TransportUDP
UsernameThe Asterisk endpoint name (shown in SIP QR dialog)
PasswordThe SIP password (shown in SIP QR dialog)
Auth UsernameSame 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:
  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

  • 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
  • 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
  • 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