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Switching via phone call

SIP Ring Switch on Asterisk telephone system

SIP Ring Switch opening garage door

Voice-over-IP access is now replacing analog telephone connections. This has had the effect of melding telephony and network technology. Even mature PBX systems no longer take up entire rooms, but rather run on normal PCs and even on embedded systems like Raspberry Pi. This is made possible for example by software packages such as the Open Source Project Asterisk and SIP protocol (Session Initiation Protocol).

The SIP Ring Switch can be connected to the telephone system over the local network just like a VoIP telephone - just that the SIP Ring Switch does not open a voice connection when a call comes in , but rather initiates a switching action.

Step 1 - Connect the SIP Ring Switch


To operate the SIP Ring Switch a supply voltage of between 12V and 24 VDC or a corresponding power supply is required.

The connected supply voltage is switched to the corresponding output terminal in the ON state.

If a potential-free contact is required for control, e.g. if the bell is operated with an AC voltage, a coupling relay can be inserted.

Wiring of the SIP Ring Switch should be performed only by trained persons.

SIP Ring Switch wiring
The supply voltage is connected to Terminal 1 and GND on Terminal 3.
For Output 0 for example to Terminal 4, with the associated ground on Terminal 5.


The SIP Ring Switch must be connected to a free LAN terminal on the Fritzbox.

SIP Ring Switch LAN connection

Step 2 - Configure network

For the Asterisk telephone system to be accessed reliably, the SIP Ring Switch should always have a static IP address assigned.

In networks where IP addresses are assigned by DHCP, the address must be either fixed assigned in the DHCP system or lie in an IP range which is excluded from DHCP address assignment.

To assign the SIP Ring Switch a static IP address, use a Windows PC to start the
WuTility tool.

SIP Ring Switch IP assignment

Click on the IP address icon to start the dialog for IP address assignment. Enter here the desired IP address and click on the Next button.

SIP Ring Switch IP assignment

The SIP Ring Switch can now be reached at the assigned address.

Step 3 - Set up an extension in the Asterisk telephone system

For the SIP Ring Switch to be able to switch using a call number, there must be an extension created and configured in the telephone system for each desired switching action.

In the menu branch Applications select Extensions. Here click on the + Add Extension and there set up a new extension for the desired switching action.

SIP Ring Switch and Asterisk

Step 4 - Configuring the SIP Ring Switch

SIP basic settings

Now the SIP Ring Switch must be configured for registering on the Asterisk telephone system. Log in as Administrator on the web interface of the SIP Ring Switch and select Communications paths>>SIP.

SIP Ring Switch and Asterisk

Under SIP basic settings enable SIP access and enter the access data for your Asterisk telephone system. For Password the password for the extension must be entered. Then confirm by clicking on the Apply button.

Extension inventory

To add a new extension to be represented by the SIP Ring Switch, click on the Add button.

SIP Ring Switch and Asterisk

The SIP address is entered using the following syntax:

<User name>@<IP address of the telephone system>

For SIP type the SIP procedure used by the telephone system must be selected. Confirm you entries with Apply so that they can be entered in the inventory list.

SIP Ring Switch and Asterisk

caller inventory

To prevent random or unauthorized calls to trigger switching events, callers can be listed in the caller inventory who have permission to access the SIP Ring Switch.

There are two ways to do this:

Learn new call numbers

Click on the Scan button.

SIP Ring Switch and Asterisk

Click on Start and then call the associated extension.

SIP Ring Switch and Asterisk

The SIP Ring Switch uses Scan mode to collect the SIP addresses for all incoming calls until you click on Stop.

SIP Ring Switch and Asterisk

Highlight the SIP addresses you with to add to the caller inventory.

Add new call numbers by entering

To manually enter an SIP address, click on the Add button.

SIP Ring Switch and Asterisk

Enter the desired SIP address. Syntax is important here:

<sip:Call number@IP of the telephone system>

Using "*" as a wildcard enables any desired number to be allowed. After clicking on Apply the SIP address is added to the inventory list.

SIP Ring Switch and Asterisk

Clicking on the Edit button allows you to modify the entries, for example to assign a meaningful alias name.

Configure output for pulses

Commonly used barrier controllers require a brief control pulse for opening and closing. To have an output reset automatically, for example 2 seconds after activation, it must be configured for Pulse mode.

In the SIP Ring Switch configuration select Basic settings>>Outputs and for the desired output Advanced Settings. There enable Pulse Mode and for pulse duration enter 2000 ms.

SIP Ring Switch and Asterisk

Set up switching action

Now you must specify which output should switch when a call comes in on which extension. In the menu tree click on Actions and then on the Add button.

SIP Ring Switch and Asterisk

As the trigger select SIP. Select the extension to be used for triggering the action when a call is made. In the inventory list determine who should switch via call. Finally, select as initiator Switch Output > Switch output on this Web-IO and then the output to be switched. As switching state ON must be enabled.

After confirming by clicking on the Apply button the SIP Ring Switch is ready for switching.

Application examples

  • Application

    SIP Ring Switch - Switching via SIP
    Phone call opens garage door

    Remote control using SIP Ring Switch and Fritz!Box

  • Application

    SIP Ring Switch - Switching via SIP
    Additional ring tone on Fritz!Box

    SIP Ring Switch switches "filling station bell"


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