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Ozeki Dialer
VoIP Service Provider VoIP Service Provider | Product Information Product Information

Connect Analog Phone lines to Ozeki Dialer

If you would like to add analog phone lines to your Ozeki Dialer, you will need an Analog Telephone Adapter (ATA) device. As Ozeki Dialer uses the VoIP network to make calls, an ATA device is required to translate the digital VoIP signals to analog. In the following I would like to show you how to setup a PSTN connection in your Ozeki Dialer software.

Step 1 - Create new connection

On the 'Quick Start' page select 'Step 3 - Connections'.

Figure 1 - Create new connection

Step 2 - Select connection details

On the new page you have to select 'Create new connection'. Then choose 'Telephone'.

Figure 2 - Create 'Telephone' connection

At connection types select the 'PSTN Gateway' connection.

Figure 3 - Choose 'PSTN Gateway' connection

Step 3 - General tab information

At the 'General' tab you can setup your PSTN Gateway connection settings.
In the 'Connection name' menu you can add a unique name for your connection so it will be easier to differintiate it from the other connections.
If you would like to connect on startup or reconnect immediately when connection is lost make sure to check 'Connect on startup:'.

At 'SIP Settings' there is the 'PSTN gateway IP address'. Provide the IP address of the gateway which will forward your call to the phone you dialed. Make sure to also provide the 'Port' number of your PSTN gateway.

Figure 4 - General tab

Step 4 - Network, Transfer, Times and Log details

Click on 'Net' tab and provide information about your network adapter so it can be connected to the PSTN network.

Figure 5 - Network settings

At 'Transfer' you can transfer the calls to a live person if DTMF response is received. Check the box next to 'Enable call transfer if DTMF response is received.' and add the phone numbers below to the suitable keypad.

Figure 6 - Transfer details

At 'Times' tab you can specify the time and day interval when communication is allowed.

Figure 7 - Times tab

In the 'Log' tab your calls can be logged. There is a possibility to enable and disable logging by checking the box next to 'Enable logging'.

  • 'File name:' Provide a name for your log file.
  • 'Directory:' You can add where you would like to save the log file on your computer.
  • 'Max. size:' You can maximize the size of the log file. After it reached the size you added it will rename the current log file and open a new one.
  • 'Rotate:' You can specify the number of log files that will be created. Typical values are 4-10.
  • 'Zip:' If you select this the log files will be compressed before the next log file created. You can save disk space on your computer.

Figure 8 - Logging information

Step 5 - How to dial

If a green pipe appeared your connection has been setup.
Click on the 'Open' button to see the event log, modify the configuration or make a test call.

Figure 9 - Click on 'Open'

Step 6 - Make a test call

Select 'Call' in order to compose and then send your test call.

Figure 10 - Select the 'Call' button

In the 'Compose' tab you can create the content of your message.

  • 'Number to call:' Type the number you would like to call.
  • In the white box, where you can see the 'Test message' you can write the message you would like to play for the called client. When you are finished with writing your message click on the 'Convert text to speech' button. After that click on 'Start test call'.

Figure 11 - How to make a test call

Go to 'Events' or 'Log' tab where you can check whether your call was successful.

Figure 12 - Event log


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