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Connect a VoIP Service Provider

Landline and GSM phones can be also called automatically via the internet with this type of connection. You will need a VoIP Service Provider to be able to call these devices. Calling analogue networks are supported too. Below I would like to show you how to setup a VoIP Service Provider connection in your Ozeki Bulk Messenger software.

Step 1 - Create new connection

Go to the 'Quick start' page and select 'Step 3 - Connections'.

Figure 1 - Create new connection

Step 2 - Create 'Telephone' based connection

After the new page appeared click on 'Create new connection'. Then select the 'Telephone' connection.

Figure 2 - Create new 'Telephone' connection

On the right in 'Connection details' window, select the middle one 'VoIP Service'.

Figure 3 - Choose 'VoIP service' connection

Step 3 - General tab information

At the 'General' tab you can provide connection details. The 'Connection name' is a unique name, it helps you to keep your connections organized, it is easier to differentiate them with different names. You can enable autoconnect and autoreconnect by checking 'Connect on statup'

At SIP Settings make sure there is a blue box next to 'Register'.

  • 'Display name': is your Caller ID. This number appeares on the phones you will call.
  • 'Username': This will be your username which is needed for logging into the system of the VoIP provider.
  • 'Register name': Usually username and register name can be the same but there are systems that requires a different one.
  • 'Password': This protects your connection to your VoIP Service Provider's system.
  • 'Domain': This is probably the most important data. Here you have to add your VoIP Service Provider's address.
  • 'Port': In most cases this is 5060. This specifies the SIP port number for the VoIP Service Provider.
  • 'Proxy': If the VoIP Service Provider has a proxy address, give it here. If it is not required you can leave it blank.

You can specify how many automated calls you would like to make at 'Simultaneous calls' and define the maximum ring time at 'Max. ring time', which may be limited by your VoIP service provider.

Figure 4 - General tab

Step 4 - Network, Transfer, Times and Log

If you go to 'Net' tab you can setup your network adapter which will be connected to the 'VoIP network'. Also 'Firewall settings' and 'Delivery speed' can be provided in this tab.

Figure 5 - Network settings

At 'Transfer' tab 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 tab

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

Figure 7 - Times tab

In the 'Log' tab you can log your calls. You can enable and disable logging by checking the box next to 'Enable logging'.

  • 'File name': Add 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 settings

Step 5 - How to dial

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

Figure 9 - Open the connection details page

Step 6 - Make a test call

Click on 'Call'.

Figure 10 - How to make a test call

  • 'Number to call': Here goes the number you want to call.
  • At the blank box you can write the message you would like to play for the called client. After you wrote your message you have to click on the 'Convert text to speech' button. And finally, click on 'Start test call'.

Figure 11 - Compose your message

Click on 'Log' or 'Events' and you can check if your call was successful.

Figure 12 - Event log

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