Monday, June 23, 2014

Setting Up Virtual IPpbx Systems - By Peter Banda

Peter Banda has been kind enough to share more on setting up vitual IPpbx systems in his latest guest post here on Network Fun!!!  Thank you Peter.  ~~Shane Killen

                                   Setting up virtual ippbx systems - By Peter Banda

Unlike traditional phone systems, voice over internet protocol (VoIP) can use existing computer network cabling, eliminating the need to have a separate network for voice communications. In this guide I will show how to setup VoIP phone systems using axon virtual ippbx for the server and aastra 6731i for the client. Axon by NCH only releases commercial versions but you can download 3cx that has a free version with a limitation of about four simultaneous calls or even free linux based virtual ippbx distributions. 3cx or Axon can be downloaded from their official websites.

When axon ippbx has been downloaded, run the exe file and you will be presented with a license agreement window. Read it , click the ‘ I accept the license terms’ radio button and click next. The next window may come with options to install additional programs (depending on which version you downloaded), but ours is only a basic installation, so we are not going to install any additional programs. Leave all options unchecked, but if you don’t have a physical VoIP phone, you may check the express talk softphone. Voip softphones work as well as the hardware based phones.  On this windows click finish.

Now its time to make configurations, on the number of extensions, enter the total number of extensions you think you can use. In axon you may increase the number of extensions later, but other phone systems does not allow adding extensions after setting up. That is why you have to at least enter a number that has some extensions reserved for future use. Compose and enter an administrator password to use anytime you want to manage the server. Default username for the server administrator is Admin, you might want to change it. Below that there is an option to configure email for notifications, I will leave that one for now, click next. From here axon attempts to open incoming ports for SIP and RTP to allow calls from the internet. If your setup is for internal use only, ignore this. But if there is need  to make and receive calls to and fro the internet then you need to make sure that UDP port 5060, 5070, 8000, 80002, 8004, 8006 ... 8020 are open on your firewall. In this post I will stick with the internal use and reserve external dialling for future post.

Axon automatically searches for firewall or routers on networks, if they are detected between your network and the internet, it will ask you to put some settings for audio routing. Then there is a windows for additional information, click ‘No thanks’. Check open axon’s web control on the next window then click finish. Axon voip server now starts, enter your user name and password to login. If it doesn’t start, double-click on the axon server icon then click on web control to login.

In the axon control panel, click on extensions and groups because we will have to setup two SIP accounts for two aastra phones to insure that the server is working. Select extension 101 and click edit (an icon with a the pencil and paper), leave the extension number as it is. Edit the display name, type ‘test phone 101’ without quotes. Type voiptest101 in the password field. This is for testing only, when making final configurations, choose meaningful display names and strong passwords. Repeat  this for the next extension number 102.

Power up the two aastra phones and connect them to your network. I will assume that the phones have no any settings on them or are reset to their factory default settings. If you have a dhcp server on the network, the phones should be able to get dynamic ip address.  If there is no dhcp server then you have to give them static ip addresses. To check if they received ip addresses, press the settings/options button on the phone. The button has a spanner symbol on it. Navigate down to ‘phone status’ the ‘ip and mac addresses. Press right arrow to see the ip address. Open a web browser on a computer that is connected to the same network, type the phone ip address on address bar of the browser to access its webUI. Enter admin and 22222 for user name and password. To give the phones static ip addresses, press options button, navigate to admin menu and enter 22222 as password. (22222 is the default aastra 6731i password). Go to network settings, dhcp settings, change it to OFF, press right arrow then you will be prompted to restart the phone. After restart go back to network settings, this time proceed to ip address, type the ip address and subnet mask. Restart the phone if you are prompted. Open a web-browser on your computer and enter the phone’s ip address.

Since we already have two extensions on the server, we have to give the phones the extensions, one extension per phone. After entering admin user name and password on the phone’s webUI, click Line 1,
1.       Screen name                     test phone 101 (and 102 on the other phone)
2.       Phone number                                 101
3.       Caller id                                101 (can be an extension or name)
4.       Auth name                         101
5.       Password                            voiptest101
6.       BLA                                        -
7.       Line mode                           generic

Auth name and password should be the same as that on the server, otherwise the phones won’t work.

Go to basic SIP settings,
1.       Proxy server                      type the FQDN or ip address of the server where axon is installed
2.       Proxy port                           5060
3.       Registrar server                                type the FQDN or ip address of the server where axon is installed
4.       Registrar port                    5060
5.       Register period                 180

Go to preferences and check directed call pick up option. Click global SIP, scroll down to RTP settings, check ‘the port is 3000’ and,
1.       Check force rfc2833 out of band dtmf
2.       Dtmf method                    rtp
3.       RTP encryption                 srtp disabled

Scroll down to advanced SIP settings,
1.       Enable MWI subscriptions
2.        Explicit MWI subscription period 1800
Click save settings at the bottom of the page.

Click preferences, go to ‘incoming intercom settings’, enable auto answer and play warning tone.

Repeat these settings on the other phone and try to make a call from one phone to the other.


  1. Thanks Peter and Shane♥ It's very useful for me.

    1. Hey Babak. good to hear from you my friend. Peter does a real good job on his articles. Very glad he does some postings for me. I know you are very busy Babak, but would love to see some more material from you on here. I know you have helped others with what you have written. :) I hope all is well with you.

  2. I'm going to write a new post soon


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