Asterisk, Linux GNU/GPL IPBX


What is it ?

Here a few times already, I was interested in VoIP (Voice on IP).

This extremely tempting technology makes it possible to exchange the voice and the image via a local network IP or Internet.

I am well amused with IP-phone sets (hardware),  "softphones" (small software able to replace an hardware IP-phone set  and runing in any OS usual), and VoIP-gateways which allows the connection of analogical telephone lines two wire (POTS) or of the telephone terminals and faxes a network IP as well as the conversion of the analogical audio signals via a "codec" (coder decoder) traditional in the world of VoIP.

However, none the solutions then validated satisfied me because, in spite of the remarkable performances and the quantity of functions embarked in some of the gateways VoIP, I could not claim to replace the sophisticated functions of automatic exchanges (PABX) top-of-the-range traditional in the large companies.

Here are a few years, I had fallen on a project from collective development of which the goal was, precisely, to have an IPBX (IP compatible PABX).  Here is finally the solution that I sought, features full, with high level functions, completely reconfigurable with wish, compatible with the totality of the usual means of telecommunications, which it is side "coppers" (operator of traditional telephony) and VoIP side:

This project had absorbed another older, initialized by a radio ham, WBÑIL Jim Dixon, helped by a second, WA6ZFT Steve Rodgers:

And moreover, the sources are available and used for the installation, compilation obliges! 

It is enough to have a standard PC, some PCI slots if need for interface boards with conventional telephony, GSM, or transceivers (repeaters). 

For the world of VoIP, it is enough to have a NIC (network interface controler) and an Internet access to reach the whole world!

I thus started to "assemble" my personal home IPBX, in order to share my telephone lines with several in-house IP-phone sets with transfer, to have comedian mail features with redirection via email and other remarkable functions,

Then, I started to plan to propose to use this machine for VoIP traffic for our community of radio hams...

It is possible to have a great number of users or "subscribers" (only related to the band-width network) and to leave at disposal certain services of my machine:
- a conference room numbered 7388
- my VoIP station in my shack:  77001
- other correspondents so some are interested
- one voice mail for each one
- an address email for each one
- a redirection of the vocal messages on the email of its choice
- an access to a station of radiocommunication in our bands (testing)
- an access to receivers (testing)
- and I forget some...

How to start ?

You need, with the choice: 
- a VoIP telephone set under protocol SIP or IAX2 (native protocol of Asterisk less consuming bandwidth)
- a free softphone: (remote radio console with PTT)
- a PC with Asterisk!

I will accompany you for your first test with me.  We will use iaxComm seen higher, for the simple reason which you will be able to alone play while connecting you in "anonymous guest" on my Asterisk. 

You will have to place a telephone call with at least two digits so that that goes (any numbering will do the business, like, for example, "123"). 

Caution:  you will not have access to nothing, a nice American girl will thank you for having contacted me, it is all!  It is already much   ; -) 

This will want to say that you will have succeeded in crossing your local area network towards mine while passing by Internet, and you will be able to judge the audio quality of the GSM codec used (of others are active, but not for guest "GUEST"). 

Here what you must enter in parameters during the first launching of your iaxComm softphone, or, if you have missed something in the different menus, by selecting "Options" and "Accounts":

You will note well that one should not input any password (empty field)! 

Caution:  take care to let pass traffic IP of port 4569 UDP (attention with your firewall/router)!

If you want to go further, it will be necessary to contact me and ask me an account code for your callsign   : -) 

The list of the users (except "red list" on request) is there:


Small addition for 2006:  You can from now dial what follows as a guest for tests.

- 600:  echo test. Makes it possible to consider latency caused by the codecs and transport while forwarding you your own voice. 
- 1000:  1kHz 0dBm signal test 
- 1009:  my IP-Phone in my shack

Big adding during June 2006 : my Asterisk box is now incorporated into the famous AllStar Link :

It is now possible to connect my Asterisk server to another of its congenerics “elsewhere”, and to interconnect two or several remote sites to form one common radio network. Any radio under operation in one of the zones will be also in range of those of the other zones

 A "Mini How To" (simplified user manual) is available here :

To be continued...

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73, de F6HQZ François BERGERET

Dernière modif le : 08/27/06