Newbie question - Could I use Asterisk for this?


I’m an Asterisk newbie and I’m not sure if I understand correctly this technology, so let me ask you for some help. I’m working on a VoIP solution, whereas the key and in fact the only one functionality should be the possibility to make a conference call for max. 10 users. I have a server located in hosting, there would be only software clients connecting to it (both Unix and Windows). Let me describe the facts and requirements:

  • server hosted at ISP, running Linux, quite powerful hardware
  • only software clients somewhere in Internet
  • most of clients has a public IP addresses, some are behind NAT
  • the goal is to make a conference call for max. 10 users
  • I don’t want to use any additional hardware
  • I want to record the actual conference call

Is the Asterisk software the right solution for me?



You will be able to use asterisk for you application.


First, thank you for your clear, detailed question which provides all the information needed to answer it. I wish all of them were like this :smiley: A good question deserves at least an attempt at a useful answer…

You will have no trouble at all using Asterisk for this.

A few things to consider that will dictate how the system is set up:

One thing to think about:
Who will be accessing the conference? If its the same bunch of guys, then each of their SIP phones (hardphones or softphones) will need an entry in sip.conf. If it’s random guys and you want anybody to be able to connect, then in [general] set the context= to something that can dial the conference room. then any random guy can dial sip:confnumber@yourserverip and be connected. You can also authenticate with a PIN code for the conference.
Another way to do it would be to link the server to a VoIP network like FWD or SIPphone. Then anybody on these networks could dial your systems number (a 5digit(FWD) or 7digit(SIPphone) access code) thru the VoIP network and be connected to you. There are also many local access numbers (from PSTN) that feed into FWD and SIPphone.
Lastly, will give you a free 360 area code DID (phone number). Set your server’s IP as the sip proxy. This system supports multiple channels. I once (for an event meeting) did something very similar to what you want to do except with IPkall- it took ~10 users dialing the same number just fine, and my P2-266 asterisk server conferenced them just fine :smiley:.

To record your conference, just add the ‘r’ flag to meetme in extensions.conf. IE,
exten => 1234,1,Meetme(1111,r,2222)
would, when you dial 1234, put you in conference 1111, which would be recorded, and make you enter the pin (2222) to get in.

Check out (wiki). Voip-info has just about everything there is to know, and asteriskdocs has the book Asterisk: The Future of Telephony for free download under creative commons license. You may not need the whole thing, but the section on dialplan design (extensions.conf) and setting up SIP channels may be useful for you.

Good luck!