A few years back I had setup an asterisk server and even bought a compatible zaptel modem card for incoming/outgoing calling. At the time, I had wanted to provide a solution for PS3 party/lobby chat, a major advantage the Xbox has over the PS3. In Xbox, you’re able to chat with your group of friends who sign on while playing the same or different games. In PS3, you’re faced with a variety of games that implement their own lobby chat system where you may or may not chat during the loading of the game. worst, yet if you’re not in the same group starting the game you’re left waiting without ability to chat and only option is to send text messages through the ps3 menu. Since most of my friends lacked a smartphone or a computer nearby, it wasn’t worth the time. Times have changed now, everybody’s either got an android or iphone which is capable of connecting to a voip account, such as our “office” PBX we are about to set up right now.
I was able to get this running reasonably well in a virtual environment running Ubuntu. Probably will have less compression issues and sound clipping out if it was run from a physical box.
apt-get install asterisk
vim /etc/asterisk/sip.conf
add the following:
[common](!)
type=friend
context=default
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=g723
allow=g729
dtmfmode=rfc2833
[employee](common)
username=guest
secret=SOMEPASSWORD
you may also consider changing the port (default is 5060) while you’re at it. look for the line:
udpbindaddr=0.0.0.0
Next set up Conference Room
vim /etc/asterisk/meetme.conf
uncomment or add the following after “[rooms]”:
conf => 1234
Next set up an Extension to direct to the conference room:
vim /etc/asterisk/extensions.conf
uncomment or add the following after “[default]”:
exten => 8600,1,Meetme(1234)
We now need to start asterisk if it hasn’t started already
asterisk
In order to interact with asterisk we need to bring up the console
asterisk -r
Here we can issue a reload command
reload
Finally all you need to do now is to connect to the server using any sip/voip dialer. I recommend linphone for android and iphone and windows. for OSX you can use telephone which is found free on the app market. Currently there are no free dialers on Windows Phone =(.
your connection information should be:
server:YOURASTERISKIP
username:guest
password:SOMEPASSWORD
If you changed the port you’ll need to try altering the server name to YOURASTERISKIP:SOMEPORTNUM if the dialer doesn’t ask for one. Don’t forget if you are trying this remotely, there is no guarantee the coffee shop or your home internet provider will have port 5060 open.
Once connected, simply dial 8600 and you’re in the conference by yourself! you can use the same account information on another device to test a 2 person conference. I haven’t found any major issues with conference calls of 4.
From here, we can set up outgoing and incoming voip account/phone numbers including gmail voice! Security should be addressed next because the sip port can be easily bruteforced.
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