It is easy to install and use an Asterisk free soft PBX on Ubuntu box – if you do it frequently. I created a step by step guide to prevent you from the usually error messages during the installation.
I use Asterisk 18.104.22.168 on my 64 bit Ubuntu Server 10.10.
After the upgrade on a fresh install, we need three packages to build up our Asterisk. Let’s install the build-essential, libxml2-dev, and ncurses-dev packages.
apt-get install build-essential libxml2-dev ncurses-dev
You can find all versions of the Asterisk at the download section. Download the latest version, and extract it!
wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-22.214.171.124.tar.gz tar zxvf asterisk-126.96.36.199.tar.gz
Then install the Asterisk, config, and the shamples.
cd asterisk-188.8.131.52/ ./configure make make install make config make samples
When the installation finished, we have a preconfigured Asterisk. To test the Asterisk with two softphones, let’s configure the sip.conf and the extensions.conf.
Here is the sip.conf file:
[general] context = default bindport = 5060 bindaddr = 0.0.0.0 tcpbindaddr = 0.0.0.0 tcpenable = yes  type = friend callerid = User One <1001> secret = 1001 host = dynamic canreinvite = no dtmfmode = rfc2833 mailbox = 1001 disallow = all allow = ulaw transport = udp  type = friend callerid = User Two <1002> secret = 1002 host = dynamic canreinvite = no dtmfmode = rfc2833 mailbox = 1002 disallow = all allow = ulaw transport = udp
This config means that the Asterisk listen all IP, port 5060, and the TCP is enabled. It has two extensions defined for User One and User Two.
Here is the extensions.conf file:
[general] static=yes writeprotect=no [default] exten => 1001,1,Answer() exten => 1001,n,Dial(SIP/1001,20,tr) exten => 1001,n,Hangup exten => 1002,1,Answer() exten => 1002,n,Dial(SIP/1002,20,tr) exten => 1002,n,Hangup.
This config means that there are two accessible extensions existing.
After configuring the Asterisk, we need to start it.
I use X-Lite to connect to the soft PBX. Let’s configure the X-Lite!
All data come from the sip.conf. The Account name, and the Display name are the callerid. The User ID and the Authorization name are the extension number – inside the square brackets. The Password is the sicret. The Domain is the IP address of the Asterisk server.
If all configurations are good, the X-Lite will inform you.
Let’s call from User One to User Two.
Here is the User One side:
Here is the User Two side:
Troubleshooting? We always need when we try a new thing. The firs tool is the tcpdump of course, but the asterisk have a good command line interface (Asterisk CLI) to debug the problem. To access the Asterisk CLI type
This screen shows a successful call from 1001 to 1002.
The Asterisk works now! Are you wondering why I configured my Asterisk to answer on TCP? Because I would like to test the Direct SIP feature of the Microsoft Communications Server and the Lync. CU next time! 😉
New post available: Astarisk on Ubuntu Server 11.04 (Natty Narwhal).
You can find the OCS 2007 R2 co-existing scenario with Asterisk 1.8 PBX article here.