Break out your SSH terminals ladies. If you don't have a base setup don't read further and go back to look at my other help articles. When you are done with this article you should be able to set up an IAX2 soft client and configure Asterisk to route this call to our sip phones a the office. We will use GSM as the single codec to keep it simple. Our options in the iax.conf file will be tailored to low bandwidth connections.
1) Open a terminal window. If you need to access the server remotely Download an SSH (Secure Shell) client to access the Asterisk server. You can use Secure Shell from a vareity of Microsoft Windows clients freely available on the world wide web. If you have Linux or Mac OS X just read the man files from a terminal. From the shell change directory to the Asterisk config folder.
Example:
[matt@localhost ~]$ cd /etc/asterisk/
2) Change to root user and open up the iax.conf file with your favorite text editor. When you are done the iax.conf should look like below. What we are doing here is defining the base options for our remote soft phone to be able to register. Once we finish this we will move to the extensions.conf to add in an IAX context.
Example:
[matt@localhost asterisk]$ su
Password:
[root@localhost asterisk]#nano iax.conf
[general]
port=4569
bandwidth=low
disallow=all
allow=gsm
jitterbuffer=yes
tos=lowdelay
[9252]
type=friend
context=from-iax
secret=2005
host=dynamic
allow=gsm
auth=plaintext,md5,rsa
callerid=9252
mailbox=9252
3) Next we open up the extensions.conf, and add a route to the IAX user we just defined. Your extensions.conf will look like below. These are the same extensions as used in previous help articles.
Example:
[root@localhost asterisk]# nano extensions.conf
[general]
static=yes ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
[bogon-calls]
exten => _.,1,Congestion
[from-sip]
exten => 9250,1,Dial(SIP/9250,20)
exten => 9250,2,Voicemail(u9250)
exten => 9250,102,Voicemail(b9250)
exten => 9250,103,Hangup
exten => 9251,1,Dial(SIP/9251,20)
exten => 9251,2,Voicemail(u9251)
exten => 9251,102,Voicemail(b9251)
exten => 9251,103,Hangup
exten => 2999,1,Voicemailmain()
exten => 2999,2,Hangup
include => from-iax
[from-iax]
exten => 9252,1,Dial(IAX2/9252,20)
exten => 9252,2,Voicemail(u9252)
exten => 9252,102,Voicemail(b9252)
exten => 9252,103,Hangup
include => from-sip
4) Finally, we need to add in our user to voicemail.conf.
Example:
[general]
format=wav
attach=no
[default]
9250=1000,matt, matt@somecompany.com
9251=1000,joel,joel@somecompany.com
9252=1000,gerald,gerald@somecompany.com
5) We're almost done we need to download a soft client. Get your speakers and microphone ready. Go to
http://www.laser.com/dante/diax/diax.html. Download the latest version of Diax. After it is installed, right click on one of the small boxes that goes from right to left in numerical order. Each one of these boxes contains the possible servers the client can register too. In our case we have one server so we'll click on the first box and enter our login information. Remember that we orginally set up GSM as the codec. So select GSM and unselect the other codecs. Leave the context blank. Oh by the way, the name and number fields on the bottom are for Caller ID information.
Thats it folks. Fire up Asterisk and enjoy.
Note: My name is Matt Birkland, I work as a VoIP Engineer for VoiceIP Solutions an Asterisk Provider in Washington State. Every week I will be submitting a one page Asterisk/VoIP tip of the week on the blog.
