Using Asterisk In Your Home - The Setup
When setting up Asterisk for home use, you’ve got a million and a half ways to do it. If you want to take the easy way, Asterisk@Home is the way to go, thus the name. It provides a GUI that you can use, instead of having to write all your own code. However, if it were me, I like writing my own code. So that’s what I would do.
Now, for this kind of thing you have a few options.
- Use VOIP
- Use your existing Landline
- Turn your Landline into VOIP
The first two are what we will talk about. The third is too much work. I don’t want to ever have to do it. If you happen to know an easy way to do it, PLEASE leave me a comment on how to do it or email me and I’ll make it a post here!
Right, so getting and using VOIP. Easy peasy. Subscribe for a service that will terminate the calls for you such as Nufone. I like Nufone, however lots of people have had trouble with them. It’s really cheap, and I’ve had good customer support from them.
When you first sign up, they will give you the details neccessary to set their service up with Asterisk. After that, you’ll want to just follow the basic guidelines that I have set up below. VOIP is just that easy. ![]()
Next, so using using your landline. To keep it all simple and userfriendly, what we will do is use a phone set that you can expand without having to use extra phone cord. Something cheap like this will do just fine. You’ll need an ATA (Analog Telephone Adapter), an FXS or FXO adapter. They basically all do the same thing, you can check out all sorts of different brands here. I’m not going to get into the installation of this, seeing as how it will come with instuctions.
We will concentrate on using a Sipura, you can modify this code to fit most of the others here, and if you can’t figure it out, contact me and I will try to help, or check out the www.Voip-Info.org wiki. The sipura functions as a SIP phone, so in SIP.conf, you’ll need to have something similar to this.
[general]
context=line1[sipura1]
type=friend
context=fone1
secret=passwordhere
host=192.168.1.106
dtmfmode=rfc2833[sipura2]
type=friend
context=line1
secret=password2
host=192.168.1.106
dtmfmode=rfc2833
This will tell Asterisk what and where the sipura is located. Next we must add it to the dialplan.
[phone1]
;exten => _X.,1,Answer
;exten => _X.,2,Wait(2)
;exten => _X.,3,Playback(tt-weasels)
;exten => _X.,4,Hangup
exten => _X.,1,Dial(IAX2/user@switch -2.nufone.net/${EXTEN})
exten => _X.,2,Hangup
[line1]
exten => homeline,1,Dial(SIP/sipura1@sipura1,10)
exten => homeline,2,Voicemail(1)
exten => homeline,3,Hangup
Ok, now we have added it to the Dialplan. By now you should have a good grasp of what’s going on in the dialplan and SIP.conf so I’m not going to explain it right now.
After adding this to the dialplan, you will pick up your phone, and it will send you to the voicemail. You can set it to dialout or whatever you want it to do. This is just a basic basic basic dialplan.
So I’ve given you a couple of options and I hope you use them wisely. Next, we’ll talk about the dialplan and the different things you can do with it for home use.
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February 27th, 2006 at 11:31 am
It’s turning out good. Keep the color layout. It’s easy on the eyes.
January 19th, 2007 at 1:28 am
Greetings….if this is truly intended for a beginner, you might want to include the point that the “dialplan” is actaully the extensions.conf file.
January 23rd, 2008 at 4:09 pm
Am unable to execute the dialplan
Am trying to connect two clients through iax softphone. installed on both clients,
am doin this on vmware work station,
Windows Professional -main host
windows prof -guest on virtual machine
i registered two iax users and gave them extensions as well in dial plan,on asterisk server
what next
How to make call from client 1 to client two