domingo, junho 27, 2010

Asterisk simple steps to instal


Install asterisk from the AUR.
You may also want to install asterisk-addons, asterisk-sounds and zaptel. You will also need a SIP soft-phone and at least two machines. Recommendations for SIP phones are kphone, in the AUR and x-lite, a binary package.



Assuming your asterisk server is up and running, we will only need to edit two files: sip.conf and extensions.conf. Change to your asterisk configuration directory (should be /etc/asterisk). Edit sip.conf and place the following:

This creates our two SIP users me1 and me2 with a password of PASSWORD in the house context.
We will be defining the context next -- edit extensions.conf with the following:
exten => 100,1,Dial(SIP/me1)

exten => 101,1,Dial(SIP/me2)
This creates the context house and assigns extension 100 to the SIP user me1, and extension 101 to the SIP user me2. Now all thats left is to see if it works.

Music On Hold

Music on hold is a really sweet feature. And once again easy to install and configure. Edit /etc/asterisk/musiconhold.conf and add, or make sure it is uncommented:
Now go into your sip.conf
And that is all there is to it. Just copy your favorite legally obtained MP3 to /var/lib/asterisk/mohmp3.


Voicemail is another feature of asterisk. There are many ways to configure it, however this article only covers a simple approach.
Create/edit your voicemail.conf:
mailcmd=/usr/sbin/sendmail -t

100 => 1234,Me,
What does this mean? Most of the [general] is pretty self-explanatory. However, do note that if you have postfix set up right the PBX will send an email notifying the user of a new voice-mail and if {{{1}}} is defined it will attach the file.
Now for the actual mailbox. The format is:
mailbox => password,user,email
In this case, we gave 'Me' (email mailbox 100, with a password of 1234.
Now we have to have a way to leave messages to this voice-mail, and a way to access it. For this, we go back to the extensions.conf and modify your existing entry as follows:
exten => 100,1,Dial(SIP/me1,20)
exten =>100,2,Voicemail(100@default)
The 20 on the end of the first 'exten' tells 'Dial()' to call for 20 seconds. If no one answers it heads to voice-mail box 100 in the default context.
Next is actually accessing your voicemail. For this we add:
exten => 600,1,VoiceMailMain,s100@default
So when we call 600, the application 'VoiceMailMain' goes to 100 in the default context. The s allows for automatic login.
Note: The 'VoiceMail' applications have a significant amount of options, so it is suggested reading over some additional documentation. This is just for a basic, home use setup. Also note that it is generally a good idea to use extensions higher then your users extensions for accessing 'VoiceMail'. This way someone dialing 208 does not hit someone's voice-mail at 205.

Connecting To The PSTN

Now that you have the previous setup, it is time to actually connect to the outside world. To do this, you will need a provider such as Junction Networks. Your provider should have instructions on connecting to asterisk, so this section is very general.

General Set-up

register =>

insecure=very    ; check with provider
[outboundwithCID]  ; this can be whatever
exten => _1NXXNXXXXXX,1,SetCIDNum(15555551234)
exten => _1NXXNXXXXXX,2,Dial(SIP/${EXTEN}@whatever)
exten => _1NXXNXXXXXX,3,Congestion()
exten => _1NXXNXXXXXX,103,Busy()

[default]  ; This should be set in your sip.conf for incoming calls

;These should to be changed to your actual number
; ie     15555555555
exten => 1NXXNXXXXXX,1,Answer()
exten => 1NXXNXXXXXX,2,Playback(ttt-weasels)
exten => 1NXXNXXXXXX,3,HangUp()
  • In the outbound context, any number dialed will be sent out to your service provider. The 'whatever' in the 2 priority should match what you have in your sip.conf.
  • Of course, the inbound dial-plan can be modified to do what you want. For instance, you can have Dial(SIP/me1) so when someone calls your number they are routed to your SIP phone on your computer. Then add in voice-mail and so on.


FWD is one of the most popular VOIP services. It is similar to AIM, where everyone gets a screen name (FWD number) and can communicate on the network. Pairing FWD with ipkall allows free incoming calls.
The first step is to log into FWD and enable their side of IAX. It is under extra features, and keep in mind that the authors claim it takes a little while to activate.
Now edit your iax.conf with the following in the 'general' section:
register => 
disallow = all
allow = ulaw
And at the bottom add:
This allows calls from FWD.
Place this at the top under '[globals]':
FWDNUMBER=MYFWDNUMBER ; your calling number
FWDCIDNAME="MyName"; your caller id
FWDRINGS=sip/office ; the phone to ring
FWDVMBOX=1000 ; the VM box for this user
Next, add this to a context for outgoing:
exten => _393.,1,SetCallerId,${FWDCIDNAME}
exten => _393.,2,Dial(IAX2/${FWDNUMBER}:${FWDPASSWORD}${EXTEN:3},60,r)
exten => _393.,3,Congestion
You can change the '393' to whatever you want. This is what you will dial before dialing a 'fwd' number. For instance, to dial '744561' you would dial '393744561'.
And lastly, the incoming calls:
exten => ${FWDNUMBER},1,Dial(${FWDRINGS},20,r)
exten => ${FWDNUMBER},2,Voicemail,u${FWDVMBOX}
exten => ${FWDNUMBER},102,Voicemail,b${FWDVMBOX}
Note: If you have problems try removing the variables from extensions.conf. These instructions are from FWD's site and I have not been tested by this article's author.
Extensions to try calling are 55555 (a volunteer maned test line) and 514 (conference).


MeetMe is the application that allows you to do conference calling. Same as everything, basic setup is easy.
Edit meetme.conf:
conf => 1000
Next is extensions.conf
exten => 999,1,MeetMe(1000|M)
Now dial 999 to get into conference 1000. The enables music on hold if no one is in there. It will automatically go away when someone joins the conference.
Note: You must have the zaptel package in order for MeetMe to work. Install it and run modprobe ztdummy before running asterisk. This provides digium timing for us without cards so we can utilize TDM.

Asterisk Console And Softphones

Now lets get Asterisk going:
# asterisk -vvvvvvc
This will give us the Asterisk CLI with verbose output. If Asterisk is already running you will need to use:
# asterisk -r
Now fire up your SIP clients and set them up with the information in the sip.conf. Switch back to your Asterisk CLI and you should see:
Registered SIP 'me1' at port 5061 expires 60
Now you should be able to dial 101 from me1 and talk to me2.


If you receive a 404 Not Found error check your extensions.conf and the number you dialed.

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