Difference between revisions of "Asterisk"

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(Installation)
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You may also want to install [https://aur.archlinux.org/packages.php?ID=10223 asterisk-addons], [https://aur.archlinux.org/packages.php?ID=10224 asterisk-sounds] and [https://aur.archlinux.org/packages.php?ID=24364 zaptel]. You will also need a SIP [[soft-phone]] and at least two machines. Recommendations for SIP phones are [http://www.wirlab.net/kphone/ kphone], in the AUR and [http://www.xten.net/index.php?menu=X-Series x-lite], a binary package.
 
You may also want to install [https://aur.archlinux.org/packages.php?ID=10223 asterisk-addons], [https://aur.archlinux.org/packages.php?ID=10224 asterisk-sounds] and [https://aur.archlinux.org/packages.php?ID=24364 zaptel]. You will also need a SIP [[soft-phone]] and at least two machines. Recommendations for SIP phones are [http://www.wirlab.net/kphone/ kphone], in the AUR and [http://www.xten.net/index.php?menu=X-Series x-lite], a binary package.
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To enable ilbc codec support add the following to the very beginning of the build section of the PKGBUILD
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<pre>  cd ${srcdir}/${pkgname}-${pkgver}/contrib/scripts
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  echo | ./get_ilbc_source.sh</pre>
  
 
==Configuration==
 
==Configuration==

Revision as of 17:12, 23 March 2013


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Reason: please use the first argument of the template to provide a brief explanation. (Discuss in Talk:Asterisk#)
Summary help replacing me
Asterisk is a complete PBX (private branch exchange) in software. It runs on Linux, BSD, Windows and OS X and provides all of the features you would expect from a PBX.

Asterisk is a complete PBX (private branch exchange) in software. It runs on Linux, BSD, Windows and OS X and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in four protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.

Asterisk provides voice-mail services with directory, call conferencing, interactive voice response and call queuing. It has support for three-way calling, caller ID services, ADSI, IAX, SIP, H.323 (as both client and gateway), MGCP (call manager only) and SCCP/Skinny.

This article will show you how to configure a simple in house network enabling us to use a SIP soft-phone to talk to another SIP soft-phone on your LAN.

Installation

Install asteriskAUR from the AUR. Start the server with
# rc.d start asterisk
.

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.

To enable ilbc codec support add the following to the very beginning of the build section of the PKGBUILD

  cd ${srcdir}/${pkgname}-${pkgver}/contrib/scripts
  echo | ./get_ilbc_source.sh

Configuration

SIP

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:

[me1]
type=friend
username=me1
secret=PASSWORD
host=dynamic
context=house

[me2]
type=friend
username=me2
secret=PASSWORD
host=dynamic
context=house

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:

[house]
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:

[default]
mode=files
directory=/var/lib/asterisk/mohmp3

Now go into your sip.conf

musiconhold=default

And that is all there is to it. Just copy your favorite legally obtained MP3 to /var/lib/asterisk/mohmp3.

Voicemail

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:

[general]
format=gsm|wav49|wav
serveremail=asterisk
attach=no
mailcmd=/usr/sbin/sendmail -t
maxmessage=180
maxgreet=60

[default]
100 => 1234,Me,me@mydomain.com

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 attach=yes 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 me@mydomain.com) 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

sip.conf
[general]
register => username:password@sip.specific.com

[whatever]                   
fromdomain=specific.com     
host=sip.specific.com
insecure=very    ; check with provider
username=usernameduh
secret=passwordduh
type=peer
extensions.conf
[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.
iax.conf

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 => FWDNUMBER:PASSWORD@iax2.fwdnet.net 
disallow = all
allow = ulaw

And at the bottom add:

[iaxfwd]
type=user
context=fromiaxfwd
auth=rsa
inkeys=freeworlddialup

This allows calls from FWD.

extensions.conf

Place this at the top under '[globals]':

FWDNUMBER=MYFWDNUMBER ; your calling number
FWDCIDNAME="MyName"; your caller id
FWDPASSWORD=MYFWDPASSWORD ; your password
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}:${FWDPASSWORD}@iax2.fwdnet.net/${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:

[fromiaxfwd]
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

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 192.168.0.142 port 5061 expires 60

Now you should be able to dial 101 from me1 and talk to me2.

Troubleshooting

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