Asterisk
Asterisk is a complete PBX (private branch exchange) in software. It runs on Linux, BSD, Windows and macOS 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 softphone to talk to another SIP softphone on your LAN.
Installation
Install the asteriskAUR package. If you are using Cisco-based phones it is recommended to use the asterisk-ciscoAUR package instead as this is prepatched with the presence patch. (See https://issues.asterisk.org/jira/browse/ASTERISK-13145)
Start the server with asterisk.service.
You will also need a SIP softphone and at least two machines. Recommendations for SIP phones are Blink (blinkAUR), Linphone (liblinphone-gitAUR) or X-Lite (xlite-binAUR).
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
Note that the following instructions assume that you want to use already obsoleted sip module. The sip module is no longer maintained, but is easier to configure then the newer pjsip module. There is a summary of configuration changes between two modules, as well as instructions on how to migrate to the newer module, on the Asterisk wiki.
In order to use sip, you need to explicitly load the older module, and ensure that the newer pjsip module is unloaded. In /etc/asterisk/modules.conf adjust the following:
noload => chan_pjsip.so noload => res_pjsip.so ; noload => chan_sip.so # add comment at this line
SIP
Add this to the following file:
/etc/asterisk/sip.conf
... [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. The context will be defined next.
Add this to the following file:
/etc/asterisk/extensions.conf
[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=mohmp3
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,[email protected]
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 [email protected]) 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,n,Voicemail(u100@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.
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 OnSIP. Your provider should have instructions on connecting to asterisk, so this section is very general.
General set-up
sip.conf
[general] register => username:[email protected] [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:[email protected] 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}
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).
Sounds
Sounds are stored in the folder /var/lib/asterisk/xx, xx stands for the code of the language for example "en" for English. To add new sounds copy them to the folder. Preserve the following folder structure:
/var/lib/asterisk/sounds/xx /var/lib/asterisk/sounds/xx/digits /var/lib/asterisk/sounds/xx/letters /var/lib/asterisk/sounds/xx/phonetic
Edit the language parameter in the sip.conf
[general] ... language=en ...
Possible sources for sounds are:
- https://downloads.asterisk.org/pub/telephony/sounds/
- https://packages.debian.org/wheezy/all/asterisk-prompt-xx
- voip-info.org
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.
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.