# Asterisk

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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 The `sip` module is no longer maintained, but is easier to configure then the newer `pjsip` module.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,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,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.

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 OnSIP. 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).

### 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
  - fr
  - de
  - ...
- 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.

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.

Categories:

- Telephony
- Voice over IP