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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

This article or section is out of date.

Reason: This recommendation for asterisk-ciscoAUR may be obsolete, this was written quite a while ago, and modern versions may already include this patch. (Discuss in Talk:Asterisk)

Install the asteriskAUR package. If you are using Cisco-based phones it is recommended to use the asterisk-ciscoAUR package instead as this is pre-patched with the presence patch. (See Issue 13145).

Alternatively, you can install the asterisk-lts-18AUR or asterisk-lts-20AUR package to have a long-term support release (current latest LTS major version is Asterisk 20). Asterisk LTS releases tend to have fewer features, but will be maintained for much longer. See the Asterisk Versions page for complete details about the release cycle for all Asterisk versions. The asterisk-lts-20AUR package is currently merely a meta-package for asteriskAUR while the current latest major version is Asterisk 20. asteriskAUR will eventually switch to Asterisk 21, a non-LTS release, by the end of 2023.

Enable/start `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

As of Asterisk 20, the legacy `chan_sip` module is no longer compiled by default. Once Asterisk 21 is released (October 2023), `chan_sip` will be completely removed (it has already been removed from the master branch). If you would rather use `chan_sip` than `res_pjsip`, please install asterisk-lts-18AUR, which was the last LTS release to build `chan_sip`.

`res_pjsip` is the newer, more performant Asterisk channel module that supports the standard SIP (Session Initiation Protocol) protocol, the primary VoIP (Voice over Internet Protocol) protocol in use by many carriers, telcos, Internet telephony service providers (ITSPs), and enterprises. However, `res_pjsip` is considerably more complicated to set up than the older `chan_sip`.

The prevailing wisdom of the wider Asterisk community suggests to only use `chan_sip` if you already have an existing configuration for it, while also planning the upgrade to `res_pjsip`. For any new installations, `res_pjsip` is recommended. As mentioned above, `chan_sip` is no longer maintained, not even for security fixes (except in older LTS releases), so new Asterisk PBX administrators should go with a supported SIP channel driver (i.e., `res_pjsip`).

If you have an existing `chan_sip` configuration, to build it in Asterisk 20, you can do the following: Edit the asteriskAUR PKGBUILD, in the `build()` function. Modify the following line:

    PKGBUILD

    make MENUSELECT_CFLAGS= OPTIMIZE= DEBUG= ASTVARRUNDIR="/run/$pkgname" NOISY_BUILD=1

Prefix it with a call to build the menuselect tool, and enable `chan_sip`:

    PKGBUILD

    make menuselect.makeoptions && \
    menuselect/menuselect --enable chan_sip && \
    make MENUSELECT_CFLAGS= OPTIMIZE= DEBUG= ASTVARRUNDIR="/run/$pkgname" NOISY_BUILD=1

Note that this will need to be done with every update of the asteriskAUR package, understanding that this will no longer work once asteriskAUR moves on to Asterisk 21.

### PJSIP

Once installed, Asterisk has the `res_pjsip` configuration in `/etc/asterisk/pjsip.conf`. The sample file included with Asterisk gives several basic examples, but is not exhaustive for all pjsip options. Readers are encouraged to read the Asterisk Wiki Security best practices article, and Configuring res_pjsip before continuing.

This article contains a basic single SIP phone, multiple SIP trunk example, using SIP Station SIP trunks, from Sangoma. Two SIP trunks with SIP Station are configured for redundancy, as recommended by SIP Station. The following example was tested using a Sangoma/Digium D60 hardware phone, but any SIP 2.0 compliant hard- or softphone should suffice.

This example assumes the Asterisk PBX server and SIP phone are on a private IPv4 LAN, with a NAT router between the server/phone and the WAN/Internet. If you would like to use IPv6, please read the Configuring res_pjsip for IPv6 article.

#### modules.conf

First, ensure `res_pjsip.so` is not prefixed with `noload =`. In this example, the `require =` prefix makes Asterisk fail to load if `chan_pjsip.so` fails to load. This is not absolutely necessary, but unless you are using other VoIP protocols it may not make sense for Asterisk to load if `res_pjsip` does not load.

    /etc/asterisk/modules.conf

    ;...
    require = chan_pjsip.so
    ;...

#### pjsip.conf

Before diving into configuring `pjsip.conf`, review the PJSIP Configuration Wizard. It is useful for the simple use case of only one SIP provider/ITSP, and handles many of the pjsip objects for this automatically.

The file `pjsip.conf` is an ini-style configuration file, with `[section-headers-in-square-brackets]`, and either a hash (`#`) or a semicolon (`;`) as the comment character. Each line under a section header is a `key=value` pair, with section-specific keys set to the specified values. Note that section headers can have `(!)` appended after the closing square bracket, which indicates the section is a template, for later use within `pjsip.conf`. To instantiate a section using a previously defined template, suffix the section header with the `(template-name)`.

##### transport section

In order to define a SIP trunk for use with Asterisk, PJSIP has the concept of a network `transport`. From the Asterisk Wiki, the transport section configures how `res_pjsip` will operate at the transport layer. For example, it supports configuration options for protocols such as TCP, UDP, or WebSockets and encryption methods like TLS/SSL.

Transport sections can be named arbitrarily, but it is recommended to name the section that makes it easy to remember what it signifies. The following example names it `[transport-udp-nat]` for identifying that this transport is using UDP for SIP, and is expected to traverse a NAT device.

    /etc/asterisk/pjsip.conf

    [transport-udp-nat]
    type=transport  ; signifies this is a transport definition
    protocol=udp  ; specifies this uses the UDP transport protocol
    ; this transport binds to all configured network interfaces,
    ; replace with the IP address of the desired interface. 0.0.0.0 means all interfaces
    bind=0.0.0.0
    ; =====  NOTE, the following directives are OPTIONAL
    local_net=10.20.30.0/24  ; specify the networks that this Asterisk server should consider as local/LAN networks. Change this to the local LAN subnet (in CIDR notation)
    local_net=127.0.0.0/8  ; specify the networks that this Asterisk server should consider as local/LAN networks (this one sets it to the IPv4 loopback network)
    external_media_address=96.69.199.60  ; specify the external/WAN IP address for RDP media (audio)
    external_signaling_address=96.69.199.60  ; specify the external/WAN IP address for SIP (signaling)

Note that only one transport is allowed for each IP address/port or IP address/protocol mapping. If you would like to configure multiple transport protocols (e.g. both TCP and UDP), you will need to bind each protocol to a different IP address. Likewise if you want to define multiple transports using the same protocol, the ports used need to be different for each transport definition.

##### registration section

Next, define the registrations the SIP trunks will use. Note that the SIP Station registrations are first defined via a template (with `(!)` appended to the section name). Each template defines the settings that are common for each section using the template; the sections themselves only contain the options that need to be set for that section.

    /etc/asterisk/pjsip.conf

    [sipstation-reg](!)
    type=registration
    transport=transport-udp-nat
    contact_user=15055551234
    retry_interval=60

    ; ==== ... several other templates and sections will appear here
    ; ====     including the authentication sections, etc., which are also used by the registration
    ; ==== ....

    [siptsation-reg1](sipstation-reg)
    outbound_auth=sipstation-auth1
    server_uri=sip:trunk1.freepbx.com  ; trunk1 provided by SIP Station
    client_uri=sip:my_username@trunk1.freepbx.com

    [siptsation-reg2](sipstation-reg)
    outbound_auth=sipstation-auth2
    server_uri=sip:trunk2.freepbx.com  ;  trunk2 provided by SIP Station
    client_uri=sip:my_username@trunk2.freepbx.com

##### authentication section

Here, define the authentication template, and the specific authentication sections for use with the SIP Station trunks.

    /etc/asterisk/pjsip.conf

    [sipstation-auth](!)
    type=auth
    password=Super*Secret@Password!
    username=my_username

    ; ==== ... more templates may appear here

    [sipstation-auth1](sipstation-auth)
    realm=trunk1.freepbx.com

    [sipstation-auth2](sipstation-auth)
    realm=trunk2.freepbx.com

##### endpoint section

An endpoint is essentially a profile for the configuration of a SIP device such as a phone or remote server. This defines the transport, the Asterisk Dialplan context where calls originating from the endpoint gets handled, and the audio codecs allowed for the endpoint. For the SIP station trunks, define the following:

    /etc/asterisk/pjsip.conf

    [sipstation-endpoint](!)
    type=endpoint
    transport=transport-udp-nat
    context=from-external  ; Dialplan context, defined in extensions.conf or related file
    disallow=all   ; disallows all codecs, including default ones, unless explicitly allowed below
    allow=ulaw     ; standard G.711, uncompressed PCM codec. Uses the most bandwidth
    allow=gsm      ; another standard, compressed codec. Uses less bandwidth than ulaw
    ; other codecs are available, and have different attributes. Some require paid licenses to use.
    aors=sipstation-aors  ; AOR - Address of Record, to be defined later
    ;direct_media=no  ; setting this may not allow audio to pass through NAT
    direct_media=yes
    rtp_symmetric=yes

    ; ==== ... more template definitions

    [sipstation-endpoint1](sipstation-endpoint)
    outbound_auth=sipstation-auth1
    from_domain=trunk1.freepbx.com

    [sipstation-endpoint2](sipstation-endpoint)
    outbound_auth=sipstation-auth2
    from_domain=trunk2.freepbx.com

##### identify section

Controls how the res_pjsip_endpoint_identifier_ip module determines what endpoint an incoming packet is from. For the SIP Station trunks, the following is defined:

    /etc/asterisk/pjsip.conf

    [sipstation-identify](!)
    type=identify
    ; this is the LAN IP address of the NAT router, which calls from the SIP Station
    ; trunks may appear to come from (because of NAT).
    match=10.20.30.254/32

    [sipstation-id1](sipstation-identify)
    endpoint=sipstation-endpoint1
    match=192.159.66.3   ;  IP address of trunk1.freepbx.com

    [sipstation-id2](sipstation-identify)
    endpoint=sipstation-endpoint2
    match=162.253.134.142    ;  IP address of trunk2.freepbx.com

##### aor (Address Of Record) section

A primary feature of AOR objects (Address of Record) is to tell Asterisk where an endpoint can be contacted. Without an associated AOR section, an endpoint cannot be contacted. For the SIP Station trunks, define this:

    /etc/asterisk/pjsip.conf

    [sipstation-aors]
    type=aor
    contact=sip:trunk1.freepbx.com
    contact=sip:trunk2.freepbx.com

##### phone sections

Finally, define the necessary sections for the SIP phone that will register to Asterisk. It has several similar sections as with the SIP Station trunks. Expand these with templates if multiple local SIP phones are intended to register to your Asterisk PBX. Remote phones (from the WAN/Internet) can also be configured, but that iss outside the scope of this example.

    /etc/asterisk/pjsip.conf

    [home-phone]
    type=endpoint
    transport=transport-udp-nat
    context=from-internal  ;  Dialplan context that gets executed when a user dials from this phone
    disallow=all
    allow=ulaw
    allow=gsm
    auth=home-phone-auth
    aors=home-phone

    [home-phone-auth]
    type=auth
    auth_type=userpass
    password=MyPhonePa$$word0
    username=home-phone

    [home-phone]
    type=aor
    max_contacts=1

#### Managing PJSIP

There are several commands regarding `res_pjsip` available in the Asterisk CLI, all prefixed with the `pjsip` command. To get to the Asterisk CLI, enter the following command, as the `asterisk` user:

    $ asterisk -rvvv

This assumes Asterisk is already running (e.g., via the systemd service unit). Once in the Asterisk CLI, you will see the prompt `hostname*CLI>` . To see a list of available PJSIP commands, type `help pjsip`.

To see a list of PJSIP registrations, type the following:

    hostname*CLI> pjsip show registrations

     <Registration/ServerURI..............................>  <Auth....................>  <Status.......>
    ---------------------------------------------------------------------------------------------

     siptsation-reg1/sip:trunk1.freepbx.com                  sipstation-auth1            Registered        (exp. 2062s)
     siptsation-reg2/sip:trunk2.freepbx.com                  sipstation-auth2            Registered        (exp. 2069s)

    Objects found: 2

To see a list of PJSIP endpoints, type the following:

    hostname*CLI> pjsip show endpoints

     Endpoint:  <Endpoint/CID.....................................>  <State.....>  <Channels.>
        I/OAuth:  <AuthId/UserName...........................................................>
            Aor:  <Aor............................................>  <MaxContact>
          Contact:  <Aor/ContactUri..........................> <Hash....> <Status> <RTT(ms)..>
      Transport:  <TransportId........>  <Type>  <cos>  <tos>  <BindAddress..................>
       Identify:  <Identify/Endpoint.........................................................>
            Match:  <criteria.........................>
        Channel:  <ChannelId......................................>  <State.....>  <Time.....>
            Exten: <DialedExten...........>  CLCID: <ConnectedLineCID.......>
     ----------------------------------------------------------------------------------------------

     Endpoint:  home-phone                                           Not in use    0 of inf
         InAuth:  home-phone-auth/home-phone
            Aor:  home-phone                                         1
          Contact:  home-phone/sip:home-phone@10.20.30.229:506 f91928f561 NonQual         nan
      Transport:  transport-udp-nat         udp      0      0  0.0.0.0:5060

     Endpoint:  sipstation-endpoint1                                 Not in use    0 of inf
        OutAuth:  sipstation-auth1/mw3FSBsPPjQ7
            Aor:  sipstation-aors                                    0
          Contact:  sipstation-aors/sip:trunk1.freepbx.com     256c242e36 NonQual         nan
          Contact:  sipstation-aors/sip:trunk2.freepbx.com     c07c6c7180 NonQual         nan
      Transport:  transport-udp-nat         udp      0      0  0.0.0.0:5060
       Identify:  sipstation-id1/sipstation-endpoint1
            Match: 10.20.30.254/32
            Match: 192.159.66.3/32

     Endpoint:  sipstation-endpoint2                                 Not in use    0 of inf
        OutAuth:  sipstation-auth2/mw3FSBsPPjQ7
            Aor:  sipstation-aors                                    0
          Contact:  sipstation-aors/sip:trunk1.freepbx.com     256c242e36 NonQual         nan
          Contact:  sipstation-aors/sip:trunk2.freepbx.com     c07c6c7180 NonQual         nan
      Transport:  transport-udp-nat         udp      0      0  0.0.0.0:5060
       Identify:  sipstation-id2/sipstation-endpoint2
            Match: 10.20.30.254/32
            Match: 162.253.134.142/32

    Objects found: 3

### chan_sip

The factual accuracy of this article or section is disputed.

Reason: It is better to provide instructions for non obsoleted module (res_pjsip) instead of for obsoleted (chan_sip). (Discuss in Talk:Asterisk)

Note that the following instructions assume that you want to use already obsoleted `chan_sip` module. The `chan_sip` module is no longer maintained, but is easier to configure then the newer `res_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 `chan_sip`, you need to explicitly load the older module, and ensure that the newer `chan_pjsip.so` module is unloaded. In `/etc/asterisk/modules.conf` adjust the following:

    noload => chan_pjsip.so
    noload => res_pjsip.so
    ; noload => chan_sip.so

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 that is 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

Hidden categories:

- Pages or sections flagged with Template:Out of date
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