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VoIP Phones - Asterisk

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Asterisk logo.png

Asterisk is extremely flexible and covering different uses for it is outside the scope of this example as the setup used here was very basic. You should read through the included documentation, especially the security documentation, before configuring Asterisk for the first time.


Asterisk has two methods to configure SIP connections. The legacy "sip.conf" (SIP) and the more modern "pjsip.conf" (PJSIP).

Newer installations of Asterisk should be configured to use PJSIP as it will be more supported as Asterisk development continues, however it's been reported that PJSIP doesn't support in-band DTMF detection properly. You may need to switch back to legacy sip.conf if this affects you. The official recommendation on the PJSIP FAQ seems to be to write your own plugin if you need it. In-band DTMF support seems like an important thing to have, so we suggest raising a bug to report a missing feature in PJSIP if this affects you!

It isn't a good idea to have an installation that mixes sip.conf with pjsip.conf.

When reading the instructions below be aware which are for sip.conf and which are for pjsip.conf.

pjsip.conf (PJSIP)

PJSIP: Trunk registration

Here is an example of a working pjsip.conf setup where Asterisk will register with A&A to receive calls.

In pjsip.conf:

   type = registration
   retry_interval = 20
   fatal_retry_interval = 20
   forbidden_retry_interval = 20
   max_retries = 9999
   auth_rejection_permanent = no
   contact_user = maininbound
   expiration = 120
   outbound_auth = auth_reg_442082881111
   client_uri =
   server_uri =
   type = auth
   password = BusinessPaidGrewCome
   username = +442082881111
   type = aor
   contact =
   type = identify
   endpoint = aaisptrunk
   match =
   type = endpoint
   context = maininbound
   dtmf_mode = rfc4733
   disallow = all
   allow = alaw
   allow = ulaw
   direct_media = no
   aors = aaisptrunk

Calls come into the context "maininbound" in extensions.conf - in this example calls get sent onto extension 222 and 205 for 20 seconds and then go to voicemail.

exten = maininbound,1,Dial(PJSIP/222&PJSIP/205,20)
exten = maininbound,n,Voicemail(222@default,us)

In extensions.conf you can dial out via the trunk with:

exten => _X.,1,Dial(PJSIP/${EXTEN}@aaisptrunk,,)

PJSIP: Keep-Alive / Anti-Idle

If you are using a firewall or NAT router with short timeouts on UDP sessions you can force packets to be sent over the connection to keep it alive.

Set qualify_frequency in the aor section; This triggers an OPTIONS message every X (as set) seconds. An example of the aor section follows:

type = aor
contact =

Status and Commands

A good command within the asterisk software is the show registration command:

asterisk*CLI> pjsip show registrations

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

 reg_442082881111/                auth_reg_442082881111  Registered

Objects found: 1

In this example it shows that the Asterisk server is successfully registered with the Andrews & Arnold SIP server.

sip.conf (SIP)

Incoming Calls

User Section

  • Accept authenticated calls and route them to a context.


  • We send Remote-Party-Id with the privacy and screen settings, setting trustrpid=yes in the incoming SIP config will allow Asterisk to pass withheld/unknown on.


  • Voiceless must authenticate so that calls are recognised as the above peer section.
  • You need to use the match_auth_username=yes setting otherwise Asterisk will not recognise Voiceless' initial requests.


  • We initially send an Authorization header with only a username, allowing Asterisk to identify Voiceless by username instead of by IP. By default Asterisk ignores the username when identifying peers.

Outgoing Calls

  • Either use a separate type=peer section or combine incoming and outgoing in one type=friend section

Separate Section



Combined Section


; incoming
; outgoing

Note: Order of sip.conf is important

In sip.conf, ensure that your incoming config is before the config for the outgoing.

Note: Asterisk and IPv6 SLAAC addresses

Asterisk will bind to all IPv6 addresses if it is set to use IPv6. This means that if you have a static IP and a SLAAC IP, Asterisk sometimes replies to invites sent to the static IP from the SLAAC IP instead which breaks things. We recommend using static IP addresses and disabling SLAAC (and privacy extensions) on the box running Asterisk until its IPv6 support is more mature.


If you're behind NAT it is helpful to make Asterisk register. It re-registers every 120 seconds by default anyway so should keep NAT sessions open. You can register (and tell Asterisk that it's behind NAT) with these settings under the [general] section:

register =>

In this example, extn is the extension that Asterisk will pass the call to. Localnet should of course be set to whatever RFC1918 range you are using on your LAN.


To make this work in a real dialplan you will want something like the following examples in extensions.conf. See Asterisk's included example sip.conf for examples of how to send the call to different contexts etc. For outbound calls:

exten => _X.,1,Dial(SIP/voiceless-out/${EXTEN})

For inbound calls (assuming you're routing call to a registered Snom):

exten => _X.,1,Dial(SIP/snom)

Firewall & Security

  • You will also want to set up firewall rules, as per the VoIP Firewall page.
  • Also see the VoIP Security page for information about securing your VoIP service.