dn42

Getting Started with telephony42 (Asterisk)

This guide will help you quickly deploy a working Asterisk PBX. By the end of this tutorial, you will be able to make and receive calls in telephony42.

Prerequisites

Before configuring Asterisk, ensure you have:

  1. An allocated telephony42 (+042) number prefix registered in the registry (e.g., +04240000).
  2. DNS NAPTR records correctly configured on your zone in tel.dn42.

Example of wildcard NAPTR record to route any calls to your PBX at pbx.yourdomain.dn42 with SIP port 5060:

*.0.0.0.0.4.2.4.0.tel.dn42. IN NAPTR 100 10 "u" "E2U+sip" "!^(.*)$!sip:\\1@pbx.yourdomain.dn42:5060!" .

Installation notes

Debian / Ubuntu

Asterisk is available in the official repositories. (For Debian, you need to install packages from sid repository)

apt update
apt install asterisk

Alpine

apk add asterisk

Docker (docker compose)

services:
  asterisk:
    image: andrius/asterisk:stable
    container_name: asterisk42
    network_mode: host
    volumes:
      - ./config:/etc/asterisk
    restart: unless-stopped

Example configuration

Asterisk relies on many configuration files. This guide will focus only on two main config files:

  • pjsip.conf: Network, Endpoints, and Authentication
  • extensions.conf: Dialplan and Routing

When copying the configurations below, replace the following variables (remove the brackets):

  • <OWN_IPv4>: Your dn42 IPv4 address (e.g., 172.20.X.X)
  • <OWN_IPv6>: Your dn42 IPv6 address (e.g., fdXX::X)
  • <OWN_PREFIX>: Your telephony prefix (e.g., +04240000)
  • <EXT_NAME>: Name for your extension (e.g., 0001 or my-phone)
  • <FULL_NUMBER>: The full +042 number for your extension (e.g., +042400000001)
  • <SECRET_PASS>: A strong password

PJSIP (pjsip.conf)

This file defines how Asterisk listens to the network, sets up templates for devices, creates your local extension, and prepares an anonymous endpoint to catch incoming ENUM calls.

[global]
type=global
# verified peers first
endpoint_identifier_order=auth_username,username,ip

; ==========================================
; Transports
; ==========================================

[transport-udp4]
type=transport
protocol=udp
bind=<OWN_IPv4>:5060
local_net=172.20.0.0/14
local_net=172.31.0.0/16
local_net=10.0.0.0/8

[transport-udp6]
type=transport
protocol=udp
bind=[<OWN_IPv6>]:5060
local_net=fd00::/8

; ==========================================
; Templates
; ==========================================

[phone-template](!)
type=endpoint
context=context-local          ; Incoming calls from this phone go here
allow=!all,opus,g722,alaw,ulaw ; Prioritize Opus and G.722 HD audio
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
direct_media=no                ; Proxy media through PBX to prevent NAT issues
trust_id_inbound=yes
send_pai=yes

; ==========================================
; Local Extensions
; ==========================================

; Replace variables to create an extension names <EXT_NAME>
[<EXT_NAME>](phone-template)
auth=<EXT_NAME>
aors=<EXT_NAME>
callerid="My Name" <<FULL_NUMBER>>

[<EXT_NAME>]
type=auth
auth_type=userpass
username=<EXT_NAME>
password=<SECRET_PASS>

[<EXT_NAME>]
type=aor
max_contacts=3   ; Allow 3 devices to register simultaneously
remove_existing=yes

; ==========================================
; Incoming calls via ENUM
; ==========================================

[peer-enum-inbound]
type=endpoint
context=context-enum   ; Send unknown incoming calls to ENUM context
allow=!all,opus,g722,alaw,ulaw
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
direct_media=no

[peer-enum-inbound-id]
type=identify
endpoint=peer-enum-inbound
; match all dn42 subnets
match=172.20.0.0/14
match=172.31.0.0/16
match=10.0.0.0/8
match=fd00::/8

; ==========================================
; Outgoing calls via ENUM
; ==========================================

[peer-enum-outbound]
type=endpoint
allow=!all,opus,g722,alaw,ulaw
send_pai=yes

Dialplan (extensions.conf)

The file tells Asterisk how to route numbers. It is divided into serveral contexts.

[globals]
; (Optional) Global variables can go here

; ==========================================
; Inbound Routing
; ==========================================

; Calls originating from your own network.
[context-local]

; 4 digits -> call local extension
exten => _XXXX,1,Goto(ext-local,${EXTEN},1)

; starts with 042 -> add + and route out
exten => _042X.,1,Goto(ext-routing,+${EXTEN},1)

; standard +042 dialing -> route out
exten => _+042X.,1,Goto(ext-routing,${EXTEN},1)

; Catch-all -> add +042 and route out
exten => _X!,1,Goto(ext-routing,+042${EXTEN},1)


Calls originating from external networks via ENUM.
[context-enum]

; starts with 00 -> replace with + and route out
exten => _00X!,1,Goto(ext-routing,+${EXTEN:2},1)

; starts with + -> route out
exten => _+X!,1,Goto(ext-routing,${EXTEN},1)

; Catch-all -> add + and route out
exten => _X!,1,Goto(ext-routing,+${EXTEN},1)


; ==========================================
; Outbound & Local Routing
; ==========================================

; Calls destined for local networks.
[ext-local]
; route <OWN_PREFIX>0001 calls to your local extension <EXT_NAME>
exten => 0001,1,Dial(PJSIP/<EXT_NAME>,30)  ; 


; Inter-PBX routing for both inbound and outbound calls.
[ext-routing]

; route calls destined for your prefix
exten => _<OWN_PREFIX>.,1,Goto(ext-local,${EXTEN:9},1)

; other external calls -> do ENUM lookup on tel.dn42
exten => _+042X.,1,Set(TARGET_URI=${ENUMLOOKUP(${EXTEN},sip,,,tel.dn42)})
 ; if found, dial via our outbound endpoint
 same => n,ExecIf($["${TARGET_URI}"!=""]?Dial(PJSIP/peer-enum-outbound/sip:${TARGET_URI},60))
 ; if unallocated or failed, naturally hangup
 same => n,Hangup()

Apply your configuration by entering the Asterisk CLI:

asterisk -rx 'core reload'

Setting up your extension

You can now connect a softphone (like MicroSIP, Zoiper, or Linphone) or a hardware IP Phone/ATA using the following details:

  • Domain / Server: pbx.yourdomain.dn42 / <OWN_IPv4> or <OWN_IPv6>
  • Port: 5060 (UDP)
  • Username: <EXT_NAME>
  • Password: <SECRET_PASS>

Once connected, ask a friend in telephony42 to call your <FULL_NUMBER>.

Tip: Install and use sngrep -c in your PBX's terminal to visually monitor SIP traffic if calls fail.

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