PJSIP configuration on Asterisk

Overview

This guide walks you through configuring a Simtex SIP trunk using the PJSIP channel driver on a vanilla Asterisk system. PJSIP is the modern replacement for chan_sip and offers improved performance, security, and standards compliance.

The main configuration file you'll be working with is pjsip.conf. You'll also need to update extensions.conf to use the PJSIP channel driver for dialling.

What you'll need

• A working Asterisk installation with PJSIP support
• Your Simtex account number (e.g. 214XXXXXXX)
• Your Simtex SIP password
• Your server's public IP address

Step 1 — Transport

First, configure a TCP transport in pjsip.conf. We recommend TCP over UDP for reliability and NAT traversal.

[Transport-TCP]
type = transport
protocol = tcp
bind = 0.0.0.0

Step 2 — Registration, Authentication & AOR

Next, configure three related objects: a registration to register with Simtex, an auth section for credentials, and an AOR (Address of Record) to define the contact endpoint.

Registration

[Simtex]
type = registration
server_uri = sip:214XXXXXXX@siptcp.simtex.com.au:5062;transport=tcp
client_uri = sip:214XXXXXXX@XXX.XXX.XXX.XXX
contact_user = 214XXXXXXX
transport = Transport-TCP
outbound_auth = Simtex_Auth
auth_rejection_permanent = no
retry_interval = 30
forbidden_retry_interval = 290
max_retries = 20

Replace 214XXXXXXX with your Simtex account number and XXX.XXX.XXX.XXX with your server's public IP address.

Authentication

[Simtex_Auth]
type = auth
username = 214XXXXXXX
password = XXXXXXXX

AOR (Address of Record)

[Simtex_Aor]
type = aor
contact = sip:siptcp.simtex.com.au:5062;transport=tcp
qualify_timeout = 4.0
qualify_frequency = 50

Step 3 — Endpoint & Identify

Define the endpoint and an identify section so Asterisk knows which endpoint to associate incoming traffic with.

Endpoint

[Simtex_Endpoint]
type = endpoint
context = From_Simtex
disallow = all
allow = alaw
allow = ulaw
outbound_auth = Simtex_Auth
aors = Simtex_Aor
from_user = 214XXXXXXX
send_pai = yes
send_rpid = yes
trust_id_inbound = yes
transport = Transport-TCP

Identify

[Simtex_Identify]
type = identify
endpoint = Simtex_Endpoint
match = siptcp.simtex.com.au

The identify section tells Asterisk to route incoming calls from Simtex to the Simtex_Endpoint.

Step 4 — Extension Configuration

With the trunk configured, you'll need at least one local extension to make and receive calls. Here's a basic PJSIP extension example:

[6001]
type = endpoint
context = default
disallow = all
allow = alaw
allow = ulaw
transport = Transport-TCP
auth = 6001_Auth
aors = 6001_Aors

[6001_Auth]
type = auth
auth_type = userpass
password = Ch@ngeM3!
username = 6001

[6001_Aors]
type = aor
max_contacts = 1
qualify_timeout = 4.0
qualify_frequency = 50

Next Steps

With your trunk registered and an extension configured, you can now set up your dialplan in extensions.conf to route calls through the Simtex trunk.

Note that the PJSIP dial syntax differs from the legacy chan_sip driver. For detailed examples of dialling patterns, caller ID management, and other PJSIP-specific tips, see our companion article:

PJSIP Tips and Tricks