Asterisk

How to Set Up a PJSIP Trunk in Asterisk with IPComms

A complete, production-ready guide to configuring PJSIP SIP trunking in Asterisk 18, 20, and 21+ using IPComms as your VoIP provider. Covers IP authentication, registration, TLS/SRTP encryption, dialplan, and troubleshooting.

IP
IPComms Team
||12 min read

Introduction: Why PJSIP and Why IPComms?

If you are running Asterisk in 2026, PJSIP is no longer optional -- it is the only supported SIP channel driver. The legacy chan_sip module was deprecated in Asterisk 19, marked for removal in Asterisk 20, and is completely gone in Asterisk 21 and later releases. Whether you are building a new PBX from scratch or migrating an existing deployment, understanding how to configure a PJSIP trunk is an essential skill for any Asterisk administrator.

This guide walks you through a complete Asterisk PJSIP trunk setup using IPComms as your SIP trunking provider. We will cover both IP-based authentication (ideal for production servers with static IPs) and registration-based authentication (perfect for dynamic IPs or home labs), plus TLS/SRTP encryption for secure communications.

IPComms has been providing enterprise-grade voice solutions since 2002. With rates starting at just $0.01 per minute for US and Canada, no contracts and real human support, we are the preferred Asterisk VoIP provider for businesses of all sizes. Our infrastructure is purpose-built for SIP trunking, supporting both UDP and TLS signaling with SRTP media encryption at no additional cost.

By the end of this article, you will have a fully functional, production-ready PJSIP trunk connected to IPComms, with inbound and outbound calling working perfectly on your Asterisk system.

Prerequisites

Before you begin your Asterisk SIP trunk configuration, make sure you have the following in place:

  • Asterisk 18, 20, or 21+ installed and running (with the res_pjsip module loaded)
  • Root or sudo access to your Asterisk server
  • An IPComms account -- sign up here if you do not have one yet
  • A static public IP address (for IP authentication) or DNS hostname
  • Firewall configured: UDP port 5060 (SIP signaling), UDP 10000-20000 (RTP media), or TCP port 5061 for TLS

Using FreePBX?

If you are running FreePBX, the GUI handles most of this configuration for you. Check out our FreePBX SIP Trunking guide instead.

Step 1: Sign Up for IPComms SIP Trunking

Getting started with IPComms takes less than five minutes. Here is what you need to do:

  1. Create your account at portal.ipcomms.net. You will receive instant access to your dashboard.
  2. Choose your SIP trunking plan. Plans start at just $5/month for the base service, which includes a free DID (phone number) and unlimited concurrent channels.
  3. Configure authentication. In your portal, navigate to Trunks and either whitelist your server's public IP address (recommended for production) or generate registration credentials.
  4. Note your SIP server details. Your assigned SIP server is s1.ipcomms.net (IP: 34.23.59.14).

Once your account is provisioned, you are ready to configure Asterisk. There is no waiting period -- your trunk is active immediately, so you can start making and receiving calls as soon as your PJSIP trunk provider configuration is complete.

Step 2: Configure PJSIP Transport

The transport section defines how your Asterisk server communicates at the network level. This is the foundation of your PJSIP transport configuration. Open your pjsip.conf file (typically at /etc/asterisk/pjsip.conf) and add the following transport section.

UDP Transport (Standard)

pjsip.conf - Transport Section
; ============================================ ; PJSIP Transport - UDP ; ============================================ [transport-udp] type = transport protocol = udp bind = 0.0.0.0:5060 ; NAT settings - REQUIRED if behind NAT/firewall ; Replace with YOUR server's public IP address external_media_address = YOUR_PUBLIC_IP external_signaling_address = YOUR_PUBLIC_IP local_net = 10.0.0.0/8 local_net = 172.16.0.0/12 local_net = 192.168.0.0/16

NAT Configuration is Critical

If your Asterisk server is behind a NAT firewall (common in AWS, GCP, Azure), you must set external_media_address and external_signaling_address to your public IP. Failure to do so results in one-way audio or no audio.

The local_net directives tell Asterisk which subnets are local (private). When communicating with addresses outside these ranges, Asterisk uses the external addresses, ensuring RTP media packets carry the correct public IP in SDP.

If your server has a public IP directly assigned (no NAT), you can omit the external_media_address, external_signaling_address, and local_net lines entirely.

Step 3: Configure the SIP Trunk (IP Authentication)

IP authentication is the recommended method for production Asterisk deployments. With IP auth, IPComms recognizes your server by its public IP address -- no usernames or passwords are transmitted over the network, making it both simpler and more secure.

This Asterisk SIP trunk configuration uses three PJSIP objects: an endpoint (defines call settings), an AOR (Address of Record, where to send calls), and an identify section (matches incoming traffic from IPComms by IP).

pjsip.conf - IPComms Trunk (IP Authentication)
; ============================================ ; IPComms SIP Trunk - IP Authentication ; ============================================ ; Endpoint - defines call behavior and codecs [ipcomms] type = endpoint context = from-ipcomms disallow = all allow = ulaw allow = alaw direct_media = no trust_id_inbound = yes send_rpid = yes send_pai = yes rtp_symmetric = yes force_rport = yes rewrite_contact = yes ice_support = no aors = ipcomms ; AOR - Address of Record (where to reach IPComms) [ipcomms] type = aor contact = sip:s1.ipcomms.net:5060 qualify_frequency = 60 qualify_timeout = 5.0 ; Identify - match inbound calls from IPComms by IP [ipcomms] type = identify endpoint = ipcomms match = 34.23.59.14 match = s1.ipcomms.net

Configuration Explained

  • context = from-ipcomms: Inbound calls from IPComms enter this dialplan context (defined in Step 5).
  • allow = ulaw / alaw: G.711 codecs provide the best voice quality. IPComms supports both ulaw (North America) and alaw (international).
  • direct_media = no: Forces all RTP media through Asterisk. Essential when behind NAT.
  • trust_id_inbound = yes: Accepts caller ID information from IPComms, including CNAM.
  • rtp_symmetric = yes: Sends RTP back to the address/port it was received from. Critical for NAT traversal.
  • qualify_frequency = 60: Sends OPTIONS every 60 seconds to verify trunk reachability and keep NAT pinholes open.
  • match = 34.23.59.14: Identifies incoming SIP traffic from IPComms and routes it to this endpoint.

Pro Tip: IP Auth is Best for Production

IP authentication eliminates registration overhead, avoids timeouts, and is immune to credential brute-force attacks. If you have a static IP, always choose IP auth for your pjsip trunk provider connection.

Step 4: Configure the SIP Trunk (Registration Authentication)

If your Asterisk server has a dynamic IP address (common for home labs, development environments, or residential ISPs), you will need registration-based authentication. Your Asterisk server registers with IPComms periodically, and IPComms uses the registration to know where to send inbound calls.

pjsip.conf - IPComms Trunk (Registration Auth)
; ============================================ ; IPComms SIP Trunk - Registration Auth ; ============================================ ; Authentication credentials [ipcomms-auth] type = auth auth_type = userpass username = your_username password = your_password ; Registration - keeps IPComms updated with your location [ipcomms-reg] type = registration outbound_auth = ipcomms-auth server_uri = sip:s1.ipcomms.net client_uri = sip:your_username@s1.ipcomms.net retry_interval = 60 expiration = 3600 ; Endpoint - defines call behavior [ipcomms] type = endpoint context = from-ipcomms disallow = all allow = ulaw allow = alaw direct_media = no trust_id_inbound = yes send_rpid = yes send_pai = yes rtp_symmetric = yes force_rport = yes rewrite_contact = yes outbound_auth = ipcomms-auth aors = ipcomms ; AOR - Address of Record [ipcomms] type = aor contact = sip:s1.ipcomms.net:5060 qualify_frequency = 60 ; Identify - match inbound calls from IPComms [ipcomms] type = identify endpoint = ipcomms match = 34.23.59.14 match = s1.ipcomms.net

The key differences are the auth and registration sections. The expiration = 3600 tells IPComms to expect re-registration every hour. The retry_interval = 60 means Asterisk retries every 60 seconds if registration fails.

Your credentials are available in the IPComms portal under Trunk Settings. Replace your_username and your_password with your actual credentials.

Step 5: Create the Dialplan

The dialplan in /etc/asterisk/extensions.conf controls call routing. You need two contexts: outbound (calls to PSTN via IPComms) and inbound (calls from IPComms to your extensions).

Outbound Dialplan

extensions.conf - Outbound Calling
; ============================================ ; Outbound Calls via IPComms ; Include from your phones' context: ; include => outbound-ipcomms ; ============================================ [outbound-ipcomms] ; US/Canada 11-digit (1NXXNXXXXXX) exten => _1NXXNXXXXXX,1,NoOp(Outbound: ${EXTEN}) same => n,Set(CALLERID(num)=${YOUR_DID}) same => n,Dial(PJSIP/${EXTEN}@ipcomms) same => n,Hangup() ; US/Canada 10-digit (prepends 1) exten => _NXXNXXXXXX,1,NoOp(10-digit: ${EXTEN}) same => n,Set(CALLERID(num)=${YOUR_DID}) same => n,Dial(PJSIP/1${EXTEN}@ipcomms) same => n,Hangup() ; International (011 + country code) exten => _011.,1,NoOp(Intl: ${EXTEN}) same => n,Set(CALLERID(num)=${YOUR_DID}) same => n,Dial(PJSIP/${EXTEN}@ipcomms) same => n,Hangup() ; Emergency exten => 911,1,Dial(PJSIP/911@ipcomms) exten => 9911,1,Goto(911,1)

Inbound Dialplan

extensions.conf - Inbound from IPComms
; ============================================ ; Inbound Calls from IPComms ; ============================================ [from-ipcomms] ; Route main DID to extension 100 exten => 15551234567,1,NoOp(Inbound: ${CALLERID(all)}) same => n,Dial(PJSIP/100,30) same => n,VoiceMail(100@default,u) same => n,Hangup() ; Route second DID to ring group exten => 15559876543,1,NoOp(Sales line) same => n,Dial(PJSIP/200&PJSIP/201&PJSIP/202,30) same => n,VoiceMail(200@default,u) same => n,Hangup() ; Catch-all exten => _X.,1,Answer() same => n,Playback(tt-allbusy) same => n,Hangup()

Understanding NANPA Patterns

The pattern _1NXXNXXXXXX is a standard North American Numbering Plan pattern:

  • _ -- Pattern match prefix
  • 1 -- Country code (US/Canada)
  • N -- Digits 2-9
  • X -- Digits 0-9
  • . -- One or more of any digit

Add include => outbound-ipcomms to your phones' context to enable outbound calling through the IPComms trunk.

Step 6: Enable TLS/SRTP Encryption (Recommended)

For production environments, we strongly recommend TLS for SIP signaling and SRTP for media encryption. IPComms supports TLS 1.2 and TLS 1.3 with SRTP at no additional cost.

TLS encrypts SIP signaling (preventing eavesdropping on call metadata), while SRTP encrypts the audio stream. Together they provide end-to-end encryption between your Asterisk server and the IPComms network.

TLS Transport

pjsip.conf - TLS Transport
[transport-tls] type = transport protocol = tls bind = 0.0.0.0:5061 cert_file = /etc/asterisk/keys/asterisk.pem ca_list_file = /etc/asterisk/keys/ca-bundle.crt method = tlsv1_2 external_media_address = YOUR_PUBLIC_IP external_signaling_address = YOUR_PUBLIC_IP local_net = 10.0.0.0/8 local_net = 172.16.0.0/12 local_net = 192.168.0.0/16

Enable SRTP on the Endpoint

Add to your [ipcomms] endpoint:

Add to endpoint section
transport = transport-tls media_encryption = sdes media_encryption_optimistic = yes

Update your AOR contact for TLS:

TLS AOR Contact
contact = sip:s1.ipcomms.net:5061\;transport=tls

TLS Certificates

Use Let's Encrypt for free certificates, or the Asterisk ast_tls_cert utility in contrib/scripts/. IPComms accepts both CA-signed and self-signed certificates.

Step 7: Apply Configuration and Test

Reload the PJSIP module and verify connectivity. PJSIP can reload without restarting the Asterisk service, so existing calls are not interrupted.

Asterisk CLI Commands
# Reload PJSIP (no restart needed) $ asterisk -rx "module reload res_pjsip.so" $ asterisk -rx "dialplan reload" # Verify endpoint is loaded $ asterisk -rx "pjsip show endpoints" # Check endpoint details $ asterisk -rx "pjsip show endpoint ipcomms" # Verify AOR contact status $ asterisk -rx "pjsip show aor ipcomms" # Check registration (if using reg auth) $ asterisk -rx "pjsip show registrations"

Expected Output

Expected Output
Endpoint: ipcomms Not in use Aor: ipcomms 1 Contact: ipcomms/sip:s1.ipcomms.net:5060 Avail 5.012

The Avail status confirms Asterisk can reach IPComms. Make a test call and check the CDR in the IPComms portal.

Troubleshooting Common Issues

Here are the most common problems and solutions when setting up your Asterisk PJSIP trunk:

One-Way Audio or No Audio

Almost always a NAT issue. Verify:

  • external_media_address set to your correct public IP
  • direct_media = no on the endpoint
  • rtp_symmetric = yes on the endpoint
  • Firewall allows UDP 10000-20000 in both directions

Registration Failures (401/403)

  • Double-check credentials in [ipcomms-auth]
  • Verify client_uri format: sip:username@s1.ipcomms.net
  • Ensure account is active in the portal

403 Forbidden on Outbound

  • IP auth: Verify your public IP is whitelisted in the portal
  • Check if IP changed (cloud instance reboot)
  • Reg auth: Confirm registration is active via pjsip show registrations

Codec Mismatch (488)

  • IPComms supports G.711 ulaw and G.711 alaw
  • Ensure disallow=all comes before allow=ulaw

Debug Commands

Debug Commands
# Enable SIP logging $ asterisk -rx "pjsip set logger on" # Watch real-time $ asterisk -rvvvvv # Check modules $ asterisk -rx "module show like pjsip" # View transports $ asterisk -rx "pjsip show transports" # Active channels $ asterisk -rx "core show channels" # Disable debug when done! $ asterisk -rx "pjsip set logger off"

Security Reminder

Always disable pjsip set logger on after debugging. SIP logs contain sensitive information and generate significant I/O.

Need Help?

IPComms provides real human technical support from VoIP engineers who understand Asterisk. Contact support for personalized help.

Why Choose IPComms for Your Asterisk Trunk

Not all SIP trunking providers are equal. Here is why thousands of Asterisk administrators trust IPComms as their Asterisk VoIP provider:

23+ Years in Business

Enterprise VoIP since 2002. Stable, reliable, not going anywhere.

$0.01/min US & Canada

Crystal-clear G.711 at one penny per minute. No hidden fees.

TLS/SRTP Included Free

Full encryption at no extra cost. Other providers charge premium for this.

Real Human Support

VoIP engineers who understand Asterisk and PJSIP. No chatbots.

Instant Provisioning

Trunk active immediately. Free DID with every account.

No Contracts

Month-to-month. Cancel anytime, no penalties.

Whether you run a home lab, small business PBX, or enterprise call center, IPComms scales with you. Learn more or get started today.

Ready to Connect Your Asterisk to IPComms?

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