SIP Trunk Setup for Voyced – FreePBX

Connect your FreePBX telephone system to Voyced for incoming and outgoing calls using a SIP trunk.

This guide covers FreePBX 17 and FreePBX 16 using PJSIP. It also includes the existing chan_sip settings for older FreePBX installations.

What You'll Need

Before starting, make sure you have:

• Administrator access to your FreePBX system.

• At least one working FreePBX extension.

• An active Voyced telephone number.

• Your Voyced SIP username and Service Password.

• A working internet connection.

Voyced SIP details

Replace the example username and password with your own Voyced credentials.

You can find your DID under Cloud PBX / IP Centrex > Phone Lines in your Voyced Hosted IPPBX. Select the wrench icon beside your number.

Your Service Password is available under My Profile > Settings.

The Service Password is different from your Voyced website login password.

Requirements Before You Begin

1. Confirm that your FreePBX system has internet access.

2. Check that your firewall permits SIP traffic to sip.voyced.eu on UDP port 5060.

3. Allow RTP audio traffic through the ports configured on your FreePBX system. The common default range is UDP 10000–20000.

4. If your FreePBX server is behind a router, check the External Address and Local Networks settings under Settings > Asterisk SIP Settings.

5. Make sure at least one extension can make and receive internal calls.

Which trunk type should you use?

• FreePBX 17: Use PJSIP (chan_pjsip).

• FreePBX 16: PJSIP is recommended for new trunks.

• Existing older installations: A working chan_sip trunk may be retained while you plan an upgrade.

The following setup uses PJSIP.

SIP Trunk Setup

Step 1: Log in to FreePBX

1. Open your FreePBX administrator interface.

2. Log in using your administrator credentials.

3. Navigate to Connectivity > Trunks.

Step 2: Create a SIP trunk

1. Click Add Trunk.

2. Select Add SIP (chan_pjsip) Trunk.

Under the General tab, enter:

Leave the other options at their defaults.

Step 3: Configure PJSIP settings

Open pjsip Settings > General.

Enter:

Select the available UDP transport if its name differs.

Keep the account enabled for registration.

Step 4: Configure advanced SIP settings

Open pjsip Settings > Advanced.

Use:

Leave other advanced settings at their defaults.

The Contact User setting helps FreePBX recognise incoming calls when your trunk serves a single Voyced DID.

If you have several telephone numbers on one trunk, the incoming DID must be identified from the information sent with each call.

Step 5: Configure audio codecs

Open pjsip Settings > Codecs.

Enable:

1. alaw (PCMA)

2. ulaw (PCMU)

These codecs provide broad compatibility with Voyced.

G.729 may also be used if supported and correctly licensed.

Click Submit, followed by Apply Config.

Step 6: Create an inbound route

This determines where incoming Voyced calls are sent.

1. Open Connectivity > Inbound Routes.

2. Click Add Inbound Route.

Configure:

The DID Number must match the number received by FreePBX.

For a single-number trunk, use the same DID entered under Contact User.

Click Submit and Apply Config.

Step 7: Create an outbound route

This determines which trunk FreePBX uses when making external calls.

1. Open Connectivity > Outbound Routes.

2. Click Add Outbound Route.

Under Route Settings:

Under Dial Patterns, add the numbers you want to route through Voyced.

For international calls, use:

This matches international numbers beginning with 00.

Add suitable national or local patterns for the countries and services you use.

Do not remove the leading 00 from international numbers.

Click Submit and Apply Config.

Step 8: Check registration

Open the Asterisk command line on your FreePBX server:

sudo asterisk -rvvv

Enter:

pjsip show registrations

Your Voyced trunk should show:

Registered

You can also check:

pjsip show endpoints

If the trunk is registered, continue with the call tests below.

Existing FreePBX 16 installations using chan_sip

If your older FreePBX system already uses a working chan_sip trunk, you can retain the existing configuration.

For reference, the original Voyced trunk used these outgoing PEER details:

username=Your-Voyced-DID
type=peer
secret=YourServicePassword
fromuser=Your-Voyced-DID
fromdomain=sip.voyced.eu
host=sip.voyced.eu
port=5060
qualify=yes
nat=force_rport,comedia
directmedia=no
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw

The registration string is:

Your-Voyced-DID:YourServicePassword@sip.voyced.eu/Your-Voyced-DID

For this legacy trunk:

• Keep the Incoming USER Details empty unless your existing configuration requires them.

• Retain the inbound and outbound routes that already work.

• Restrict incoming SIP traffic to trusted sources using your PBX firewall.

• Use PJSIP when creating a new FreePBX 17 trunk.

To check a legacy trunk's registration:

sip show registry

The sip commands apply to chan_sip. For PJSIP, use the pjsip commands shown earlier.

Test Your Connection

Once your trunk shows Registered, complete the following tests.

Test 1: Outgoing calls

1. Use a FreePBX extension.

2. Dial another telephone number.

3. Confirm the call connects.

4. Check audio in both directions.

5. Verify that the recipient sees your authorised Caller ID.

Test 2: Incoming calls

1. Call your Voyced DID from another telephone.

2. Confirm FreePBX receives the call.

3. Check that the inbound route sends it to the correct extension, ring group or IVR.

4. Answer the call and test audio.

Test 3: International calls

All international calls through Voyced must use:

00 + Country Code + Telephone Number

For example:

00442035000202

Make sure FreePBX passes the complete number to Voyced without removing 00.

Test 4: Voicemail

Voyced voicemail is available by dialling:

1233

If you want FreePBX extensions to access Voyced voicemail, create an outbound route for 1233 through your Voyced trunk.

Check that this does not conflict with a local FreePBX feature code.

FreePBX also offers its own voicemail system. You can use that instead if you prefer to keep voicemail within your PBX.

Test 5: Simultaneous calls

If your Voyced service supports multiple simultaneous calls, test the required call capacity.

The available number of calls depends on your Voyced service, not simply on the number of extensions configured in FreePBX.

Troubleshooting

Trunk does not register

Check:

• SIP Server: sip.voyced.eu

• Port: 5060

• Transport: UDP

• Username: Your Voyced DID

• Secret: Your Service Password

• Authentication: Outbound

• Registration: Send

Make sure your PBX can reach the SIP server.

Check that your password is the Voyced Service Password, not your website login password.

Incoming calls do not work

• Confirm the trunk is registered.

• Check the inbound route and its destination.

• Make sure Contact User matches your DID on a single-number trunk.

• Check that the inbound DID matches the number received by FreePBX.

• Review your PBX firewall settings.

If FreePBX reports no matching inbound route, check the DID shown in the Asterisk logs.

Outgoing calls do not work

• Check that your extension matches an outbound route.

• Confirm the route uses the Voyced trunk.

• Verify the dialled number.

• Check your available Voyced call credit.

• Confirm the destination is permitted.

International calls fail

Check that the dialled number begins with 00.

Review the outbound route's Prefix and Prepend fields and the trunk's number manipulation rules.

The international prefix must reach Voyced unchanged.

No audio or one-way audio

• Check External Address and Local Networks under Asterisk SIP Settings.

• Confirm your firewall allows the configured RTP ports.

• Disable Direct Media on the trunk.

• Start with PCMA and PCMU.

• Check that the telephone and PBX network settings are correct.

Incorrect Caller ID

FreePBX lets you set Caller ID at extension, outbound route and trunk level.

The number you send must be authorised on your Voyced account.

If you need another outbound Caller ID, open a Voyced sales support ticket with the subject:

Outbound Caller ID adjustment

Include your Voyced DID and the telephone number you want displayed.

Numbers outside your Voyced account require proof of ownership and additional verification.

SIP debugging

For PJSIP, open the Asterisk CLI and enter:

pjsip set logger on

Make a test call and review the SIP responses.

To stop logging:

pjsip set logger off

Useful commands:

When contacting Voyced Support, provide the time, destination number and result of the failed call.

Final Tips

• Use sip.voyced.eu as your main SIP server.

• Keep registration enabled for your Voyced PJSIP trunk.

• Use UDP port 5060 for the standard connection.

• Use PCMA and PCMU for reliable audio compatibility.

• Keep international numbers in the required 00 format.

• Make sure your inbound and outbound routes point to the correct destinations.

• Use only Caller ID numbers authorised for your Voyced service.

• Keep your FreePBX system updated and your SIP access protected.

• Confirm your configured call limits match your Voyced service.

• Test incoming and outgoing calls after changing your trunk or routing settings.

Once registration, incoming calls, outgoing calls and audio have been tested successfully, your FreePBX system is ready to use with Voyced.

Last updated: 27/10/2026

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All information provided is copyrighted and purely informative in nature. It is maintained by Voyced with the greatest of care using reliable sources only.
Given the fast evolution of the telecom sector, Voyced does not offer any guarantee about the accuracy and completeness of the information offered. 
We explicitly do not give anyone permission to use this information in any other way than to inform themselves about what (legal) requirements there may be and they need to comply with when registering Services and/or DIDs with Voyced.