SIP Trunk Setup for Incredible PBX / PBX in a Flash

Use Voyced as an external SIP provider

This guide explains how to connect your Voyced telephone number and SIP trunk to Incredible PBX or PBX in a Flash (PIAF).

Both platforms use Asterisk and FreePBX to manage telephone calls, SIP trunks, extensions and routing.

Once configured, your telephone system can make and receive business calls through Voyced using SIP registration.

This manual covers:

• Incredible PBX 2025 and 2026

• Incredible PBX 2027 installations

• Earlier Incredible PBX installations

• PBX in a Flash systems using FreePBX

Modern installations use PJSIP. Some older systems use the legacy SIP driver, chan_sip.

The main instructions use PJSIP. A separate section covers older systems.

What You'll Need

Before you begin, make sure you have:

• Administrator access to Incredible PBX or PBX in a Flash

• Access to the FreePBX web administration interface

• A working internet connection

• At least one configured telephone extension

• A registered telephone or softphone for testing

• Permission to manage SIP trunks and call routing

• An active Voyced telephone number

Your Voyced SIP details:

Replace the example username and password with your actual Voyced service details.

Requirements Before You Begin

Your PBX must have:

• A working IP address, subnet mask and gateway

• DNS resolution for sip.voyced.eu

• Internet access for SIP signalling

• Suitable RTP connectivity for telephone audio

• Sufficient processing capacity for your simultaneous calls

• A working Asterisk service

• At least one internal extension

Recommended audio codecs:

• G.711 A-law / PCMA

• G.711 μ-law / PCMU

Use full international telephone numbers without a leading + when sending calls to Voyced.

For example:

442012345678

Check Your Software Version

Before creating the trunk, check which SIP driver your installation supports.

For new connections, use PJSIP whenever your installation supports it.

The traditional chan_sip driver is not available in Asterisk 21 and later.

If your system uses an older version of Asterisk with chan_sip enabled, follow the legacy instructions later in this guide.

SIP Trunk Setup Steps

1. Log in to FreePBX Administration

1. Open your Incredible PBX or PIAF web administration interface.

2. Log in using your administrator credentials.

3. Open the FreePBX administration area.

4. Navigate to Connectivity → Trunks.

This page lists the configured telephone providers.

Incredible PBX may include several preconfigured or disabled provider trunks.

Do not overwrite another provider's configuration.

For Voyced, create a separate SIP trunk.

2. Create the Voyced PJSIP Trunk

Click:

Add Trunk → Add SIP (chan_pjsip) Trunk

The trunk editor opens.

Under the General tab, enter:

Trunk Name

Use:

voyced

This identifies the SIP trunk inside FreePBX and Asterisk.

Outbound Caller ID

Enter your authorised Voyced telephone number.

For example:

442012345678

You can also use the format:

"Business" <442012345678>

Replace the example number and business name.

CID Options

For a simple installation using one outbound telephone number, select:

Force Trunk CID

This makes the trunk present its configured caller ID rather than accepting another number from an extension.

If you have several Voyced numbers authorised for caller ID presentation, use the caller ID configuration appropriate to those numbers.

Maximum Channels

This setting limits simultaneous outbound calls using the trunk.

For example:

Use the actual capacity included in your Voyced service.

This field does not impose a combined inbound and outbound channel limit.

3. Configure PJSIP Authentication

Open:

pjsip Settings → General

Enter:

Username

Enter your Voyced SIP username.

For example:

442012345678

Secret

Enter your Voyced SIP password.

Authentication

Select:

Outbound

This allows the PBX to authenticate with Voyced.

Registration

Select:

Send

This tells FreePBX to register its SIP account with Voyced.

Do not select Receive or None.

SIP Server

Enter:

sip.voyced.eu

SIP Server Port

Enter:

5060

Context

Use:

from-pstn

This is the standard incoming trunk context for FreePBX.

Transport

Select the UDP transport configured on your PBX.

Depending on your version, it may appear as:

0.0.0.0-udp

The local UDP listening port can differ from the provider destination port.

Do not change your working local SIP transport simply because Voyced uses destination port 5060.

4. Configure the PJSIP Advanced Settings

Open:

pjsip Settings → Advanced

Configure:

Contact User

Enter your Voyced telephone number:

Your-Voyced-DID

This controls the user portion of the Contact header used for registration.

For a single-number installation, using the registered telephone number helps keep the inbound configuration clear.

From User

Enter:

Your-Voyced-DID

This makes the outgoing SIP From username match the registered account.

For the initial setup, use the same telephone number for the SIP account and outbound caller ID.

If your organisation uses separate authorised numbers for extensions, the From User setting may need adjustment after the initial connection has been tested.

From Domain

Enter:

sip.voyced.eu

Direct Media

Select:

No

This keeps telephone audio routed through Asterisk rather than attempting to connect your telephone directly to the external provider.

5. Check the Registration URI Settings

Some FreePBX versions show additional PJSIP settings:

• Client URI

• Server URI

• AOR Contact

These may be available under the Advanced tab or an advanced configuration mode.

For a standard Voyced account, the expected values are:

Example Client URI:

sip:442012345678@sip.voyced.eu

Example Server URI:

sip:sip.voyced.eu:5060

These values describe the account identity and SIP registrar.

FreePBX normally generates the required registration details automatically from the General settings.

If the automatically generated settings are correct, leave the advanced URI fields unchanged.

Only enter them manually if your FreePBX version requires them or troubleshooting identifies a problem.

6. Configure the Audio Codecs

Open:

pjsip Settings → Codecs

Select:

1. alaw

2. ulaw

Recommended configuration:

If the interface allows codec ordering, place alaw first.

These are the two standard G.711 telephone codecs.

For DTMF signalling, retain the standard RFC4733 configuration.

This supports keypad input for voicemail, IVR menus and other telephone services.

7. Save and Activate the Trunk

After completing the settings:

1. Review the username and password.

2. Confirm Authentication is Outbound.

3. Confirm Registration is Send.

4. Check the SIP server and destination port.

5. Confirm the trunk is enabled.

6. Click Submit.

7. Click Apply Config.

FreePBX saves the configuration and applies it to Asterisk.

The Voyced trunk should now appear in the trunk list.

A saved trunk does not necessarily mean registration has succeeded.

Check its registration status next.

8. Verify SIP Registration

If you have administrator access to the Incredible PBX server, open a terminal.

Run:

asterisk -rx "pjsip show registrations"

Look for the Voyced connection.

Expected status:

Registered

The registration should identify:

sip.voyced.eu

For additional information, run:

asterisk -rx "pjsip show endpoints"

You can also inspect the specific trunk:

asterisk -rx "pjsip show endpoint voyced"

A successful registration confirms that your PBX has authenticated with Voyced.

It does not confirm incoming and outgoing routing.

If registration fails, check the troubleshooting section before continuing.

9. Create an Incoming Call Route

To receive calls on your Voyced telephone number:

1. Navigate to Connectivity → Inbound Routes.

2. Click Add Inbound Route.

3. Open the General tab.

Enter:

For example:

The DID Number must match the telephone number received by FreePBX.

If the SIP provider delivers the called number in another format, enter the matching DID value.

Click Submit.

Then click Apply Config.

10. Configure an Incoming Ring Group

If several employees should answer the same telephone number, create a ring group.

Navigate to:

Applications → Ring Groups

Click Add Ring Group.

Enter:

Example:

This example rings extensions 101, 102 and 103.

Use extension numbers configured on your PBX.

Click Submit and Apply Config.

Return to:

Connectivity → Inbound Routes

Edit the Voyced Reception route.

Under Set Destination, select:

Ring Groups → Voyced Reception

Save the changes.

Incoming calls will now follow the ring group's settings.

11. Configure Additional Voyced Numbers

If several telephone numbers are delivered through the same Voyced SIP account, you can use the existing registration for those numbers.

Create a separate inbound route for each DID.

Example:

For every number:

1. Open Connectivity → Inbound Routes.

2. Click Add Inbound Route.

3. Enter the DID.

4. Choose the destination.

5. Submit and apply the configuration.

The DID format must match the incoming telephone number received by FreePBX.

If your numbers belong to different SIP accounts with separate credentials, configure the appropriate additional trunks.

12. Create an Outbound Route

To make outgoing calls through Voyced:

1. Navigate to Connectivity → Outbound Routes.

2. Click Add Outbound Route.

3. Open Route Settings.

Enter:

Under:

Trunk Sequence for Matched Routes

Select:

voyced

as the first trunk.

Example:

If your organisation uses several providers, configure the correct priority for each route.

The trunk sequence determines which provider handles a call after the outbound route matches the dialled number.

13. Configure International Dialling

Open the outbound route's Dial Patterns tab.

FreePBX uses three important fields:

• Prepend

• Prefix

• Match Pattern

These control how outgoing numbers are modified.

Example: Users Dial 9 for an Outside Line

Suppose your users dial:

9442012345678

The first digit, 9, is the internal outside-line access code.

The destination number is:

442012345678

Configure:

The prefix field tells FreePBX to match and remove the leading 9.

The Match Pattern identifies the remaining number.

The result is:

The pattern ZXXXXXX. is a broad starting example for international numbers.

It matches a number beginning with digits 1 to 9 and containing at least eight digits.

Use more specific patterns where you need to restrict destinations.

If Users Dial International Numbers Directly

If users already dial:

442012345678

without an access prefix, configure:

No digits need to be removed.

If Users Dial 00 for International Calls

Suppose a user dials:

00442012345678

If this is your international access format, create a matching rule:

FreePBX removes 00.

The number sent to Voyced becomes:

442012345678

Do not combine several conflicting prefix-removal rules.

Use the dialling method already familiar to your users.

14. Check Trunk Dial Manipulation

FreePBX also provides number modification settings at trunk level.

Navigate to:

Connectivity → Trunks → voyced

Open:

Dialed Number Manipulation Rules

For the initial setup, leave these rules empty.

Your outbound route should already remove the required access prefix.

Configuring additional trunk-level rules may cause the system to remove too many digits or add unwanted prefixes.

For example, if the outbound route has already converted:

00442012345678

to:

442012345678

the trunk should not perform that conversion again.

Save any required changes.

15. Configure Outbound Caller ID

Your outgoing calls should present a telephone number authorised for use with Voyced.

For a simple setup, open:

Connectivity → Trunks → voyced

Under General, enter your authorised telephone number in:

Outbound CallerID

Example:

"Reception" <442012345678>

If all outgoing calls should use this number, select:

CID Options → Force Trunk CID

Individual Extension Caller IDs

If you have several authorised Voyced telephone numbers, FreePBX can present different caller IDs for individual extensions.

Navigate to:

Applications → Extensions

Select an extension.

Find:

Outbound CID

Enter the authorised telephone number.

For example:

The trunk and outbound route caller ID settings determine whether the individual extension's caller ID is retained or overridden.

If you use separate extension caller IDs, make sure Force Trunk CID and the PJSIP From User setting do not prevent the required identity from being sent.

Only present numbers authorised for use through your Voyced account.

16. Configure Outbound Route Priority

FreePBX checks outgoing routes in their displayed order.

Navigate to:

Connectivity → Outbound Routes

Review the current route list.

If several routes match the same dialled number, FreePBX selects the first matching route.

For example:

Keep your approved emergency and special-service routes correctly prioritised.

If Voyced should handle selected international destinations, position its route accordingly.

Do not move a broad Voyced route above more specific routes without checking which calls it will capture.

Click Apply Config after changing the route order.

17. Configure Simultaneous Calls

Your available simultaneous calls depend on:

• The calling capacity of your Voyced service

• Your PBX hardware or virtual server resources

• Any configured trunk or route restrictions

For example:

The Maximum Channels field in FreePBX limits outbound calls through the trunk.

It does not enforce your complete inbound and outbound service capacity.

If your installation needs a combined call limit, configure the appropriate supported call-limit controls.

18. Configure Network and NAT Settings

Navigate to:

Settings → Asterisk SIP Settings

Open the general SIP settings.

If your Incredible PBX system is behind a router performing NAT, review:

Example local network:

192.168.1.0/24

Use your actual network values.

If FreePBX offers Detect Network Settings, you can use it to populate the external address and local network information.

Check the detected values before saving.

SIP Transport

Open the PJSIP transport settings.

Confirm the UDP transport is enabled.

The local SIP listening port depends on your configuration.

Your PBX may use local port 5060 or another configured local SIP port.

The Voyced destination remains:

sip.voyced.eu:5060

RTP Ports

FreePBX installations commonly use:

UDP 10000–20000

for RTP audio.

Check the actual range displayed in your Asterisk SIP Settings before configuring firewall rules.

Save and apply the settings if changes are required.

19. Check the Incredible PBX Firewall

Incredible PBX includes a firewall configuration that can be more restrictive than a standard FreePBX installation.

Depending on the version, it may use:

• IPtables

• IP sets

• Fail2Ban

• Trusted address or hostname rules

The firewall must allow the SIP signalling and RTP traffic required by your PBX.

A successful SIP registration proves the PBX can reach the registrar and receive its response.

Incoming calls may still fail if the provider sends SIP requests from an address that the PBX firewall does not accept.

When this happens, check the actual incoming signalling source and the rules configured for that connection.

Do not open SIP signalling to the entire internet simply to make the trunk work.

For an Incredible PBX installation, use its supported firewall management method rather than changing firewall rules through unrelated tools.

If your current firewall configuration already works, retain it.

20. Configure Incoming Number Recognition

The normal PJSIP trunk context is:

from-pstn

FreePBX uses the called telephone number to select an inbound route.

For example:

442012345678

If registration succeeds but the inbound DID is not recognised:

1. Check the SIP INVITE received by Asterisk.

2. Inspect the Request-URI.

3. Compare the called number with the configured inbound route.

4. Check the Contact User setting.

5. Confirm FreePBX associates the incoming INVITE with the correct trunk.

Some providers present the called number in the SIP To header.

If the DID is available in the To header but not the Request-URI, FreePBX supports the alternative incoming context:

from-pstn-toheader

To use it:

1. Open Connectivity → Trunks → voyced.

2. Select pjsip Settings → General.

3. Change Context to from-pstn-toheader.

4. Submit the change.

5. Click Apply Config.

6. Repeat the incoming call test.

Only use this alternative when the incoming SIP message confirms it is needed.

Legacy SIP Setup for Older Incredible PBX / PBX in a Flash

Older installations may use the traditional Asterisk chan_sip driver.

The following steps apply only to systems where that driver is already supported and enabled.

They do not apply to modern Asterisk 21 or later installations.

21. Create a Legacy SIP Trunk

Navigate to:

Connectivity → Trunks

Select:

Add SIP (chan_sip) Trunk

Under General, enter:

Next, open:

SIP Settings → Outgoing

Enter the PEER Name:

voyced

Under PEER Details, enter:

type=peer
host=sip.voyced.eu
port=5060
username=Your-Voyced-DID
fromuser=Your-Voyced-DID
fromdomain=sip.voyced.eu
secret=YourServicePassword
context=from-trunk
qualify=yes
insecure=port,invite
canreinvite=no
dtmfmode=rfc2833
disallow=all
allow=alaw&ulaw

Replace the username and password.

This configuration sets up the Voyced SIP server and outbound authentication.

The from-trunk context handles incoming calls.

The alaw and ulaw codecs provide standard G.711 audio.

22. Configure Legacy SIP Registration

Open:

SIP Settings → Incoming

Locate:

Register String

Enter:

Your-Voyced-DID:YourServicePassword@sip.voyced.eu:5060/Your-Voyced-DID

Replace the placeholders with your account details.

This tells Asterisk to register with Voyced.

The final telephone number after / helps identify incoming calls.

For example:

442012345678:YourServicePassword@sip.voyced.eu:5060/442012345678

Leave the USER Context and USER Details fields empty unless your installation specifically requires separate incoming user configuration.

Click Submit.

Then click Apply Config.

23. Verify Legacy SIP Registration

On an older system using chan_sip, run:

asterisk -rx "sip show registry"

Look for:

Registered

You can also inspect the peer:

asterisk -rx "sip show peer voyced"

If registration succeeds, use the same incoming and outgoing route configuration described earlier.

For a modern PJSIP trunk, use the PJSIP commands instead.

Do not mix chan_sip and PJSIP configuration fields.

Testing Your Setup

SIP Registration Test

For PJSIP:

asterisk -rx "pjsip show registrations"

For legacy chan_sip:

asterisk -rx "sip show registry"

Expected result:

Registered

If registration is successful, continue with a test call.

Inbound Test

1. Call your Voyced telephone number from an external mobile telephone.

2. Confirm Incredible PBX receives the call.

3. Check that the correct inbound DID matches.

4. Verify the selected extension, ring group or IVR rings.

5. Answer the call.

6. Check audio in both directions.

If several Voyced numbers are configured, test each DID separately.

Outbound Test

1. Pick up a telephone connected to Incredible PBX.

2. Dial an external telephone number using your normal dialling method.

3. Confirm the call uses the Voyced trunk.

4. Check the destination rings.

5. Verify two-way audio.

6. Confirm the correct caller ID appears.

Example with outside-line access code 9:

9442012345678

The PBX must send:

442012345678

to Voyced.

Additional Test

Place a second simultaneous call while the first call is active.

Confirm both calls work within your available service capacity.

Also test an incoming call while an outbound call is in progress.

This checks the combined calling capacity of your connection.

Troubleshooting

SIP Registration Fails

Check:

• SIP server is sip.voyced.eu.

• Username is correct.

• Secret/password is correct.

• Authentication is Outbound.

• Registration is Send.

• SIP server port is 5060.

• UDP transport is enabled.

• DNS resolution works.

For PJSIP, run:

asterisk -rx "pjsip show registrations"

For legacy chan_sip, run:

asterisk -rx "sip show registry"

401 or 407 Error

These responses indicate an authentication challenge.

An initial challenge is normal.

If registration continues to fail, check:

• SIP username

• SIP password

• Authentication realm

• Client URI

• Subsequent authenticated REGISTER requests

403 Error

Check:

• Account status

• SIP credentials

• Caller ID authorisation

• Calling permissions

• Account restrictions

404 Error

Check:

• Outgoing telephone number

• Outbound dial pattern

• Number prefix handling

• SIP Request-URI

• Incoming DID matching

• Which system returned the response

For example, if the PBX sends:

9442012345678

instead of:

442012345678

review the outbound route's Prefix field.

Incoming Calls Do Not Ring

Check:

• SIP registration

• Inbound route

• DID Number

• Incoming trunk context

• Contact User

• Incoming SIP Request-URI

• Extension or ring group destination

If the DID is present in the To header instead of the Request-URI, review whether from-pstn-toheader is required.

Incoming Calls Are Rejected as Unidentified

Check whether Asterisk can associate the incoming INVITE with the Voyced trunk.

Review:

• PJSIP endpoint identification

• Incoming SIP source

• Registration contact information

• Contact User

• Firewall rules

If the provider's incoming SIP signalling uses addresses different from its registrar, inspect the actual source addresses before adjusting the relevant inbound identification settings.

Outbound Calls Fail

Check:

• SIP registration

• Outbound route

• Dial pattern

• Trunk sequence

• Caller ID

• Available account credit

• Calling permissions

• Simultaneous call capacity

Outbound Calls Use the Wrong Provider

Open:

Connectivity → Outbound Routes

Review the route order.

FreePBX selects the first route whose dial pattern matches.

Move the relevant Voyced route to the appropriate position.

Keep emergency and other special-purpose routes correctly prioritised.

Outgoing Number Has the Wrong Format

The final number sent to Voyced must contain the country code without a leading +.

Correct:

442012345678

Check:

• Outbound route Prepend

• Outbound route Prefix

• Outbound route Match Pattern

• Trunk dial manipulation rules

Avoid removing or adding the same prefix in more than one place.

Incorrect Outbound Caller ID

Check:

• Trunk Outbound CallerID

• CID Options

• Route CID

• Extension Outbound CID

• PJSIP From User

• Authorised telephone numbers

For the first test, use your registered Voyced telephone number.

One-Way Audio

Check:

• External Address

• Local Networks

• RTP port configuration

• Firewall rules

• NAT handling

• SDP media addresses

• SIP ALG

• Codec negotiation

Successful SIP registration does not guarantee correct RTP audio.

Calls Disconnect After Approximately 30 Seconds

Check:

• SIP ACK delivery

• NAT configuration

• Firewall session timeouts

• SIP signalling addresses

• Session timers

Trunk Appears Unavailable

For PJSIP:

asterisk -rx "pjsip show endpoints"

Check the Voyced endpoint.

SIP OPTIONS monitoring and SIP registration are different checks.

A trunk may be registered while its OPTIONS status reports unavailable.

Confirm the issue using an actual test call.

Firewall Blocks Incoming Calls

Incredible PBX may use restrictive firewall rules.

Check whether incoming SIP packets reach your PBX.

If registration succeeds but incoming calls never appear in the SIP log, review the firewall and the source of the incoming SIP requests.

Do not change your existing network configuration without confirming where the traffic is being blocked.

Reviewing SIP Logs

Open the Asterisk CLI:

asterisk -rvvv

For PJSIP, enable SIP logging:

pjsip set logger on

Repeat the failed call.

Look for:

• REGISTER

• INVITE

• Request-URI

• From

• To

• Authorization

• SIP response code

• SDP audio information

Disable logging afterwards:

pjsip set logger off

For older chan_sip installations, enable debugging with:

sip set debug on

Disable it afterwards:

sip set debug off

Remove passwords and authentication details before sharing SIP traces.

Registration Works but Calls Still Fail

A registered trunk confirms that the SIP account has authenticated.

It does not confirm complete telephone service.

Test separately:

1. SIP registration

2. Incoming DID routing

3. Outbound route selection

4. International number formatting

5. Outbound caller ID

6. RTP audio

Final Tips

• Keep the SIP server set to sip.voyced.eu.

• Use your Voyced telephone number as the SIP username.

• Use your Voyced service password for authentication.

• Use PJSIP for current Incredible PBX installations.

• Set Authentication to Outbound.

• Set Registration to Send.

• Use UDP port 5060 as the SIP destination.

• Enable G.711 A-law and G.711 μ-law.

• Configure the correct inbound DID route.

• Configure an outbound route using the Voyced trunk.

• Remove internal dialling prefixes before sending calls.

• Use full international telephone numbers without a leading +.

• Present an authorised caller ID.

• Check your Incredible PBX firewall and NAT settings.

• Keep existing emergency and special-service routing correctly configured.

• Test SIP registration, inbound calls and outbound calls separately.

Incredible PBX and PBX in a Flash support external SIP providers through their FreePBX and Asterisk telephone systems.

With the correct SIP registration, inbound routing and outbound dial patterns, your PBX can use Voyced for incoming and outgoing business telephone calls.

Last updated: 27/10/2026

© Voyced
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.