SIP Trunk Setup for ASTPP

Use Voyced as an external SIP provider

This guide explains how to connect your Voyced telephone number and SIP trunk to ASTPP.

ASTPP is a VoIP billing and switching platform built on FreeSWITCH. It supports external SIP providers, SIP registration, call routing, telephone number management and call billing.

Once configured, your ASTPP system can make and receive business telephone calls through Voyced.

This manual covers ASTPP v7 and recent ASTPP installations. Older versions use similar settings, although some menu names differ.

The instructions use the ASTPP web administration interface.

What You'll Need

Before you begin, make sure you have:

• Administrator access to your ASTPP installation

• A working ASTPP and FreeSWITCH system

• At least one configured SIP device or telephone

• A working internet connection

• Permission to manage gateways, providers and trunks

• Permission to configure routing and tariffs

• An active Voyced telephone number

Your Voyced SIP details:

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

Requirements Before You Begin

Your ASTPP installation must have:

• A working FreeSWITCH service

• Correct network settings

• DNS resolution for sip.voyced.eu

• An available SIP profile

• Sufficient simultaneous call capacity

• A configured customer account for testing

• A SIP device registered to ASTPP

Recommended audio codecs:

• PCMA / G.711 A-law

• PCMU / G.711 μ-law

Telephone Number Format

Voyced expects outgoing telephone numbers in full international format without a leading +.

For example:

442012345678

Do not send:

+442012345678

If your ASTPP system uses another dialling format internally, configure the relevant routing rules to convert the number before sending it to Voyced.

Permitted Calling Traffic

ASTPP is designed for VoIP switching and billing. It can handle a wide range of call volumes and traffic patterns.

Voyced provides retail-quality telephone services for normal business conversations.

You can use Voyced for:

• Incoming customer calls

• Ordinary business conversations

• Individually initiated outbound calls

• Normal customer callbacks

Predictive dialling, automated bulk calling and short-burst traffic are not permitted.

Configure your ASTPP routing and calling limits to match normal business telephone usage.

SIP Trunk Setup Steps

1. Log in to ASTPP

1. Open your ASTPP administration interface.

2. Log in using your administrator credentials.

3. Open the main administration dashboard.

4. Locate the Carriers menu.

ASTPP separates external SIP providers into several components.

These components work together to provide outbound calling.

Incoming telephone numbers are configured separately under the Inbound menu.

2. Check Your SIP Profile

Navigate to:

Carriers → Gateways

ASTPP associates gateways with a FreeSWITCH SIP profile.

Before creating Voyced, check which SIP profile your installation uses for external carriers.

Common FreeSWITCH profiles include:

• external

• A custom external carrier profile

• Another profile created for your ASTPP installation

For a standard installation, an existing external SIP profile is normally suitable.

Check:

The local SIP listening port is separate from Voyced's destination port.

For example, FreeSWITCH may listen locally on one port while sending calls to Voyced on UDP port 5060.

If your external SIP profile already works with another carrier, retain its existing network settings.

3. Create the Voyced Gateway

Navigate to:

Carriers → Gateways

Click Create or Add Gateway.

Enter:

The Gateway Name identifies the connection inside ASTPP and FreeSWITCH.

Use:

voyced

The Proxy / Host must be:

sip.voyced.eu

Do not enter a fixed server IP address.

The gateway must be configured to register with Voyced.

4. Configure SIP Registration

Open the registration settings for the new gateway.

Depending on your ASTPP version, these settings may appear directly in the gateway form or under its advanced settings.

Enter:

The Authentication Username is sometimes called auth-username.

For the standard Voyced connection, use the same username for registration and authentication.

Example:

442012345678

The registration expiry may be adjusted during negotiation with the SIP server.

Use the interval accepted by the registrar.

Authentication Realm

If ASTPP provides a Realm field, use:

sip.voyced.eu

as the initial value.

If the SIP server returns a different realm during authentication, use the realm supplied in that challenge.

Save the registration settings.

5. Configure SIP Identity

Open the gateway's advanced settings.

Configure:

The From User identifies your account in outgoing SIP messages.

The From Domain identifies the SIP domain.

The Extension setting controls the incoming extension associated with the registration.

For example:

442012345678

This is useful for receiving calls to your primary Voyced telephone number.

If you have several DIDs on the same account, the actual called number received by FreeSWITCH must still be checked during incoming call testing.

6. Configure the SIP Contact Header

FreeSWITCH normally generates a SIP Contact header during registration.

Depending on your gateway options, you may see:

Extension in Contact

For an initial single-number setup, you can enable this setting.

It tells FreeSWITCH to use your configured Extension value in the Contact header rather than its generated gateway identifier.

For example, the Contact username becomes:

442012345678

instead of a generated gateway username.

If your installation already registers and receives calls successfully without this option, retain the working configuration.

The Contact host address must be reachable for incoming calls.

For systems behind NAT, check the public SIP address configured on the external profile.

7. Configure Audio Codecs

Select the codecs permitted for the Voyced connection.

Recommended:

Where your ASTPP gateway provides a codec list, use:

PCMA,PCMU

For FreeSWITCH, the codec setting may also be applied through the SIP profile or outbound call variables.

The effective codec list for the Voyced connection should contain PCMA and PCMU.

DTMF

Use RFC2833/RFC4733 telephone-event signalling.

This supports keypad input during calls, including voicemail and IVR menus.

RTP Packet Time

Use 20 ms where a packet-time setting is available.

Save your audio settings.

8. Save and Activate the Gateway

After completing the gateway configuration:

1. Check the SIP hostname.

2. Confirm your username and password.

3. Confirm registration is enabled.

4. Check the selected SIP profile.

5. Confirm UDP transport.

6. Review the audio settings.

7. Save the gateway.

Make sure the new gateway is enabled.

ASTPP manages its FreeSWITCH configuration through the administration system.

Allow the updated gateway configuration to be applied before testing registration.

9. Verify SIP Registration

If you administer your own ASTPP server, FreeSWITCH provides a command to check gateway registration.

Open a terminal and run:

fs_cli -x "sofia status gateway voyced"

Look for the registration status.

The expected state is:

REGED

This indicates that FreeSWITCH has successfully registered with Voyced.

You can also list all configured gateways:

fs_cli -x "sofia status gateway"

Find:

voyced

If the gateway is not registered, review its credentials and SIP profile.

A gateway status such as NOREG is expected for gateways deliberately configured without registration.

For the standard Voyced connection, registration must be enabled.

10. Create the Voyced Provider

ASTPP uses provider accounts to manage external carrier relationships.

Navigate to:

Accounts → Providers

Click Create Provider.

Enter the provider details.

Complete any additional account fields required by your ASTPP installation.

The provider account represents Voyced inside ASTPP's billing and routing system.

It does not replace your SIP gateway credentials.

Those remain configured in the gateway.

Save the provider.

11. Create the Voyced Trunk

Navigate to:

Carriers → Trunks

Click Create Trunk.

Enter:

The selected Gateway must match the gateway created earlier.

For example:

voyced

The trunk connects the provider account to the gateway that FreeSWITCH uses to make calls.

Simultaneous Call Capacity

Suppose your Voyced service supports four simultaneous calls.

You may configure:

Use the actual capacity available on your service.

If ASTPP offers a separate calls-per-second limit, use a setting suitable for normal, individually initiated calls.

Do not configure rapid automated calling or high-volume dialler traffic.

Save the trunk.

12. Configure Termination Rates

ASTPP uses Termination Rates to determine the cost of sending calls through a provider.

Navigate to:

Carriers → Termination Rates

Click Create.

Select:

Voyced

as the trunk or provider, according to the form shown by your ASTPP version.

Enter the required destination rates.

Important fields include:

Use the call rates and billing increments applicable to your Voyced service.

Do not assume the same tariff applies to every country or telephone number type.

Example Destination Pattern

For calls to the United Kingdom, a destination prefix might be:

44

This is a routing example, not a tariff quotation.

Number Modification

For telephone numbers already in Voyced's required format, leave Strip and Prepend empty.

Example:

If your users dial with an international access prefix such as 00, configure the relevant number conversion so the final number sent to Voyced contains only the country code and telephone number.

Save the termination rate.

Repeat for the destinations your organisation is permitted to call.

13. Create an Outbound Rate Group

ASTPP uses Rate Groups to determine which outbound calling tariffs are available to customers.

Navigate to:

Tariff → Rate Groups

Click Create.

Enter:

Select the Voyced trunk as the preferred trunk where the form provides trunk selection.

The Rate Group controls the calling rates made available to customer accounts.

The provider's Termination Rates control the underlying carrier costs.

These are separate configurations.

Save the Rate Group.

14. Create Origination Rates

Navigate to:

Tariff → Origination Rates

Click Create.

Select:

Voyced Business

as the Rate Group.

Add the destinations that your customer account should be allowed to call.

Important fields include:

For a simple setup, configure the routing type as Priority where supported.

Select the Voyced trunk as the first route.

Example:

Enter the actual rate and billing increment that applies to your account.

Do not enter invented prices simply to complete the form.

Save the Origination Rate.

ASTPP requires appropriate rate coverage before it can route billed outbound calls.

15. Check Your Outgoing Number Format

ASTPP can modify telephone numbers through:

• Customer localisation

• Provider localisation

• Trunk-specific localisation

• Termination Rate Strip and Prepend settings

For Voyced, the final destination must be international digits-only.

For example:

442012345678

Example: International Prefix 00

A user dials:

00442012345678

ASTPP must remove the international access prefix:

00

The final destination becomes:

442012345678

Example: Direct International Dialling

If the customer already sends:

442012345678

no prefix removal is required.

Avoid applying the same digit conversion more than once.

For example, if a customer localisation rule already removes 00, do not create another termination rule that removes two more digits.

16. Configure Localisation If Required

Navigate to:

Switch → Localizations

Create a localisation profile if your existing number format needs conversion.

ASTPP supports separate localisation rules for:

• Customer accounts

• Providers

• Individual trunks

For this connection, a trunk-specific rule is useful when Voyced requires a different number format from other providers.

Review the outgoing destination number conversion.

Configure it so the SIP Request-URI sent to Voyced contains the full international number without a leading +.

The exact rule depends on your starting dialling format.

For example:

Save the localisation rule.

Apply it to the appropriate Voyced trunk or account.

Check the resulting number with a test call.

17. Create a Customer Account

To test outgoing calls through ASTPP, create or select a customer account.

Navigate to:

Accounts → Customers

Click Create Customer.

Enter the required account details.

For the first test, configure:

ASTPP supports prepaid and postpaid accounts.

For prepaid accounts, ensure sufficient balance is available for the test call.

For postpaid accounts, configure a suitable credit limit.

The customer must have the appropriate Rate Group assigned.

Without a matching Rate Group and Origination Rate, outbound calls may be rejected even when the Voyced gateway is successfully registered.

Save the customer.

18. Register a SIP Telephone to ASTPP

ASTPP can create a SIP Device for the customer account.

Navigate to:

Accounts → Customers → SIP Devices

Select the test customer's SIP device.

Find:

• SIP username

• SIP password

• SIP server address

• SIP transport

Configure your telephone or softphone using these details.

The telephone must register with your ASTPP server.

It does not register directly with Voyced.

The connection is:

Telephone
   |
   v
ASTPP / FreeSWITCH
   |
   v
Voyced SIP Gateway
   |
   v
External Telephone Network

Confirm the device has successfully registered.

Make a test internal call if your installation supports internal extension calling.

19. Configure Outbound Caller ID

The number presented to called parties must be authorised for your Voyced service.

ASTPP supports outbound caller ID settings at account and routing level.

For your test customer, use:

Your-Voyced-DID

For example:

442012345678

Review the account's:

• Caller IDs

• Force Caller ID

• CLI Pool settings

The exact caller ID source depends on your account configuration.

For a first test, use one fixed authorised number.

If ASTPP offers an option to reject unrecognised caller IDs, you can use it to prevent accounts presenting unapproved numbers.

After confirming the connection, configure additional authorised caller IDs if your business uses several numbers.

20. Configure Incoming Telephone Numbers

ASTPP manages incoming numbers separately from outbound trunks.

Navigate to:

Inbound → DIDs

Click Create DID.

Enter:

For example:

442012345678

Assign the DID to the correct customer account.

If your installation tracks DID purchase costs or monthly charges, enter the applicable values.

Those ASTPP billing entries do not change your Voyced service charges.

Save the DID.

21. Configure Incoming DID Forwarding

After creating the DID:

1. Open Inbound → DIDs.

2. Select your Voyced number.

3. Open Forwarding.

4. Locate Call Type.

5. Select the appropriate destination type.

For a normal SIP telephone registered to ASTPP, select:

SIP Username

On some versions, the equivalent option is labelled DID-Local.

Enter or select the receiving SIP device.

Example:

Save the forwarding settings.

Incoming calls to your Voyced number should now follow the assigned route, provided FreeSWITCH receives and recognises the called number.

22. Configure a Ring Group or IVR

If you use ASTPP's Multi-Tenant IP PBX addon, additional incoming destinations may be available.

These include:

• Ring Group

• Queue

• IVR

• Conference

• Time Condition

• Voicemail

For example, to route incoming calls to a reception group:

1. Navigate to Applications → Ring Group.

2. Create or select the receiving group.

3. Add the relevant SIP extensions.

4. Save.

Return to:

Inbound → DIDs → Forwarding

Select:

Save.

The Ring Group option requires the relevant ASTPP PBX addon.

If that addon is not installed, use the available SIP Username destination instead.

23. Configure Multiple Voyced Numbers

If several telephone numbers are delivered through the same Voyced SIP account, they can share the existing registration.

Add each number under:

Inbound → DIDs

Example:

For each number:

1. Create the DID.

2. Assign it to the correct ASTPP customer.

3. Configure its forwarding destination.

4. Save.

5. Place an incoming test call.

The DID must match the telephone number received in the SIP INVITE.

A separate SIP registration is not normally required for additional numbers delivered through the same account.

If the additional numbers use separate Voyced SIP credentials, configure separate gateways as required.

24. Check Incoming SIP Number Recognition

The primary gateway registration identifies your ASTPP server to Voyced.

When an incoming call arrives, FreeSWITCH must recognise the called number.

Check:

• The SIP Request-URI

• The SIP To header

• The FreeSWITCH destination number

• The ASTPP DID entry

For example, if Voyced sends:

442012345678

ASTPP must have a matching DID.

Some FreeSWITCH gateway configurations use the registration's Extension value as the incoming destination.

For installations with several DIDs, make sure ASTPP uses the actual called number rather than treating every call as the primary account number.

If several numbers all reach the same destination, inspect the incoming SIP information before changing the DID forwarding rules.

25. Configure Simultaneous Call Limits

ASTPP supports concurrent call controls at several levels.

These include:

• Customer accounts

• Trunks

• Providers

• SIP resources

Your combined calling usage must stay within the capacity of your Voyced service.

For example:

These numbers are examples.

Use the actual limits included with your Voyced account.

If several customers share one Voyced SIP trunk, check their combined usage.

Do not configure rapid automated calling or bulk dialling.

26. Configure Firewall and NAT

Your ASTPP server must communicate with:

sip.voyced.eu

Recommended settings:

Your FreeSWITCH SIP profile determines its local SIP listening port.

This may differ from Voyced's destination port.

If ASTPP Is Behind NAT

Check the SIP profile's:

• External SIP IP address

• External RTP IP address

• Local network information

• RTP port range

FreeSWITCH uses these settings to advertise the correct signalling and media addresses.

If your ASTPP installation already uses a working external SIP profile, retain its network settings.

Your firewall must permit the SIP signalling and RTP audio traffic needed for the connection.

Disable SIP ALG if it interferes with signalling or audio.

Testing Your Setup

SIP Registration Test

On your ASTPP server, run:

fs_cli -x "sofia status gateway voyced"

Expected status:

REGED

If the gateway is not registered, check its hostname, username, password and registration settings.

Outbound Test

1. Register your test SIP telephone to ASTPP.

2. Confirm the customer has the Voyced Business Rate Group.

3. Confirm a matching Origination Rate exists.

4. Confirm a matching Termination Rate exists.

5. Dial an authorised international destination.

6. Check that ASTPP selects the Voyced trunk.

7. Verify the destination rings.

8. Answer the call.

9. Check audio in both directions.

10. Verify the correct caller ID appears.

Example destination:

442012345678

This is the format that should reach Voyced.

Inbound Test

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

2. Confirm FreeSWITCH receives the SIP call.

3. Check the received DID.

4. Verify ASTPP matches the correct DID record.

5. Confirm the assigned SIP device, group or IVR rings.

6. Answer the call.

7. Check two-way audio.

Test each additional DID separately.

Billing Test

After a successful outbound call:

1. Open the ASTPP call detail records.

2. Find the test call.

3. Confirm the destination number.

4. Check the selected provider and trunk.

5. Review the recorded duration.

6. Check the applied Origination Rate.

7. Check the applied Termination Rate.

The test should confirm both call routing and billing are working correctly.

Troubleshooting

SIP Registration Fails

Check:

• SIP server is sip.voyced.eu.

• Username is correct.

• Password is correct.

• Register is enabled.

• Proxy is correct.

• UDP transport is selected.

• Destination port is 5060.

• DNS resolution works.

• The FreeSWITCH SIP profile is running.

Run:

fs_cli -x "sofia status gateway voyced"

401 or 407 Error

These responses indicate an authentication challenge.

An initial challenge is normal.

If registration repeatedly fails, check:

• SIP username

• Authentication username

• SIP password

• Authentication realm

• Subsequent authenticated REGISTER requests

403 Error

Check:

• SIP credentials

• Account status

• Calling permissions

• Caller ID authorisation

• Any service restrictions

404 Error

Check:

• Destination telephone number

• Origination Rate pattern

• Termination Rate pattern

• Selected trunk

• Number conversion

• Incoming DID matching

• Which system returned the error

Gateway Is Configured but Not Registered

A gateway saved in ASTPP does not automatically confirm a successful SIP connection.

Check whether FreeSWITCH has loaded the gateway and sent a REGISTER request.

The expected status for Voyced is:

REGED

If the gateway remains NOREG, confirm that registration is enabled.

Outbound Calls Fail

Check:

• Gateway registration

• Provider account

• Trunk assignment

• Termination Rates

• Customer Rate Group

• Origination Rates

• Customer account balance or credit limit

• Concurrent call limit

• Caller ID

ASTPP may reject an outbound call because no suitable tariff or route exists, even if SIP registration is successful.

No Matching Origination Rate

Check the customer's assigned Rate Group.

Confirm that the dialled number matches an Origination Rate in that group.

For example, a destination beginning with 44 needs an appropriate matching destination rule.

Also confirm the rule selects the Voyced trunk.

No Matching Termination Rate

Open:

Carriers → Termination Rates

Confirm that a matching rate exists for the destination.

Check the trunk assignment and number format.

Outbound Number Has the Wrong Format

Voyced expects digits-only international numbers.

Correct:

442012345678

Check:

• Customer localisation

• Provider localisation

• Trunk localisation

• Termination Rate Strip

• Termination Rate Prepend

Do not apply duplicate prefix removal rules.

Incorrect Outbound Caller ID

Check:

• Account Caller IDs

• Force Caller ID

• CLI Pool

• Trunk caller ID handling

• FreeSWITCH From header

• Authorised telephone numbers

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

Incoming Calls Do Not Ring

Check:

• SIP registration

• Incoming SIP Request-URI

• SIP To header

• FreeSWITCH destination number

• ASTPP DID assignment

• DID Forwarding

• Receiving SIP device registration

If the call reaches FreeSWITCH but ASTPP cannot find a matching DID, compare the received destination number with the configured DID.

All Incoming Numbers Reach One Destination

Check the gateway's Extension setting.

Inspect the actual called number in the incoming SIP INVITE.

If the system routes every call using the registration username, review the incoming number identification.

Confirm that ASTPP can distinguish the DIDs delivered through the same account.

SIP Device Cannot Receive Calls

Check:

• The SIP device is registered to ASTPP.

• The DID belongs to the correct account.

• The forwarding destination is correct.

• The account is active.

• The device is available.

Remember that the customer's SIP device registers to ASTPP, not directly to Voyced.

One-Way Audio

Check:

• RTP connectivity

• FreeSWITCH external RTP address

• Firewall configuration

• NAT handling

• Codec negotiation

• SDP media addresses

• SIP ALG

Successful registration does not guarantee working audio.

Calls Disconnect Unexpectedly

Check:

• SIP ACK delivery

• NAT handling

• Firewall timeouts

• Session timers

• RTP connectivity

• Network packet loss

Too Many Concurrent Calls

Check the trunk and customer concurrent call limits.

The combined incoming and outgoing call usage must stay within your Voyced service capacity.

Reviewing SIP Logs

FreeSWITCH provides SIP tracing for troubleshooting.

To inspect the gateway:

fs_cli -x "sofia status gateway voyced"

To inspect the SIP profile:

fs_cli -x "sofia status profile external"

Replace external with the actual profile name if your gateway uses another profile.

For detailed SIP signalling, you can enable tracing on the relevant profile:

fs_cli -x "sofia profile external siptrace on"

Repeat the failed call.

Look for:

• REGISTER

• INVITE

• Request-URI

• From

• To

• Authorization

• SIP response code

• SDP media information

Disable tracing afterwards:

fs_cli -x "sofia profile external siptrace off"

Remove passwords and authentication information before sharing SIP traces.

Registration Works but Calls Still Fail

A registered gateway confirms that the SIP account has authenticated.

It does not guarantee complete telephone service.

Check these separately:

1. SIP registration

2. Provider and trunk assignment

3. Origination Rate

4. Termination Rate

5. Incoming DID routing

6. Outgoing number formatting

7. Caller ID

8. RTP audio

9. Billing records

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.

• Enable SIP registration on the FreeSWITCH gateway.

• Use UDP destination port 5060.

• Use PCMA and PCMU for audio compatibility.

• Confirm that the gateway reports REGED.

• Create the Voyced provider and trunk in ASTPP.

• Configure Termination Rates using your actual Voyced tariff.

• Create a Rate Group and the required Origination Rates.

• Assign the Rate Group to the customer account.

• Configure the outgoing number format without a leading +.

• Create each incoming DID and assign its forwarding destination.

• Use an authorised outbound caller ID.

• Limit traffic to normal business conversations.

• Test registration, incoming calls, outgoing calls and billing separately.

ASTPP supports external SIP providers through its FreeSWITCH gateway and carrier routing system.

With the correct SIP registration, provider settings, trunk, tariff rules and DID configuration, your ASTPP installation 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.