I. Product Setup Guides

The DrayTek VigorBX 2000 can have up to 12 SIP trunks configured for use with ITSPs (Internet Telephony Service Provider) with up to 50 Alias / DDI numbers that can be linked to the SIP Trunks.

Once a SIP Trunk is configured on the PBX system, the PBX will handle calls as specified by the Answer Mode of each SIP Trunk; this can be used to route calls to the Auto Attendant, Hunt Groups or individual Extensions. Usage of the Answer Modes is explained in this guide: Routing Incoming Calls on the VigorBX 2000

With a SIP Trunk configured, Extensions registered with the PBX system can make calls using the SIP Trunks either by dialling the trunk number then dialling the desired number or by being routed using the Dial Plan facility. This requires that the extension has permission to use the trunk to make outgoing calls. Configuration of the PBX system to make outgoing calls is explained in further detail in this guide: Making Outgoing Calls with the VigorBX 2000

The PBX system listens on UDP port 5070 by default for each SIP Trunk configured, because the PBX system's SIP Extension Registration server operates on UDP port 5060. This will normally work without issue with SIP Trunks that operate using registration, the ports used will be configured automatically during the SIP Trunk Registration process.

SIP Trunks that are IP Authenticated may need to be configured by the ITSP to send SIP packets to PBX system on UDP port 5070. Otherwise, it would be necessary to change the SIP ports used by the PBX system, this guide demonstrates how to change the SIP port configuration: Changing SIP Ports on the VigorBX 2000

To configure a SIP Trunk on the PBX system, go to [IP PBX] > [Trunks] > [SIP Trunk] and select an available Index number:

Details of SIP Trunk configuration options:

• Profile Active: Enable
• Profile Name: Give the SIP Trunk a suitable name
• Registration: Set this to Enable if the SIP trunk registers with a username and password. If the trunk is IP authenticated, set this to Disable
• Register Interface: This can be left on its default setting of Auto so that the SIP Trunk will register over the first available internet connection. If the SIP trunk is IP authenticated, set this to the WAN interface of the internet connection that the SIP Trunk is associated with
• SIP Local Port: Leave this set to 5070. If the SIP Trunk is IP authenticated, ensure that the ITSP is sending SIP packets to UDP port 5070. This settings controls the port that SIP traffic, such as incoming calls for this trunk, are received on
• Domain/Realm: Enter the address of the SIP Server as specified by your ITSP
• Proxy: This is usually the same as Domain/Realm, if your ITSP specifies a proxy to use, enter it here
• Proxy Port: This settings controls the port that SIP traffic for this trunk is sent to. This should be set to the SIP port specified by your ITSP
• Display Name: Set this to the desired SIP caller ID
• Account Number/Name: Enter the account number or name as specified by your ITSP, this can be in the form of a number or account name.
  If the SIP Trunk is IP Authenticated, the number specified here will be matched against incoming calls from the ITSP. If the number the call is addressed to does not match this number or an Alias number configured. the call will be rejected by the PBX system because it would be unable to route the incoming call.
• Authentication ID: Tick this option if the authentication details differ from the Account Number / Name setting and specify the authentication ID as specified by your ITSP
• Password: Set this to the password for the SIP Trunk
• Expiry Time: 1 hour is the default setting, configure this to the value specified by your ITSP
• Trunk Number: This will default to the index number on the PBX system i.e. Trunk 1 = 001. This is the number that Extensions registered with the PBX dial to select a Trunk if Dial Plan is not configured
• Out-Going Call CLI: This can be used to specify a different outgoing Caller ID from the "Account Number" specified above. Set this to Alias number and specify the number if required
• Answer Mode: These options specify where the call is routed depending on which state the PBX system is in i.e. during Office Hours, calls are Forwarded to Hunt Group 101
• Time Budget: If enabled, the PBX will not route calls over this SIP trunk once the budget value has been reached, calls would instead be routed through any Backup Trunk specified in the Dial Plan. This budget value resets at midnight
• Max Simultaneous Call Number: Specify this value if your SIP provider does not limit the number of calls that can go over a trunk at any one time
• Enable Waiting Music: If enabled, the PBX will play this audio to callers in place of the default ringing tone when the PBX has picked up the call i.e. going through Auto Attendant. This will play the prompts selected in order and repeat once the selected prompts have finished playing

Click OK to save and apply these settings.

The [IP PBX] > [Trunks] > [SIP Trunk] list will then show the SIP Trunk with the status of the SIP Trunk displayed on the right:

• R - the SIP Trunk is registered and can be used to make and receive calls
• u - the SIP Trunk has been configured to call without registration (IP authenticated) and can be used to make and receive calls if the SIP settings, IP address and port details are correct
• - Indicates that the SIP Trunk is configured to register and is currently not registered. It cannot be used to make or receive calls in this state

How do you rate this article?