Prerequisites

1. SIP Trunk credentials

Description

This document describes how to link a legacy PBX (e.g. Nortel, Shoretel, Cisco, Avaya, etc...) to a Grandstream PRI Gateway, the gateway then connects to Unitedcloud via a SIP Trunk and can process inbound and outbound calls.

If this is a brand new device, please log in and follow the setup wizard for the network settings, all other settings will be setup by following this KB. 

SIP Trunk Setup

1. Log into the Grandstream GXW4501's webgui. The IP address can be found on the screen of the device itself. 
   
   
2. Once logged in, from the left hand menu navigate to Trunks -> VoIP Trunks
   
   
   
3. Select "Add SIP Trunk" from the top menu. 
   
   
   
4. Fill in the required information. 
   
   
1. Type - Register SIP Trunk
   
2. Provider Name - Enter the desired provider name you wish to use to identify the connection
   
3. Hostname - sip.siplogin.ca
   
4. Keep Original CID - this is your preference on how CID is handled. 
   
5. Keep Trunk CID - this is your preference on how CID is handled. 
   
6. NAT - This is dependent on how your network is setup. If the PRI is behind a router, it is advised to leave this disabled. 
   
7. TEL URI - Leave as Disabled
   
8. From Domain - the domain name as shown in the portal. 
   
9. Transport - Leave at UDP for the default, unless otherwise directed to use TCP by support
   
10. Need Registration - Enabled
    
11. Allow Outgoing calls if registration fails - Disabled
    
12. CallerID Name: Trunk caller ID if using a global caller ID name. 
    
13. Username - SIP Username 
    
14. Password - SIP Password
    
15. AuthID - SIP Username. 
    
    
5. Press Save in the top right hand corner. 
   
   
   
6. You will now be taken back into the VoIP Trunk list. Click Edit on the trunk you just created. 
   
   
   
7. Navigate to the Advanced Settings Tab. 
   
   
   
8. For codecs, remove all but PCMU, G729 and G722 from the Selected section
   
   
   
9. Enable Outbound Proxy Support and enter in the outbound proxy of sip.siplogin.ca
   
   
   
10. Set the maximum number of call lines to the number of channels the trunk is subscribed for. 
    
    
    
11. leave all other settings at their defaults and press save. 
    
    
12. Apply the changes, by pressing the apply changes button at the top of the screen. 
    
    
    

Confirm Successful Registration

1. From the left hand menu, navigate to System Status -> Dashboard. 
   
   
   
2. On this page, in the bottom right hand corner there is a section for Trunks. Locate your new trunk and confirm the status is showing as Blue. This indicates it successfully registered to our servers. 
   
   
   

Digital Trunk Setup

1. From the left hand menu, navigate to Trunks -> Digital Trunks
   
   
   
2. If you have an existing digital trunk, create a new one. Otherwise, edit the default Digital Trunk (Digital_1)
   
   
   
3. Set the following values. 
   
1. Trunk Name - Can be any desired name, or left at the default value
   
2. Port - Enabled
   
3. Hide CallerID - Your preference on this setting for how you want to handle caller ID. 
   
4. Keep Trunk CallerID - Your preference on this setting for how you want to handle caller ID. 
   
5. CallerID Number - Your Desired Caller ID number
   
6. CallerID Name - Your Desired Caller ID Name
   
7. DAHDI Out Line Selection - your preference on how calls will flow
   
8. Fax Gateway - Disabled
   
   
4. Press Save
   
   
   
5. Then apply the changes by pressing the apply changes button at the top of the screen. 
   
   

Additional settings can be found under Gateway Settings -> Interface Settings, which controls the analog connection. 

Interface Settings

1. On the GXW45XX Web UI, go to the PBX Settings menu and click on Interface Settings.
   
2. Ensure that the settings match the legacy PBX. Below is an example that matches a Shoretel System
   

Inbound Routes

Digital Trunk Inbound Routes

1. From the left hand menu, navigate to Trunks -> Inbound Routes
   
   
   
2. On the right hand side, from the "Trunks" dropdown, select the Digital Trunk that was created previously. 
   
   
   
3. Click the Add button
   
   
   
4. Enter the following patterns to allow for 10 and 11 digit inbound from the PRITrunk
   _XXXXXXXXXX and 1XXXXXXXXXX
   
   
   
5. Press save to save the changes. 
   

SIP Trunk Inbound Routes

1. Follow Steps 1 through 3 of the digital trunk inbound routes. 
   
   
2. This time, select the SIP Trunk. 
   
   
3. In the pattern options, enter in all DID's that are assigned to your account to allow for inbound calls to those DID's to reach the system. 
   
   If all DID's start with the same NPA or NPANXX, then a wildcard (X) can be used as shown in the example above. 
   
   
4. Press the save button to save the changes. 
   
   
   
5. Press Apply Changes to apply all changes. 
   
   

Outbound Call Routes

Outbound to PSTN

1. From the left hand menu, navigate to Trunks -> Outbound Routes. 
   
   
   
2. Select the Add button from the new window. 
   
   
   
3. Enter in the following details. 
   
   
   
1. Calling Rule Name - ToPSTN (but can be any unique identifying name you would like to use)
   
2. Pattern - use _XXXXXXXXXX and _1XXXXXXXXXX to allow for 10 and 11 digit dialling to route to the SIP Trunk
   
3. Trunk - Select the SIP Trunk that was setup earlier. 
   
   
4. Press save to save the changes. 
   
   
   

Outbound to PRI

This will direct calls from the SIP trunk connection to the PBX, essentially handling the inbound calls

1. In the same menu as the previous steps, click Add to add another outbound route. 
   
   
   
2. Enter the following details. 
   
1. Calling Rule Name - this is just a reference but is suggested to use "ToPRI" to identify the route. 
   
2. Pattern - enter in all DID's that are assigned to your account
   
3. Trunk - select the PRI trunk that was setup previously. 
   
   
3. Press the save button to save the changes. 
   
   
   
4. Press apply to apply all changes. 
   

You are now able to make inbound and outbound calls from the legacy PBX to and from the SIP Trunk connection. 

Please note, the inbound call routes while they will work for most setups, some additional adjusting may need to be done based on the clients unique setup.