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- In the WebUI, click the Settings tab.
- In the left navigation pane, go to SIP > SIP Server Tables.
- Create a new IP Server Table. Click on the Create () icon.
The SIP Server Tables page is a table of SIP Server Tables.
Configure the new SIP Server Table as shown in the illustration below.
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- In the left navigation pane, go to SIP > SIP Server Tables > (The table you created in the previous step).
- Click the Create SIP Server > IP/FQDN.
- Enter the IP address of the eUM Server in the Host as shown below.
Click OK.
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Transformation Table
- In the WebUI, click the Settings tab.
- In the left navigation pane, go to Transformation.
- Click the Create () icon.
Create a new Transformation Table.
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- In the left navigation pane, go to Transformation >(the table created in the previous step).
Add an entry as shown below.
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Call Routing Table
- In the WebUI, click the Settings tab.
- In the left navigation pane, go to Call Routing Table.
Create two entries (Calls from ISDN, Calls to ISDN) as shown below.
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Port Configuration
- In the WebUI, click the Settings tab.
- In the left navigation pane, go to Node Interfaces > Ports.
- Click the port that you wish to connect to the physical PRI line.
Fill in the fields as shown below.
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Signaling Groups
- In the WebUI, click the Settings tab.
- In the left navigation pane, go to Signaling Groups.
Click Create Signaling Group and select ISDN Signaling Group text at the top of the table.
Fill in the fields as shown below.
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- In the WebUI, click the Settings tab.
- In the left navigation pane, go to Signaling Groups.
- Click Create Signaling Group and select SIP Signaling Group.
Fill in the fields as shown below.
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The Signaling Groups should appear like the following after adding the ISDN and SIP Signaling Groups:
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Call Routing Table Entries
- In the WebUI, click the Settings tab.
- In the left navigation pane, go to Call Routing Table > (call route entry for ISDN).
Fill in the fields as shown below.
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- In the WebUI, click the Settings tab.
- In the left navigation pane, go to Call Routing Table > (call route entry for SIP)
Fill in the fields as shown below.
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System Timing Configuration
- In the WebUI, click the Settings tab.
- In the left navigation pane, go to System > System Timing.
Configure the Clock Source either as System or the Network, depending on your PBX timing settings.
In this exercise, the Sonus SBC 1000/2000 node is reading the system time from the network therefore the configuration is done as shown below:
For more information about system timing see, Configuring the System Clock Source.
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Exchange 20xx UM Server Configuration Steps
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- Logon to Exchange / UM Server PC
- Launch Exchange Management Console
- Create a New Dial Plan using the following information:
Subscriber Access Number
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Dialing Rule Group for 4 digit dialing used in this example:
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Dialing Restriction
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UM Mailbox Policy
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UM IP Gateway
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UM Auto Attendants
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Associating Dial Plan to UM Server
SBC Edge Signaling Group Status Verification
After having completed both
and Exchange 20xx Unified Messaging Server configuration steps, using the
monitor view, ensure both PRI and SIP ports are up and in idle as shown below:
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Common Use Case
In a PBX –
– eUM Server scenario the following configuration steps allow PBX users to have UM voice mail boxes:
- On the PBX, a user must be configured with a diversion number as the Subscriber Access Number (i.e: 6255).
This number is configured in the Exchange Management Console >> Organization Configuration >> Unified Messaging >> UM Dial Plan >> Subscriber Access tab.
On the eUM, the relevant user entry must be enabled for Unified Messaging as shown below:
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- When a PBX user-1 receives a call (i.e., from PBX user-2) and does not answer, the call is diverted by the PBX to the diversion number (6225). The call arrives at the, and the sends an INVITE to the UM Server. Once the call reaches the UM Server, the Calling Party can leave a message in PBX user-1's voice mail box.
- In addition, PBX user(s) can call a Subscriber Access (SA) Number (6255) and an Auto Attendant (AA) Number (6256) via 4 digit dialing in above setup to navigate UM prompts via DTMF or voice commands
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