This article describes how to configure the
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such that it can be deployed downstream in a
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- MS Exchange 2007/2010 Unified Messaging Server topology.
Configuring the
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for this type of deployment requires specific configuration steps to be completed in the following order:
Topology |
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Configuration Overview
The configuration process comprises the following 10 steps, each of which is explained in detail in the body of this article.
- Default Tone Table: A factory default
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configuration contains the "Default Tone Table" entry. This article's configuration requires it, ensure its presence in your configuration. - Default Media Profile(s): the default G.711A/Mu-Law profile is used in this example.
- Default Media List: the Default Media List is used in this example.
- Create a SIP Server Table: In this example we add a SIP Server table entry for the Exchange 2007/2010 Server.
- Create a SIP Profile: you can create a custom SIP Profile or use the default. The default is used in this example.
- Create a Transformation Table: this table is used in the relate Call Routing tables and for Called/Calling Number transformations.
- Create a Call Routing Table: in this example we create two table entries, one for ISDN and another for SIP. They are required in the relevant Signaling Groups.
- Configure a DS1 Port: this port is associated with the Physical PRI (T1/E1) connection.
- Create two Signaling Groups: One Signaling Group will be designated for for SIP and the other is for ISDN.
- Set the System Timing: you can configure the Clock Source to use the
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system clock or a network source, depending on your PBX.
Configuring SBC Edge
SIP WebUI Login and verification of the Node Level Settings:
- In the WebUI, click on Settings.
- In the left navigation pane, go to System > Node-Level Settings.
Verify that the settings are configured correctly as indicated below.
SIP Server Table
- 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.
- 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.
- In the WebUI, click the Settings tab.
- In the left navigation pane, go to Transformation.
- Click the Create () icon.
Create a new Transformation Table.
Create Transformation Table
- In the left navigation pane, go to Transformation >(the table created in the previous step).
Add an entry as shown below.
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.
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.
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.
Create ISDN Signaling Group
- 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.
Create SIP Signaling Group
The Signaling Groups should appear like the following after adding the ISDN and SIP Signaling Groups:
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.
Call Route Entry for ISDN
- 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.
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.
System Timing Configuration
Exchange 20xx UM Server Configuration Steps
Exchange 20xx UM Server and Sonus SBC 2000 Gateway integration requires the following items to be configured on Exchange Management Console:
- A Dial Plan configuration with:
- Subscriber Access Number
- UM Mailbox Policy
- UM IP Gateway
- UM Auto Attendant
- UM Dial Plan association
Dial Plan Configuration
- Logon to Exchange / UM Server PC
- Launch Exchange Management Console
- Create a New Dial Plan using the following information:
Subscriber Access Number
Dialing Rule Group for 4 digit dialing used in this example:
Dialing Restriction
UM Mailbox Policy
UM IP Gateway
UM Auto Attendants
Associating Dial Plan to UM Server
SBC Edge Signaling Group Status Verification
After having completed both
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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:
Common Use Case
In a PBX –
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– 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:
- 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
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, and the Unable to show "metadata-from": No such page "_space_variables"
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