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Register VoIP Phone to SIP Server

In this example we will register a Polycom VoIP interface from a Soundstructure to a SIP Call Server. Most market leading SIP platforms will work as long as they adhere to the SIP standards. There are a few key pieces of configuration to deal with and they are listed here:

SIP server IP address

This is the IP address of the SIP Call Server. Popular choices include Cisco Call Manager, Alcatel OmniPCX, Avaya IP Office, Microsoft Lync, Mitel, and Broadsoft.

Display Name

The display name used in SIP signaling that your phone uses as the default caller ID. This name is displayed on the call recipient’s phone.

Address

The line identification address that the phone uses to register with the server. The address may include a user name, the host of the phone’s SIP URI, or the H.323 ID/extension. For example, if the phone’s line is 1002@polycom.com, enter 1002 as the SIP where polycom.com is the server. Or, you can enter 1002@polycom.com. Any address entered will be displayed as the phone’s line if the display name and label are not specified.

Authentication User ID

The user name used to authenticate this line registration.

Authentication Password

The password used to authenticate this line registration.

Label

The text that will display next to the associated line key. If a label isn’t defined, the label will be derived from the user part of the address.

Set up the Call Server First!

In this example we’ll use Cisco Call Manager.

Phone Security Settings

To create the security profile that is used by the phone, follow these steps:
1. Open a Cisco Unified Communications Manager web administration session. You will prompted to enter your user name and password.
2. Select System > Security > Phone Security Profile.

Phone Security Profile

3. Click Add New.

Click Add New

4. Select Third-party SIP Device (Advanced) and click Next.

Third-party SIP Device

5. Enter the Phone Security Profile Information. In the Name text box, enter a profile name appropriate for the system, optionally enter a Description, and enable the Enable Digest Authentication check box.

Enable Digest Authentication

6. Click the Save button. In the status bar near the top of the page, “Update Successful” should appear as shown next.

Click the Save button

Add a Phone User

Once the phone security settings have been created, you need to create a user for the SoundStructure VoIP Interface as detailed in the following steps.
1. To create a user, select User Management > End User

Create A User

and click Add New as shown in the following figure.

Click Add New

2. In the User ID text box, enter a user ID according to system and account policies. In this example, the user name is set to sstvoipuser.

Create User ID

If you cannot add a user here, your system may be LDAP integrated, in which case you can use an existing user ID (essentially associating the phone to an existing user) or have your LDAP administrator create a new user ID for this phone.
3. In the Last Name text box, enter a last name. In this example, the value entered was LastName.
4. In the Digest Credentials text box, enter the digital credentials for the phone. Enter the same credentials in the Confirm Digest Credentials field. This password will be used with the User ID as the authentication password in the phone’s configuration file or when entering the line registration information with the Web Configuration Utility.

Configure Digest Credentials
5. Click the Save button. In the status bar near the top of the page, “Update Successful” appears.

Add a Device Entry

Once the user information has been entered, you may add the device information for the SoundStructure VoIP Interface.
1. Select Device > Phone and click Add New.

Add New Phone
2. Select Third-party SIP Device (Advanced) as shown below, and click Next.

Third Party Sip Device
3. Enter the Device Information as shown in the following screen and described next. Note that the data shown in this section is only an example.

Device Information
4. In the MAC Address text box, enter the MAC Address of the SoundStructure VoIP Interface. The MAC address is on the rear panel of the SoundStructure VoIP Interface as shown in the following figure and can also be found on the SoundStructure Studio Wiring page.

mac address
5. (Optional) In the Description text box, enter a description.
6. From the Device Pool list, select the device pool appropriate for your Cisco Unified Communications Manager system phones.
7. From the Phone Button Template list, select Third-party SIP Device (Advanced).
8. (Optional) From the Calling Search Space list, select an appropriate calling search space for the phone.
9. From the Location list, select an appropriate location for the phone.
10. From the Device Security Profile list, select the profile created in step 5 of Phone Security Settings.
The following screen appears. Note that the data shown in this section is only an example.

SIP Profile
11. In the SIP Profile field, enter the desired SIP profile.
12. In the Digest User field, select the user created in step 2 of Part II: Adding a Phone User. In this example, the user is sstvoipuser.
13. Click the Save button. In the status bar near the top of the page, “Update Successful” appears.
14. Click Apply Config. The resulting status message will appear. Click OK to continue.

Apply Config
15. In the Association Information area on the left side of the window, click on the Line [1] — Add a new DN link.

Add New DN
16. In the Directory Number text box, enter an appropriate phone extension. In this example, the extension 1234 is used.

Directory Number
17. From the Route Partition list, select an appropriate route partition.
18. In the Alerting Name text box, enter an appropriate alerting name. In this example, sstvoipuser is used.
19. In the ASCII Alerting Name text box, enter an appropriate ASCII alerting name. In this example, sstvoipuser is used.
20. Set the Voice Mail Profile according to the Cisco Unified Communications Manager system requirements. The default settings were used in the following figure.

Voice Mail Profile
21. Set the Call Forward and Call Pickup Settings to appropriate values for your system. In this example, no settings were changed.

Call Forward
22. In the Display (Internal Caller ID) text box, enter an appropriate caller ID. In this example, Conference Room was entered. This text will be shown on the recipient’s phone when a call is received from the SoundStructure VoIP Interface.

Display Caller ID
23. In the ASCII Display (Internal Caller ID) text box, enter an appropriate caller ID. In this example, Conference Room was entered.
24. In the Maximum Number of Calls text box, enter an appropriate value for your system. Currently the maximum supported value for the SoundStructure VoIP Interface is 24.

Maximum Number of Calls
25. In the Busy Trigger text box, enter an appropriate value for your system.
26. Click the Save button. In the status bar near the top of the page, “Update Successful” appears.
Congratulations, you have completed the configuration of Cisco Unified Communications Manager.  Get a cup of coffee, tea, or beverage of your choice, and get ready to move over to the client side configuration.

Set Up The SoundStructure VoIP Interface!

To configure the SoundStructure VoIP Interface, follow these steps:
1. Connect to your SoundStructure System by selecting Search for Devices from the Connect menu as shown in the following figure.

Search For Device
2. If the desired system was discovered, then select the system, highlight the Get configuration from devices option and click Connect as shown below. If your system is not present, you may either enter the IP address manually or use the Address Book tab to find systems that are stored in your network.

Connect To Device
3. Navigate to the Wiring Page by left clicking on the Wiring entry as shown in the following figure.

Wiring Page
4. Confirm the SoundStructure VoIP Interface is installed in this system by reviewing the information on the wiring page. You should see that there is a SoundStructure VoIP Interface listed with an ok status and a valid IP address.

Status OK
5. If the interface is found, then click on Web Configuration button to launch the Web Configuration Utility in your PC’s browser.

Web Configuration
6. Log into the Web Configuration Utility by selecting Admin and entering the default password of 456 and click Submit.
7. Navigate to the Settings > Lines Page.
8. Select the desired line to configure. By default this will be Line 1. The screen below will display.

Line 1
9. Enter the Display Name to be displayed on the dialing page. In this example, enter 1234 as that is the extension that was defined in step 16 of Add a Device Entry in Configuring Cisco Unified Communications Manager.
10. Enter the Address. In this example, enter 1234 to match the Display Name.
11. Enter the Authentication User ID. This is the User ID field that was entered as sstvoipuser in step 2 Add a Phone User in Configuring Cisco Unified Communications Manager.
12. Enter the Authentication Password. This is the Digest Credentials value that was entered in step 4 of Add a Phone User in Configuring Cisco Unified Communications Manager.
13. Enter the Label that will appear on the Phone Settings page within SoundStructure Studio. In this example, enter 1234. The resulting settings are shown in the following figure.

Label
14. Next, expand the Server 1 control and enter the IP address or hostname of the Cisco Unified Collaborations Manager. In this example the CUCM has an IP address of 172.24.144.119.

Sip Server Address
15. Click Save to apply the settings.
16. Navigate to the Channels page in SoundStructure Studio and open the Phone Settings control by clicking on the phone icon. If the line has been successfully registered, the phone icon will appear solid as shown in the following figure.

Fully Registered
If you made it this far, there is cause for Celebration. Now that your VoIP interface is successfully registered to the Call Server, the system will be able to make outgoing calls, receive incoming calls, and conduct useful meetings with the power of Polycom SoundStructure and Cisco Call Manager.

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