Manual Configuration

Manually adding new IP phones to the network is often tedious, but it can constitute a large part of day-to-day voice network management. Provisioning a Cisco SIP Phone is just like provisioning a SCCP phone.

The configuration procedure consists of these high-level steps:

Step 1 Add the IP phone. Step 2 Configure the phone. Step 3 Configure one or more DNs.

To manually add an IP phone to CUCM, go to Device > Phone and click the Add New button. Select the phone type (in Figure 9-24, a 7960 was selected). Select the device protocol that should be used with the Cisco IP Phone (SCCP or SIP), and click Next to get to the phone configuration page.

Figure 9-24 Manual Phone Configuration

Figure 9-24 Manual Phone Configuration

Each phone in the CUCM configuration database is uniquely identified by a device ID that is built from the phone's MAC address. The MAC address of a Cisco IP Phone is printed on a label at the back of the IP phone and can be viewed at the phone itself by pressing the Settings button.

In addition to the MAC address, the following mandatory parameters have to be set (as shown on Figure 9-25):

■ Phone button template

■ Device security profile

All of these mandatory parameters have to be configured. Any configuration item on the screen with an asterisk (*) next to it is a required field. Many of these fields have default values. The only exceptions are the preceding three bullet point items.

Figure 9-25 Manual Phone Configuration

Phone Configuration

Product Type: Cisco 7960 Device Protocol: SCCP

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Location* | Hub_None 3

Built In Bridge* | Default 3

Follow this procedure to configure the directory number for the manually added IP phone:

Step 1 After the Phone is Saved, the left side of the Phone Configuration window will have an Associated Information column. Click the Line [x] — Add a new DN link to configure the first directory number.

Step 2 When the Directory Number Configuration window appears (shown in Figure 9-26), enter the DN of the IP phone in the appropriate field.

Step 3 Click Save.

Use the same procedure to configure additional lines if the phone has more than one line.

Figure 9-26 Manual Phone Configuration

Figure 9-26 Manual Phone Configuration

Endpoint Registration Verification

Figure 9-27 shows an example of a phone listing from the Find and List Phones function available by clicking the Find button at the Device > Phone menu. Successful phone registration can be verified by checking the following items:

■ Look at the Status column and verify that the phone is registered.

NOTE If the phone is displayed as unregistered, the phone has previously registered but is no longer registered. If a phone is being reset, it is shown as unregistered during the short time until it reregisters with CUCM. If it is shown as unknown, it means that the phone has never successfully registered to the CUCM. If the phone is registered, its IP address is shown in the Status column. The IP address is used for CUCM to send signaling to.

■ Look at the IP Address column to verify whether the IP phone registered to the intended CUCM server.

By clicking the device name of a specific phone in the list, you can show the phone configuration page of the corresponding phone. The line configuration (DNs) parameters can be verified by clicking a line in the Phone Configuration page.

Figure 9-27 Phone Registration Verification

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Third-Party SIP Phone Configuration

The example used is based on a third-party SIP phone from Linksys. Step 4 will vary based on the vendor of the SIP phone, but the first three steps are the same for other third-party SIP phones.

The high-level steps for adding a third-party SIP phone are as follows:

Step 1 Configure the end user in CUCM.

Step 2 Configure the device in CUCM.

Step 3 Associate the device to the end user.

Step 4 Configure the third-party SIP phone to register with CUCM.

Figure 9-28 displays the addition of an end user in CUCM to be used for the third-party SIP phone. Third-party SIP phones register with CUCM by sending their username and password in a process called digest authentication. Third-party SIP phones do not send their MAC address like SCCP or Cisco SIP Phones. A MAC address must be configured for a third-party SIP phone, but it can be an arbitrary value because there is no MAC address verification. To add a new end user, navigate to User Management > End User and click the Add New button. The end user must be associated with the third-party SIP phone after the SIP phone is created.

Figure 9-28 End-User Configuration

Figure 9-29 displays the addition of a new third-party SIP phone. Navigate to Device > Phone and click the Add New button. Choose Third-Party SIP Device (Basic) or Third-Party SIP Device (Advanced) from the Phone Type drop-down menu. A Basic third-party SIP phone can have only one line and consumes three device license units (DLU). An Advanced third-party SIP phone can have up to eight lines and consumes six DLUs. Third-party SIP devices must have a SIP profile.

Figure 9-29 displays the addition of a new third-party SIP phone. Navigate to Device > Phone and click the Add New button. Choose Third-Party SIP Device (Basic) or Third-Party SIP Device (Advanced) from the Phone Type drop-down menu. A Basic third-party SIP phone can have only one line and consumes three device license units (DLU). An Advanced third-party SIP phone can have up to eight lines and consumes six DLUs. Third-party SIP devices must have a SIP profile.

Third-party SIP devices usually support RFC 2833 DTMF-Relay, which may cause compatibility issues with Type A Cisco IP Phones. Type A Cisco IP Phones require a media termination point (MTP) to communicate with a device that passes Dual-Tone Multifrequency digits in the RTP UDP channel (RFC 2833). MTPs are covered in detail in Chapter 15, "Media Resources."

The Digest User drop-down field must be populated with the end user configured in the preceding figure. After the phone has been configured with the digest user, update the enduser configuration by associating the end user with the third-party SIP phone.

Figure 9-29 Third-Party SIP Phone Configuration

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The final step to add a third-party phone takes place on the third-party phone itself. Therefore, the configuration depends on the product that is used. Figure 9-30 shows the configuration of a Linksys SPA 942 third-party SIP phone.

In the Proxy Address field of the third-party phone, specify the IP address or fully qualified domain name of CUCM.

The user ID has to be set to the DN that is assigned to the IP phone in CUCM. The auth ID has to match the digest user that was assigned to the phone in CUCM. The password needs to be set only if the digest credentials have been configured when configuring the end user and if the check box Enable Digest Authentication has been checked in the phone security profile.

NOTE If the Enable Digest Authentication check box has not been checked in the phone security profile, only the username of the digest authentication is verified; the password (digest credentials in CUCM end-user configuration) is not checked.

Some third-party SIP phones do not have a separate user ID and auth ID. In this case, the user ID has to be set to the DN at the third-party SIP phone, and on the CUCM side, the end-user name has to be identical with the DN of the IP phone. The shown Linksys phone simulates that behavior when the Use Auth ID parameter is set to No. Figure 9-30 is a configuration display from the Linksys SPA 942 product.

Figure 9-30 Third-Party SIP Phone Configuration

Figure 9-30 Third-Party SIP Phone Configuration

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