Cisco 8800 Series Manual page 160

Telephony Features for Cisco IP Phone
Feature
Call History for Shared Line
Call Park
Call Pickup
Call Waiting
Caller ID
Caller ID Blocking
Calling Party Normalization
Cisco Unified Video Advantage (CUVA)
Conference
Configurable RTP/sRTP Port Range
Cisco IP Phone 8800 Series Multiplatform Phone Administration Guide for Release 11.3(1) and Later
140
Description and More Information
Allows you to view shared line activity in the phone Call History. This feature:
• Logs missed calls for a shared line.
• Logs all answered and placed calls for a shared line.
Allows users to park (temporarily store) a call and then retrieve the call by using another
phone.
Allows users to redirect a call that is ringing on another phone within their pickup group
to their phone.
You can configure an audio and visual alert for the primary line on the phone. This alert
notifies the users that a call is ringing in their pickup group.
Indicates (and allows users to answer) an incoming call that rings while on another call.
Incoming call information appears on the phone display.
Caller identification such as a phone number, name, or other descriptive text appear on
the phone display.
Allows a user to block their phone number or name from phones that have caller
identification enabled.
Calling party normalization presents phone calls to the user with a dialable phone number.
Any escape codes are added to the number so that the user can easily connect to the caller
again. The dialable number is saved in the call history and can be saved in the Personal
Address Book.
Allows users to make video calls by using a Cisco IP Phone, a personal computer, and
an external video camera.
Note Configure the Video Capabilities parameter in the Product Specific
Configuration Layout section in Phone Configuration.
See the Cisco Unified Video Advantage documentation.
Allows a user to talk simultaneously with multiple parties by calling each participant
individually.
Allows a noninitiator in a standard (ad hoc) conference to add or remove participants;
also allows any conference participant to join together two standard conferences on the
same line.
Be sure to inform your users whether these features are activated.
Note
Provides a configurable port range (2048 to 65535) for Real-Time Transport Protocol
(RTP) and secure Real-Time Transport Protocol (sRTP).
The default RTP and sRTP port range is 16384 to 16538.
You configure the RTP and sRTP port range in the SIP Profile.
Cisco IP Phone Configuration