Cisco Ip Phone 8800 Series Multiplatform Phones Administration Guide - Cisco 8851 Administration Manual

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
Conference
Configurable RTP/sRTP Port Range
Directed Call Pickup
Divert

Cisco IP Phone 8800 Series Multiplatform Phones Administration Guide

156
Description and More Information
Allows you to view shared line activity in the phone Call History. This feature will:
• Log missed calls for a shared line
• Log 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 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
Note
activated.
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.
Allows a user to pick up a ringing call on a DN directly by pressing the GPickUp softkey
and entering the directory number of the device that is ringing.
Allows a user to transfer a ringing, connected, or held call directly to a voice-messaging
system. When a call is diverted, the line becomes available to make or receive new calls.