Cisco 7821 Administration Manual page 116

For cisco unified communications manager 10.0 (sip)
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Telephony Features for Cisco IP Phone
Feature
HTTPS for Phone Services
Hunt Group
Improve Caller Name and Number
Display
Incoming Call Toast Timer
Intercom
Jitter Buffer
Join Across Lines
Cisco IP Phone 7821, 7841, and 7861 Administration Guide for Cisco Unified Communications Manager 10.0
(SIP)
102
Description and more information
Increases security by requiring communication using HTTPS.
Note
IP Phones can be HTTPS clients; they cannot be HTTPS
servers.
See
Set Up HTTPS for Phone Services, on page
Provides load sharing for calls to a main directory number. A hunt group contains a series of
directory numbers that can answer the incoming calls. When the first directory number in the
hunt group is busy, the system hunts in a predetermined sequence for the next available
directory number in the group and directs the call to that phone.
See:
• Cisco Communications Manager Administration Guide, "Hunt Group Configuration"
• Cisco Unified Communications Manager System Guide, "Understanding Route Plans"
Improves the display of caller names and numbers. If the Caller Name is unknown then the
Caller Number is displayed instead of Unknown.
Allows you to set the length of time that an incoming call toast (notification) appears on the
phone screen.
See
Set Up Incoming Call Toast Timer, on page
Allows users to place and receive intercom calls using programmable phone buttons. You
can configure intercom line buttons to:
• Directly dial a specific intercom extension.
• Initiate an intercom call and then prompt the user to enter a valid intercom number.
If your user logs into the same phone on a daily basis using their Cisco Extension
Note
Mobility profile, assign the phone button template that contains intercom information
to their profile, and assign the phone as the default intercom device for the intercom
line.
See the Cisco Unified Communications Manager Feature and Services Guide, "Intercom"
chapter.
The Jitter Buffer feature handles jitter from 10 milliseconds (ms) to 1000 ms for both audio
and video streams.
Allows users to combine calls that are on multiple phone lines to create a conference call.
Some JTAPI/TAPI applications are not compatible with the Join and Direct Transfer feature
implementation on the Cisco IP Phone and you may need to configure the Join and Direct
Transfer Policy to disable join and direct transfer on the same line or possibly across lines.
See
Set Up Join and Direct Transfer Policy, on page
135.
125.
135.

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