Cisco SMART BUSINESS COMMUNICATIONS SYSTEM - FEATURE REFERENCE GUIDE 12-2010 Feature Reference page 24

Schedules for
holidays and
business
hours
Secondary dial
tone
Secure Socket
Layer (SSL)
Phone Client
Sequential
ephone hunt
groups
Session
Initiation
Protocol (SIP)
SIP IP Phone
dial plans
SIP IP phones
SIP trunk
SIP
supplementary
services,
disabling
Shared lines
Shared-line
overlay
ephone-dns
Simple Auto
Attendant
script
Single number
reach (SNR)
Softkeys
Software-
based
conferencing
© 2010 Cisco and/or its affiliates. All rights reserved. This document is Cisco Public Information.
Voice-system The Auto Attendant on Cisco Unified Communications 500 offers holiday and business schedules to enable time-
features of-the-day routing of incoming calls.
Voice-system A secondary dial tone is available for Cisco Unified IP Phones connected to Cisco Unified Communications 500 in
features the PBX mode. The secondary dial tone is generated when you dial a predefined PSTN access prefix and
terminates when you dial additional digits.
Voice-system The SSL Phone Client on the SPA525G IP phone provides secure connectivity to a UC500 or SR500 series over
features the internet. The SSL Phone Client at the remote site does not need a teleworker router, making it an ideal
solution for simple scenarios that require voice-only connectivity.
Voice-system The sequencing method of hunting always starts with the first member of the hunt group and hunts through all the
features members in the sequential order.
Voice-system SIP is a signaling protocol widely used for controlling multimedia communication sessions such as voice and
features video calls over IP. Cisco Unified Communications 500 supports Cisco SIP endpoint devices as well as SIP trunks
to SIP providers.
Voice-system SIP dial plans enable call routing using the SIP protocol. The dial plans route calls toward a SIP server, which
features could be either another Cisco Unified Communications 500 at a remote site or a SIP server hosted by a SIP
provider.
A dial plan is a set of dial patterns that SIP phones use to determine when digit collection is complete after you go
off-hook and dial a destination number. Dial plans allow SIP phones to perform local digit collection and recognize
dial patterns as your input is collected. After a pattern is recognized, the SIP phone sends an INVITE message to
the Cisco Unified Communications 500 to initiate the call to the number matching your input. All of the digits you
entered are presented as a block to the Cisco Unified Communications 500 for processing. Because digit
collection is done by the phone, dial plans reduce signaling messages overhead compared to Keypad Markup
Language (KPML) digit collection.
SIP dial plans eliminate the need for you to press the Dial softkey or # key, or to wait for the interdigit timeout to
trigger an outgoing INVITE. You can configure a SIP dial plan and associate the dial plan with a SIP phone. The
dial plan is downloaded to the phone in the configuration file.
Users, IP phones can communicate using several protocols, including H.323, SIP, and Media Gateway Control Protocol
phones, and (MGCP). SIP IP phones use the SIP protocol to communicate with a SIP server.
extensions
Voice-system SIP trunks provide an alternative to the traditional PSTN (digital and analog) connectivity options. SIP trunks are
features provisioned through SIP service providers that provide PSTN connectivity.
Voice-system The SIP call transfer and call forwarding supplementary services feature introduces the ability of SIP gateways to
features initiate blind, or attended, call transfers by passing the call control back to the originating devices. Disabling these
supplementary services forces the SIP gateway to handle these call conditions locally.
Users, A shared directory number allows the same number to appear on two different IP phones. A call made to a shared
phones, and directory number rings all the IP phones that have a button assigned to the shared number. A call made from the
extensions shared directory number ties up the shared-dn-buttons on the rest of the IP phones. If a call on the shared
directory number is put on hold, any of the IP phones can resume the on-hold call.
Users, The overlay feature allows a single IP phone button to be associated with multiple directory numbers. A call to any
phones, and of the associated directory numbers rings the IP phone on the overlaid button. Shared directory numbers can also
extensions be part of the overlay configuration. Primary extensions cannot be part of the shared overlay ephone-dns.
Voice-system The basic Auto Attendant is included in the Cisco Unified Communications 500 as aa_simple.aef script. It
features supports dial-by-extension, alternate, holiday, and business-hours greetings.
Voice-system SNR allows you to have incoming calls to a single number simultaneously ring an IP phone and a remote
features destination such as a home or mobile phone. You can answer incoming calls to your SNR number on your IP
phone or at your remote destination and pick up in-progress calls on your desktop phone or the remote
destination without losing the connection.
If you do not answer the call within 5 seconds, the Cisco Unified Communications 500 system rings the remote
number while continuing to ring your IP phone extension. If you answer the call on your IP phone, you can send
the call to the remote phone by pressing the Mobility softkey.
If you answer the call on your remote phone, you can pull back the call to the IP phone by pressing the Resume
softkey. You can also change the SNR remote destination using the IP Phone menu.
Users, Softkeys are keys that appear on the bottom of the IP phone LCD. They allow you to access various features such
phones, and as call forward, call transfer, conferencing, and call park. The softkeys available for use change dynamically
extensions according to whether the phone is in connected, ringing, idle, or seized (handset is lifted). Additional softkeys are
also automatically enabled when advanced features, such as SNR and live record, are enabled.
Voice-system Software conferencing is the type of voice conferencing that is supported by default on the Cisco Unified
features Communications 500. It does not require any hardware resources; the audio mixing is done within the application
software. Software conferencing allows a maximum of three parties in a conference, with maximum of eight
simultaneous conferences.
Feature Description Guide
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