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IP Telephony:

Products & Applications


Update
October 17, 2001
Stergios Alexiou
Systems Engineer
salexiou@cisco.com
IPT_Update

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Agenda

Products Update
IPT Applications
Demo

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Products Update
Call Manager v3.1(x)
Unity v3.0(x)
Access Gateway: 4224
Survivable Remote Site Telephony
All-in-one box: ICS-7750
Gateways: VG200, VG248, DSP Farm
IP Phones: 7935, 7914
Analog Telephone Adaptor: ATA-186
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TDM PBX Architecture


Mainframe Cabinet
Voice Path (TDM)
Proprietary
Processor
Card(s)

Signaling Path

Proprietary
Interface
Applications
(Voice Mail/IVR)

Proprietary
TDM Switch
Card(s)
Proprietary
Interface
Proprietary
Line Card(s)

Proprietary
Trunk Card(s)

PSTN
Proprietary
Terminals

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Proprietary
Interface

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Standard Interface

Enterprise IP Telephony
Architecture
High Availability Server

Voice Path (TDM)

Call Processing

Signaling Path

Application
Standard
Processor

Standard
OR
Proprietary
Interface

Applications
Server
(Voice Mail /IVR)

IP
TCP/IP
Network

IP to PSTN
Gateway

PSTN

IP
IP phones
and
PC Applications

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Standard
Interface

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Standard Interface

Standard Interface

Call Manager v3.1(x)


new features
Music on Hold
Extension Mobility (log on to any 7960 or 7940)
Transcoder usage efficiency enhancements
Shared resource usage efficiency enhancements
Shared resource topological association
Wideband Audio (better than toll quality audio)
Serviceability enhancements
Gateway call preservation on CallManager failover
TAPI/JTAPI/XML enhancements
Configuration Database Interface API
Greek messages in phones (available in Dec 01)
Other CallManager 3.1 features
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File99

File-6

File-5

File-4

File-3

File-2

File-1

Music-On-Hold Service
Sound
Card

MOH App installed to MCS


Scalable, redundant with multiple
MCS

src-51

src-50

src-49

src-4

src-3

src-2

src-1

Audio sources:
50 cont. looping files
1 fixed source (optional)
MC or unicast per source

250 simultaneous streams per


server
IP phones

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SoftPhones Gateways

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G.711, G.729a, and wideband


audio per source

Extension Mobility
Central Site

Users office
IP Phone 7960
(x5000)

IP LAN
IP Phone
Services
Web Server

IP WAN

LDAP
Directory

Log onto any 7940/7960 in a


CM cluster to get your
extension and productivity
services
Extension, services, class of
service restrictions applied to
phone
Login modes
auto-logout other phones
keep login on other phones

Logout modes
7960

Visiting user
logged onto
phone (x5000)

explicit logout at phone


timed logout

Branch Office
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Cisco MNET -AVVID Integration

Mobile office NETwork(MNET)Enterprise-based pico-cellular


GSM network

Interface to MCS network through SS7 protocol gateway (SC2200)

AVVID integration project


CallManager endpoint capabilitiesadd GSM-EFR and GSM-FR
Add GSM-EFR and GSM-FR codecs to transcoder DSPs in:
Catalyst 6000 Voice Services DSP card
Mileslow-cost, stand-alone DSP services device

GSM
Mobility
Controller (GMC)

H.323

Cisco CallManager

GSM

Transcoder

MCS
GSM
Port (GP)

GSM

V
SC2200

PSTN

SS7

IP
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Shared Resource Operation Using


Default Group
MOH1

CM1

MTP1
SW-CONF1

MOH2

CM2

MTP2
SW-CONF2

NM-1
NM-2

V
XCODE1

HW-CONF1

XCODE2

V
HW-CONF2

After configuration all media resources belong to the


default group.
All resources are accessible by CM1 and CM2.
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Wideband Audio - IP phones

16-bit linear, 16 kHz sampling (256 kbps BW)


Extends audio range to >5 kHz
7910, 7940, 7960 IP phones
Handsets only

Music-on-hold service
No support in
7935, 12SP+, 30VIP
Gateways (TDM side bandwidth passes signal anyway)

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Serviceability Enhancements
SNMP agents in devices or device proxies
Real-time statistics, alerts
Access by SNMP manager/SNMP app
Broadened MIB support in devices/apps

http server/xml parser in devices/device proxies


Real-time statistics, alerts
Access by xml over http
Device monitoring app
http viewer

Related Project - CiscoWorks 2000 Enhancements


Voice Health Monitor (VHM) version 1.0
User Tracking (UT) enhancements

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Call Preservation to VoIP


Gateways

Preserve calls to/from VoIP gateways in event of CallManager failure

Requires MGCP signaling protocol from CallManager to gateway

Scope
Cat 6K - E1, T1 voice services cards
Cat 6K - 24 port FXS
Cat 4K - Access Gateway Module (AGM)
VG200 - MGCP support for additional TDM interfaces
DT-24+, DE-30+
(26XX, 36XX push TBD)

Out-of-scope
DPA 7610/DPA-7630 (skinny client versus skinny gateway)
Active Voice Unity (skinny client)
AT-2, 4, 8; AS-2, 4, 8 (skinny gateway protocols do not support call preservation)

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API Enhancements
TAPI/JTAPI
CallManager redundancy through CTI Manager(s)
This allows redundant CallManager to ICM connection

Phone XML (7940 and 7960 only)


Idle URL
Media Streaming
This allows continuously running advertisements

Database Layer API


Read/write access to CallManager config database and
embedded directory objects
Devices, lines, users
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CTI Layer Redundancy


CM Cluster
CM Server
CTI
Manager

CM Server
CCM

Application
(JTAPI/TAPI)

CTI
Manager
CCM

Application Platform

CM Server
CTI
Manager
CCM

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No Single Point of Failure

CM failure - CTIManager will manage fail-over


to another CM in cluster.

CTIManager failure application


(JTAPI/TAPI) will be able to reconnect to
another CTIManager.

Application failure - CTIManager will route


calls to a configured forward on failure DN
(configured per device).

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Additional version 3.1


Enhancements
Survivable Remote Site
Telephony support
Call forward number
expansion

Overlap sending
Support for new Hardware
Catalyst 4224 (Jobim)

Call forward reason codes

TAPI/JTAPI - zip tone and


auto-answer

RDNIS - Re-directed Dialed


Number Identification
Service

DT-30+, Cat 6K E1 TDM


protocol support

VG200 TDM protocol support

T1-CAS

E1/T1-PRI
T1-CAS

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Products Update
Call Manager v3.1(x)

Unity v3.0(x)
Access Gateway: 4224
Survivable Remote Site Telephony
All-in-one box: ICS-7750
Gateways: VG200, VG248, DSP Farm
IP Phones: 7935, 7914
Analog Telephone Adaptor: ATA-186
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Cisco Unity 3.0(x) Delivers


Windows 2000 mixed/native mode support. Cisco Unity
3.0(1) will not support Windows NT on the Cisco Unity
server, although it can be installed into an NT
environment
Increased number of text-to-speech sessions per server
(actual number TBD pending QA testing and
qualification)
Up to 48 ports per server
Up to 5,000 subscribers per server

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Cisco Unity 3.0(x) Delivers contd

Up to 100,000 users in Global Address List


Support Cisco CallManager v3.1(x) and maintenance
releases
Support for AMIS (tested specifically against the Octel 250,
350, 100, and Unified Messenger)
Pre-MTA queue to take messages and provide basic
message access while Exchange is off-line
MCS 7825 support with USB key
Additional Cisco branding in GUI

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Cisco Unity Unified Messaging


Delivers
True Unified Messaging
Automated attendant
Browser-based personal and
system administration interfaces
Convergence-ready communications
Advanced features, including
text-to-speech and mobile message
notification

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Comparison: Cisco Unity Voice Mail &


Unified Messaging
Voice Mail

Unified Messaging

Message Access

Message Access

Voice/Fax from TUI


Notification via SMTP

Address Book
Voice

Administration
Two directories

Installation
As separate NT
domain server

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E-mail/Voice/Fax from MS Outlook


& TUI
Notification via SMTP

Address Book
Global Address List

Administration
One directory

Installation
As part of the
Windows/Exchange network
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Comparison: Cisco Unity Voice Mail &


Unified Messaging
Voice Mail

Unified
Messaging

Remote System Admin.


ActiveAssistant

Yes
Yes

Yes
Yes

Third-party Fax Integ.


Unified Messaging
Text-to-Speech
Digital Networking
Automated Attendant

No
No
No
Yes
Yes

Yes
ViewMail for Outlook
Up to 6 sessions
Yes
Yes

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Cisco Unity
ViewMail for Outlook

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Cisco Unity
Outlook Web Access

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Cisco Unity
ActiveAssistant

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Cisco Unity Administration

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Cisco Unity Administration

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Products Update
Call Manager v3.1(x)
Unity v3.0(x)

Access Gateway: 4224


Survivable Remote Site Telephony
All-in-one box: ICS-7750
Gateways: VG200, VG248, DSP Farm
IP Phones: 7935, 7914
Analog Telephone Adaptor: ATA-186
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Catalyst 4224 - What is it?


It is a Micro Branch single box solution targeted at Global
2000 accounts
4224 provides the following services to the Micro Branch:
LAN Switch (with Inline Power for IP Phones)
WAN Router (IOS Based IP Routing only)
Voice Gateway (Analog & Digital IOS Based)
Survivable Remote IP Telephony (SRST)

Catalyst 4224 is an integral part of a Multi-site WAN with


Centralized Call Processing Solution

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Catalyst 4224 - What Does It Look


Like?
24 Ports 10/100
(Support Inline Power)

2U
VWIC Slots

VIC Slot
Switch
Utilization
LEDs

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Console

Management
Ethernet

8 Port FXS
Module
(RJ21)

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Catalyst 4224 Solutions


A

Router/GW
V

Single Box Micro Branch Solution


Centralized or local CallManager
Up to 24 users
ISDN Dial Backup capable
Single E1 branch access ( Savings)
SRST backup in IOS

Cisco CallManager Cluster

Headquarters

Catalyst 4224

IP WAN
VIC/WIC

PSTN
IPT_Update

ISDN BU

MFT-E1

12 Channels Voice /
12 Channels Data
Single E1 Access to
each branch office
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8 x FXS
24 10/100 Ports

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Centralized Call Processing


(Popular Deployment Model)
Cat 4224
with SRST

CallManager Cluster

IP WAN
Branch 1

Cat 4224
with SRST
Unity
Server

V
Application
Server(s)

PSTN
Branch 2

Application
Server(s)

ISDN
Backup for
IP WAN

Cisco 2600
with SRST

V
Central Site
Branch (n)

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Products Update
Call Manager v3.1(x)
Unity v3.0(x)
Access Gateway: 4224

Survivable Remote Site Telephony


All-in-one box: ICS-7750
Gateways: VG200, VG248, DSP Farm
IP Phones: 7935, 7914
Analog Telephone Adaptor: ATA-186
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What Is SRS Telephony?


Survivable Remote Site Telephony
Unique, industry-first capability in branch office routers
for IP Telephony redundancy
Cost-effective enterprise wide deployment of IP telephony enabled by
Cisco AVVID (Architecture for Voice, Video and Integrated Data)
Always available branch IP telephony
Ideal for enterprise customers utilizing Centralized Cisco CallManager
deployment for IP telephony at their branch offices

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Survivable Remote Site


Telephony How it works
IP Phones exchange Keep alive messages and Call Processing
messages with Centrally located CallManager (CCM)
WAN Link fails IP phones lose contact with CCM
IP Phones register with local router as router of last resort
Router queries pones for configuration and auto-configures itself
Router provides call processing for duration of failure via PSTN
Upon restoration of WAN, IP Phones revert back to CCM
Applications
Applications
Server
Server
CallManager
Cluster

WAN

Cisco 7200

PSTN

Cisco 2600 Router


with SRS Telephony

Headquarters
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Cisco IOS Platforms Supported


Available now:
Cisco IAD 24xx
Cisco 26xx: 261x, 262x, 265x
Cisco 36xx: 3620, 3640, 3660
Cisco Catalyst 4224

Future availability 12/01:


Cisco Catalyst 4000 AGM
Cisco 1750-V/1751-V, Cisco 7200

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PricingSurvivable Remote
Site Telephony
Cisco IOS:
Plus image set required on Cisco 2600, 3600
Standard Cisco IOS image on Catalyst 4224

Additionally:
Feature license required to use SRS Telephony
Platform

# of Phones
Supported

Part
Number

Spare Part
Number

License

IAD24xx, 26xx,
3620, Catalyst 4224

Up to 24
phones

FL-SRSTSMALL

FL-SRST-SMALL=

$750

3640, 3660

Up to 48
phones

FL-SRSTMEDIUM

FL-SRSTMEDIUM=

$1250

3660

Up to 144
phones

Up to 3 FLSRST-MEDIUM
licenses

Up to 3 FL-SRSTMEDIUM= licenses

$1250 per
license

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Products Update
Call Manager v3.1(x)
Unity v3.0(x)
Access Gateway: 4224
Survivable Remote Site Telephony

All-in-one box: ICS-7750


Gateways: VG200, VG248, DSP Farm
IP Phones: 7935, 7914
Analog Telephone Adaptor: ATA-186
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ICS 7750 - Delivering AVVID converged


infrastructure and applications
Designed for Enterprise Branch
Offices and Midmarket Businesses
(<200 users)
Integrated communications platform
for voice, data and application
services
Web-based System Management
simplifies deployment and
management
Designed for expandability

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Integrated Communication
System 7750
Cisco Catalyst Switch

.
..

Modular platform delivers future-ready


solution

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Call Manager

Adding users add IP phones and Catalyst switch


ports
Adding voice gateway or WAN capacity
Connect to existing router, add router blades or add
an external router
Adding analog ports add more analog interfaces
and/or router blades
Adding New World voice/data applications
add application blades and/or connect to external
server(s)

Application Blade

Application Blade

Scales cost-effectively to meet


changing requirements

Router/Voice Gateway

Chassis-based system with modular blades


Six Universal Slots support CallManager,
Trunking/Routing, and Application (SPE) blades
Hot-swappable, Hot-standby system components

Router/Voice Gateway
Spare

Fan Tray

100Mb
Links

Power
Supply

Power
Supply
(2nd
Opt)

External
Application
Server
(Optional)

Cisco Integrated Communication


System (ICS) 7750

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ICS 7750 Application Summary


Unity Voice Mail and Unified Messaging
On external server: Available Today
Integrated in chassis (on SPE blade): Jan 02

Cisco AVVID Voice Applications


IVR/ICD/AA/WebAttendant: Available Today
Softphone: Available Today
FXO/FXS/WAN integration: Available Today
Call Manager v3.1(x): Jan 02
IPCC integration: Dec 01
ISDN support: Dec 01

AVVID Partner Applications


MindCTI, ISI (Call Detail Reporting apps): Available Today
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AVVID Partner Applications


Available today
Call Detail Reporting: MindCTI, ISI

Available FY2001/02
Call Acctg. & Billing: SwitchView, Veramark
E-conferencing: Hear Me
Call Centers: Apropos, EasyRun
Overhead Paging: Valcom, Bogen
Personal Productivity: Tygart, Webley
Unified Messaging: AVT, Baypoint
Low cost phones: Circa, NEC
Wireless Phones: Symbol
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Products Update
Call Manager v3.1(x)
Unity v3.0(x)
Access Gateway: 4224
Survivable Remote Site Telephony
All-in-one box: ICS-7750

Gateways: VG200, VG248, DSP Farm


IP Phones: 7935, 7914
Analog Telephone Adaptor: ATA-186
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Overview
Cisco VG200-VoIP Gateway
Proven modular architecture
Uses Cisco network
modules and VIC ports
Cisco IOS software-based
Worldwide homologation
Very high reliability (MTBF)
Broad interface type support
Multiple protocols supported
Remote manageability

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Overview (Cont.)
Cisco VG200-VoIP Gateway
Field upgradeable-switch between analog and digital ports
Up to four voice interface card (VIC) ports of Foreign Exchange
Station (FXS), Foreign Exchange Office (FXO),
E&M, or Basic Rate Interface (BRI)
Mix and match port types
PBX trunking, fax, modems,
analog phones
Basic Rate ISDN (BRI)-user
and network side
Caller ID on FXS and FXO
Analog direct inward dial (DID)

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Overview (Cont.)
Cisco VG200-VoIP Gateway
Up to two digital interfaces, T1 or E1
Maximum of 60 voice channels (E1)
T1 CAS-FGD
E1 CAS-R2
T1 and E1 PRI support
(user and network side)

Voice codec support


G.711-64kbps
G.729 (A,B,AB)-8kbps
G.723.1(A)-6.3/5.3kbps

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New Features

IOS 12.1(5)YH
DSP Farm Support
Transcoding up to 60 Sessions G.711, G.729a, GSM-FR and
GSM-EFR
Conferencing up to 90 participants

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DSP Farm Capacity Sizing

Each DSP SIMM contains 3 DSPs


Each DSP supports either 4 transcoding
sessions or 1 six party conference call.
DSP resources are defined using Call Manager.
Multiple VG200s can be used to provide greater
than 15 conference call capacity or greater than
60 transcoding sessions capacity.

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New Application

Teal Transcoding for GSM calls to IP


Phone
Magenta Voicemail connectivity for
remote station using transcoding to
conserve WAN resources.
Orange Branch office conferencing
without utilizing WAN resources

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New Modules

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Description

Part Number

DSP Farm Network Module 2 SIMMs

NM-HDV-FARM-C36(=)

DSP Farm Network Module 3 SIMMs

NM-HDV-FARM-C54(=)

DSN Farm Network Module 5 SIMMs

NM-HDV-FARM-C90(=)

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Cisco VG248 Analog Phone Gateway


The VG248 is a high density gateway for using analog
phones, fax machines, and modems with a Cisco
CallManager system

The VG248 provides


48 port FXS connectivity
1u 19 stackable unit
2 RJ-21 telco connectors
Cisco CallManager 3.1 support
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Features and Benefits

Analog phones Needed when the infrastructure (wiring) or application does not
support or require IP telephones. Full featured analog phone lines, allow organizations to
deploy IP Telephony without having to purchase IP telephones for all the users.

Fax & Modem fax machines and modems are required in many locations. Fax &
Modem support allows organizations to support these legacy devices alongside the new
world IP telephones.

Voicemail many legacy voicemail systems require analog and SMDI connectivity.
Voicemail support allows for legacy voicemail systems to be supported in more configurations
and with higher reliability that was previously possible.

Investment protection -Customers have the ability to re-use phones, legacy


analog devices while taking advantage of IP Telephony.

Cost effective Entry - With a low cost alternative for low end phones organizations
can take advantage of IP Telephony with a lower overall investme nt.

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Where the VG248 fits


PSTN

Gateway

VG248
CallManager
Cluster

Analog

IP Phones

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IP phones and
analog phones
can co-exist on
the same Cisco
CallManager
system

Legacy Analog
Phones

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VG248 Analog Phone Features

Caller ID
Transfer

Caller ID on
Call Waiting

Conference

Group PickUp

Message Waiting
Indicator

Call Forward

Stuttered Dialtone

Speed Dial

Redial

Call Waiting

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VG248 Fax & Modem Features

Cisco FAX Relay


Modem Pass Through

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SMDI for Legacy Voicemail


Legacy
Voice Mail

Analog

VG248

SMDI

Multiple SMDI links per


Cisco CallManager
SMDI fail-over capability

Cisco
CallManager

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Voicemail location
independence

57

SMDI Multiplexing
Legacy
Voice Mail

Analog

Multiple SMDI
sources
An
alo
g

SMDI

VG248
SMDI

Multiple
Clusters

Cisco
CallManager
Gateway

Legacy PBX

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Smooth
Migration

Single
voicemail
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Analog Voicemail Scalability


Legacy
Voice Mail

Analog

SMDI

Single SMDI
port into
voicemail

VG248
SMDI
passed
between
VG248s

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SMDI passed
over IP

Cisco
CallManager

59

Products Update
Call Manager v3.1(x)
Unity v3.0(x)
Access Gateway: 4224
Survivable Remote Site Telephony
All-in-one box: ICS-7750
Gateways: VG200, VG248, DSP Farm

IP Phones: 7935, 7914


Analog Telephone Adaptor: ATA-186
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Cisco IP Phone Family

Cisco 7960+7914
Cisco 7960

Cisco SoftPhone

Cisco 7935

Cisco 7940
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7910 and 7910+SW


61

Products Update
Call Manager v3.1(x)
Unity v3.0(x)
Access Gateway: 4224
Survivable Remote Site Telephony
All-in-one box: ICS-7750
Gateways: VG200, VG248, DSP Farm
IP Phones: 7935, 7914

Analog Telephone Adaptor: ATA-186


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Cisco ATA 186 Benefits & Features


Cisco ATA 186
Analog Telephone Adaptor

Chassis dimensions (H x W x D)
6.5 x 6 x 1.5 in.

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Interfaces traditional
phones to IP-based
networks

RJ-45 connection to
10BaseT Ethernet
Supports two voice ports
(RJ-11 connections)

Standard protocols and


voice codecs for
interoperability and
deployment flexibility

H.323 and SIP


G.729A (only one port at a time),
G.723.1, G.711A, G.711
TCP/UDP/IP

Clear, natural-sounding
voice

VAD/CNG
Dynamic jitter buffer
Line echo cancellation

Flexible configuration
and provisioning
options

Automated or manual
configuration and control

63

Cisco ATA 186 Updates

New features found in Version 2.10


Auto-provisioning of profiles via TFTP servers
Supplementary services (three-way calling, call waiting, call
hold/retrieve, caller ID)
G.711 fax pass-through
Fast start, alternate gatekeeper, admission level security with MD5
hashing in H.323
Authentication and firewall support in SIP

Upcoming features
MGCP 1.0/NCS Q4 CY 01 release
SCCP Q4 CY 01 release

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Agenda

Products Update

IPT Applications
Demo

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IPT Applications

Web Attendant
Interactive Voice Response
Integrated Contact Distribution
Personal Assistant
Unity
IP Contact Center

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What Is Cisco WebAttendant?

Cost-Effective Tool for Enterprise Attendants


and Receptionists to Answer and Greet
Callers, then Efficiently Dispatch Calls
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Cisco IP IVR

Cisco IP IVR is an IP-powered Interactive


Voice Response solution which offers an
open, extensible, feature-rich foundation for
the creation and delivery of IVR solutions via
Internet technology.

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Cisco IP ICD
Call Manager 3.[01](x)
Application
Engine 2.[12] Software

Cisco IP Integrated Contact Distribution (ICD) provides an entrylevel contact interaction management solution that is fully
integrated with current and future Cisco Customer Response
Application (CRA) platform solutions (e.g. IP IVR)

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Cisco Personal Assistant 1.2


Enhanced Productivity
Route calls by user-defined rules
Time of day
Calendar
Who the caller is

Forward calls to user-defined phone numbers


Single number
Series of numbers (i.e., office, mobile, or home)
Follow me

NEW!
NEW!

Screen calls
Select which calls to accept in real time
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Cisco Unity Communications


Solutions

Cisco Unity Unified Messaging

Full Unified Messaging (UM)


Works w/Cisco CallManager
Compatible with MS Exchange
Will be compatible with Lotus Notes

Cisco Unity Voice Mail

IPT_Update

IP Voice Messaging
Upgradeable to Unity UM
Works w/Cisco CallManager

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71

The IP Contact Center

Intelligent
Contact
Management

CallManager

IPT_Update

2000, Cisco Systems, Inc.

Agent
Desktop

Integrated
Voice
Response

72

Agenda

Products Update
IPT Applications

Demo

IPT_Update

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73

Demo: Centralized Call Processing


Call Manager

PSTN

Unity

Exchange 5.5

C
A

2610 SRST
V

Cat 3524 PWR

V
Cat 3524 PWR

3640

IP WAN

4000

Branch B
8300

Central Site A

Call Manager
2105

Branch C
IPT_Update

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74

Presentations & Documents

This presentation and all relevant


technical documents and data sheets are
available under this link:
http://www.cisco.com/hellas/events/

IPT_Update

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75

IPT_Update

1999,
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76

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