cisco SBCS Webpage

SBCS Webpage

http://www.cisco.com/web/go/sbcs/index.html

Cisco Selling Edge Program (2 hrs) – Qualifying the customer, Identifying appropriate products & positioning the solution

http://cisco.partnerelearning.com/Saba/Web/Main/goto/searchOffering?title=The Selling E.D.G.E. Program – Q1/2008 Cisco SBCS Solution

Smart Business Roadmap –Documentation designed to be used with a customer to help discover the major business challenges (Cost containment, operation efficiency, security, customer responsiveness) and then link this challenge to a standard Cisco solution.

http://www.cisco.com/web/partners/sell/smb/tools_and_resources/smart_business_roadmap.html#~tab-1

SMB Community Sales Support – Portal that describes resource to assist you in selling SBCS

https://www.myciscocommunity.com/docs/DOC-1451

SBCS Smart Designs (4 hrs) – Design & Implementation Guide for SBCS deployments

http://www.cisco.com/web/partners/sell/smb/tools_and_resources/smart_business_comm_system.html

Self paced UC500 technical enablement labs (CCA)

https://www.myciscocommunity.com/docs/DOC-10395

SBCS Demo Box – Order a demo box using Cisco Not For Resale Program (NFR)

http://www.cisco.com/web/ANZ/partners/promotions/nfr/index.html

Cisco Voice Fundamentals (6 hr)

http://cisco.partnerelearning.com/Saba/Web/Main/goto/searchOffering?title=Cisco Voice Over IP Fundamentals (CVF)

Resources

Cisco IP Telephony Express Training (1 Day)

http://cisco.partnerelearning.com/Saba/Web/Main/goto/searchOffering?title=IP Telephony Express (IPTX) 4.0

Cisco UC500 Best Practices Sessions (1.5 hr)

https://www.myciscocommunity.com/docs/DOC-14451

Cisco Press Book – Cisco IP Communications Express: Callmanager Express with Cisco Unity Express.

http://www.ciscopress.com/bookstore/product.asp?isbn=158705180x

Cisco Support Community (Including VoD’s, Forums and How-to’s)

https://www.myciscocommunity.com/community/smallbizsupport

Cisco Support Community – Design & Deployment Support

https://www.myciscocommunity.com/docs/DOC-1377

Resources

Learning Map for Cisco Unified Communication Manager Express

http://cisco.partnerelearning.com/Saba/Web/Main/goto/searchCurricula?title=Express Unified Communications – Engineer

Cisco IronPort Training & Certification

http://www.ironport.com/products/

Cisco IronPort Training & Certification

E-Learning – What’s New in AsyncOS 7.2 for Security Management (Web)

E-Learning – Activating Cisco IronPort Web Usage Controls

E-Learning – AnyConnect Secure Mobility

E-Learning – Application Visibility and Control

E-Learning – Cisco IronPort Hosted and Hybrid Email Security

E-Learning – What’s New in AsyncOS 6.7 Security Management (Email)

E-Learning – What’s New in AsyncOS 6.7.3 for Security Management (Web)

E-Learning – What’s New in AsyncOS 7.1 for Email

E-Learning – Cisco IronPort Hosted Security Services

E-Learning – Introduction to Email Messaging

E-Learning – Introduction to HTTP/HTTPS

E-Learning – Centralized Configuration Manager

E-Learning – Centralized Reporting and Tracking

E-Learning – Domain Keys Identified Mail (DKIM)

E-Learning – Email Encryption

E-Learning – Essential Layers of Web Security

E-Learning – Image Analysis for Email

E-Learning – Reporting with Sawmill

E-Learning – Safelists/Blocklists

E-Learning – Sawmill for IronPort 7.3

E-Learning – Sender Policy Framework (SPF)

E-Learning – Smart Identifiers & Dictionary Weighting

E-Learning – The Advantages of Next Generation Hardware for Web Security

E-Learning – Using Content Filters to Prevent Data Loss

E-Learning – Web Reputation

Upgrading IOS Procedure tutor

Trainings

Operation of Cisco IOS (e-Lab Simulation)

Preparing for Boot System Commands e-Lab

Managing the Startup Configuration File (e-Lab Simulation)

IOS 15.x: The New IOS Licensing Model

Cisco IOS Software Upgrade Procedure (In general)

Cisco IOS Software Upgrade Procedure for Cisco 801, 802, 803, 804, and 805 Series Routers

A Wrong system software for this hardware error message is displayed after upgrading the Cisco IOS Software image

User cannot get into ROMmon mode to download a Cisco IOS Software image into the router

How to choose and install software for the 801, 802, 803, 804, 805, 806, 826, 827, 828, 831, 836, SOHO70, 1000, 1400, 1600, 1600-R, 1700, 2500, 2600, 3000, 3600, 3700, 4000, 4500, 4700, AS5100, AS5200, AS5300 and MC3810 products

Per Call Bandwidth Consumption

Per Call Bandwidth Consumption

CUCM Locations-Based CAC Bandwidth Values
You can find this information in CUCM Admin > System > Locations > Add Location > Help > About This Page

• G.711 call uses 80 kbps
• G.722 call uses 80 kbps
• G.723 call uses 24 kbps
• G.728 call uses 16 kbps
• G.729 call uses 24 kbps
• GSM call uses 29 kbps
• Wideband call uses 272 kbps

Gatekeeper-Based CAC Bandwidth Values
A “debug h225 asn1” will reveal “bandwidth 1280” in the ARQ. Note well, H225 debugs tack a 0 (zero) on the end of your values. Therefore, 128 kbps (G.711) will show 1280 and 16 kbps (G.729) will show 160.

• G.711 call uses 128 kbps
• G.729 call uses 16 kbps

QoS Bandwidth Values
Best document is the QoS SRND, page 33.
[L2 (G.729 = 20 kbps; G.711 = 80 kbps) + L3] * 8 * 50 = Value per call

• 802.1Q Ethernet adds (up to) 32 bytes of Layer 2 overhead.
• Point-to-point protocol (PPP) adds 12 bytes of Layer 2 overhead.
• Multilink PPP (MLP) adds 13 bytes of Layer 2 overhead.
• Frame Relay adds 4 bytes of Layer 2 overhead; Frame Relay with FRF.12 adds 8 bytes.

RSVP Bandwidth Values
A “debug ip rsvp messages” will show the worst-case scenario bandwidth value used. Look for “start requesting XX kbps” in the output.

The ip rsvp bandwidth value should be equal to (x-1 calls at normal scenario + 1 call at worst case scenario). For example, two G.729 calls would be calculated as 24 + 40 = 64 kbps. In the example below, I have configured RSVP for one G.729 calls (1-1 calls at normal scenario + 1 call at worst-case scenario of 40 kbps)

• G.711 call uses 96 kbps as a 10 ms worst-case scenario.
• G.729 call uses 40 kbps as a 10 ms worst-case scenario.

Voice Translation Rules in Media Gateways

Voice Translation Rules in Media Gateways

Voice Translation Rule is a key part in the Voice Gateways which helps you to translate the Calling Number, Called Number and the

Plan type customized for your customers / partners while sending or recieving calls in the gateway.

This document helps you to create the Voice Translation Rules for configuring the Calling and Called Dial patterns and plans specific for

National and International dialing plans. There are examples given for changing the Number type and plan and Rejecting calls.

Example 1

This example replaces the first occurrence of the number “123″ with “456″.

voice translation−rule 1

rule 1 /123/ /456/

These are test voice translation−rule examples:

router#test voice translation−rule 1 123

Matched with rule 1

Original number: 123 Translated number: 456

router#test voice translation−rule 1 1234

Matched with rule 1

Original number: 1234 Translated number: 4564

router#test voice translation−rule 1 6123

Matched with rule 1

Original number: 6123 Translated number: 6456

router#test voice translation−rule 1 6123123

Matched with rule 1

Original number: 6123123 Translated number: 6456123

Original number type: none Translated number type: none

Original number plan: none Translated number plan: none

In this example, the rule matches the first occurrence of the number that contains the pattern “123″ anywhere

in the number. Specifically, you can use the start and end of number indicators. The Example 2 and Example

3 sections show this.

Example 2

This example shows how to replace any occurrence of “123″ at the start of a number with “456″.

voice translation−rule 1

rule 1 /^123/ /456/

These are test voice translation−rule examples.

router#test voice translation−rule 1 123

Matched with rule 1

Original number: 123 Translated number: 456

router#test voice translation−rule 1 1234

Matched with rule 1

Original number: 1234 Translated number: 4564

router#test voice translation−rule 1 6123

6123 Didn’t match with any of rules

Example 3

If you want only the match of an exact number, specify both the start and end number indicators:

voice translation−rule 1

rule 1 /^123$/ /456/

router#test voice translation−rule 1 123

Matched with rule 1

Original number: 123 Translated number: 456

router#test voice translation−rule 1 1234

1234 Didn’t match with any of rules

router#test voice translation−rule 1 6123

6123 Didn’t match with any of rules

Example 4

This example replaces all numbers with “5554000″.

voice translation−rule 2

rule 1 /.*/ /5554000/

router#test voice translation−rule 2 123

Matched with rule 1

Original number: 123 Translated number: 5554000

router#test voice translation−rule 2 86573

Matched with rule 1

Original number: 86573 Translated number: 5554000

router#test voice translation−rule 2 “”

Matched with rule 1

Original number: Translated number: 5554000

Example 5

This example replaces any number that starts with a combination of zeros (0, 00, and so forth) with “909″.

voice translation−rule 5

rule 1 /^0+/ /909/

router#test voice translation−rule 5 0123456

Matched with rule 1

Original number: 0123456 Translated number: 909123456

router#test voice translation−rule 5 00123456

Matched with rule 1

Original number: 00123456 Translated number: 909123456

NUMBER TYPE AND PLAN TRANSLATION

Example 1

In this example, if a number starts with “4″ and the type is “national”, the rule adds “90″ as a prefix. If the

type is “international”, the rule adds “900″ as the prefix.

voice translation−rule 7

rule 1 /^4/ /904/ type national national

rule 2 /^4/ /9004/ type international international

router#test voice translation−rule 7 493456567 type national

Matched with rule 1

Original number: 493456567 Translated number: 90493456567

Original number type: national Translated number type: national

Original number plan: none Translated number plan: none

router#test voice translation−rule 7 493456567 type international

Matched with rule 2

Original number: 493456567 Translated number: 900493456567

Original number type: international Translated number type: international

Original number plan: none Translated number plan: none

This is useful when telephone companies (Telcos) remove access codes on national and international

numbers. You can add the correct prefix with the number type as a basis.

Example 2

This example changes the number type and plan.

voice translation−rule 8

rule 1 /^2\(…$\)/ /01779345\1/ type unknown national plan unknown isdn

This rule matches any four−digit number that starts with “2″. The rule removes the “2″, adds the number

“01779345″ as a prefix, and sets the plan to “isdn” and the type to “national”.

router#test voice translation−rule 8 2001 type unknown plan unknown

Matched with rule 1

Original number: 2001 Translated number: 01779345001

Original number type: unknown Translated number type: national

Original number plan: unknown Translated number plan: isdn

Reject Calls

Use the reject keyword to reject calls that match. This example rejects all calls that start with “234″.

rule 1 reject /^234/

router#test voice translation−rule 10 1234

1234 Didn’t match with any of rules

router#test voice translation−rule 10 2345

blocked on rule 1

VoIP Monitoring tools

VoIP Monitoring

ManageEngine VQManager Download ManageEngine VQManager

ManageEngine VQManager is web-based, VoIP Quality Monitoring software that can monitor any VoIP equipment that supports SIP, H.323, Cisco Skinny and RTP/RTCP. It enables IT administrators to monitor their VoIP network for QoS parameters, bandwidth utilization & Call traffic trends.

· Real-time monitoring of VoIP QoS metrics & MOS scores

· Proactive operator alerts, Email notifications, SNMP traps

· Call flow ladders, raw packet views for troubleshooting

· User-configured call history reports for capacity planning

NetQoS VoIP Monitor

NetQoS VoIP Monitor is a network-based call setup and call quality monitoring product that tracks the call quality users experience, provides alerts on call performance problems, and isolates performance issues to speed troubleshooting and MTTR. NetQoS VoIP Monitor is integrated with the NetQoS Performance Center so you can monitor VoIP quality of experience while managing network quality of service, from a single, Web-based console.

SolarWinds VoIP Monitor

VoIP Monitor allows you to proactively analyze VoIP quality across WAN links, as well as monitor the underlying systems and protocols that the VoIP environment relies upon, providing complete integration with Orion. VoIP Monitor’s simulation-based approach with IP SLA alerts you to problems and enables you to fix them with its robust capabilities, including:

Collecting and analyzing VoIP performance statistics, including MOS, jitter, network latency, packet loss, and other important QoS metrics

Facilitating capacity planning for existing multi-vendor VoIP deployments and measuring voice quality in advance of new VoIP deployments

Automatically configuring IP SLA on Cisco routers without you having to lift a finger

NimBUS

With NimBUS solutions, organizations can monitor and manage every service and system within their entire Cisco VoIP (Voice over IP) ecosystem:

· VoIP Networks

· VoIP Call Activity

· VoIP Messaging

Smarts VoIP Performance Manager

EMC Smarts VoIP Performance Manager delivers the performance data you need to ensure the highest possible call quality and reliability. With VoIP Performance Manager, organizations can manage, monitor, and diagnose Voice over IP (VoIP) services.
VoIP Performance Manager provides intelligent alerting, deep diagnostics, and extensive reporting to help you gain in-depth, real-time views into the performance of VoIP services and the telephony infrastructure on which they rely while showing how that detailed information relates to the end user experience.

VQmon/EP

VQmon/EP detects packet loss and jitter buffer discard events, extracts key information from DSP software and produces call quality scores and diagnostic data. VQmon/EP has been integrated with products from Audiocodes, Global IP Sound and Texas Instruments. Other leading DSP software vendors work closely with Telchemy to ensure that integration of VQmon with CODEC and Jitter Buffer software is seamless.

Free training Voice/Unified Communications part 2

CTT-TAC: Introduction to Basic Analog Voice over IP Learn how to use emerging voice internetworking technology with Introduction to Basic Analog Voice over IP. CCVP Unified Communications VoIP Lab

CTT-TAC: Analog Voice Internetworking with E&M Signaling Learn how to use emerging voice internetworking technology with Analog Voice Internetworking with E & M Signaling. CCVP Unified Communications E&M Lab

CTT-TAC: Basic Analog-to-Digital Voice over IP Learn how to use emerging voice internetworking technology with Basic Analog-to-Digital Voice over IP. CCVP Unified Communications Analog VoIP Lab

CTT-TAC: Link Efficiency Mechanisms Learn how to use emerging voice internetworking technology with Link Efficiency Mechanisms. CCVP Unified Communications Queuing Lab

CTT-TAC: Queuing Techniques Learn how to use emerging voice internetworking technology with Queuing Techniques. CCVP Unified Communications Queuing Lab

Unified Communications Architecture

Unified Communications Systems Release Approach
Defines the objectives and benefits of a System release approach (10 min).

Unified Communications Architecture Focus Areas
Discusses the general and product specific Focus Areas within the architecture (10 min).

Unified Communications Installation and Upgrade

Deploying DHCP Server on Unified CallManager 5.0
Configure DHCP services on Unified CallManager 5.0 (20 min).

Unified CallManager 5.0 Directory Integration
Describes the characteristics of Unified Communications CallManager 5.0 LDAP Directory with that of other CallManager releases (25 min).

Deploying AutoQoS on Wan Routers
Configure Cisco AutoQoS Enterprise feature on WAN Edge Routers (15 min).

Unified Communications Deployment Methods
Discusses the recommended steps in deploying Unified Communication Systems (20 min).

Unified Communications IP Telephony Architecture
Defines the purpose for IP Telephony Deployment Models (20 min).

Unified IPCC 7.0 Enterprise Deployment Models
Discusses each deployment module under IPCC Enterprise 7.0 (20 min).

Unified IPCC Express 4.5 Deployment Models
Discusses the single server deployment models under IPCC Express 4.5 (15 min).

Unified Communications IP Telephony Deployment Models
Identifies 7 deployment models and their components (20 min).

Unified Communications QoS for Security
Describes the steps outlined by Cisco on how to use QoS as a security tool (20 min).

Unified Communications RSVP Overview
Discuss the concepts, terms and functionality of RSVP (20 min).

Unified Communications Troubleshooting

Unified Communications Systems Approach to Troubleshooting
Discusses the approach to troubleshooting Unified Communications Systems (20 min)

free Trainings for cisco R&S

CTT-TAC: Link Efficiency Mechanisms à Start Here

Developing an Optimum Design for Layer 2

http://www.cisco.com/web/learning/le31/le46/cln/qlm/CCDP/arch/layer-2-design-2/player.html

Describing & Implementing Spanning Tree Tutorial:

http://www.cisco.com/E-Learning/bulk/public/ccnp/QLM_STP/player.html

Understanding VLANs and Trunks

http://www.cisco.com/web/learning/le31/le46/cln/qlm/CCNA/icnd2/understanding-vlans-and-trunks/player.html

Introducing Access Control List Operation

http://www.cisco.com/web/learning/le31/le46/cln/qlm/CCNA/icnd2/introducing-access-control-list-operation/player.html

Introducing the OSPF Protocol

http://www.cisco.com/web/learning/le31/le46/cln/qlm/CCNP/bsci/Introducing-OSPF-AUG07/player.html

Introducing EIGRP Protocol

http://www.cisco.com/E-Learning/bulk/public/ccnp/EIGRP_QLM/player.html

Introducing BGP Considerations

http://www.cisco.com/web/learning/le31/le46/cln/qlm/CCIP/bgp/introducing-bgp-confederations-2/player.html

Introducing Traffic Policing and Shaping

https://www.cisco.com/web/learning/le31/le46/cln/qlm/CCNP/ont/Introducing-Traffic-Policing-and-Shaping/player.html

Explaining WLAN Technology and Standards

http://www.cisco.com/web/learning/le31/le46/cln/qlm/CCNA_Wireless/iuwne/Emerging-WLAN-Technology-Standards-BCMSN/player.html

Class-Based Weighted Fair Queuing (CBWFQ) and Low Latency Queuing (LLQ)

https://www.cisco.com/web/learning/le31/le46/cln/qlm/CCNP/ont/CBWFQ-and-LLQ_2/player.html

Congestion Management Configuring CBWFQ and LLQ

http://www.cisco.com/web/learning/le31/le46/cln/qlm/CCVP/qos/congestion-management-configuring-cbwfq-and-llq-2/player.html

Introducing VPNs

http://www.cisco.com/E-Learning/bulk/public/cln/qlm/ccip/mpls/MPLS-QLM-Intro-to-VPNs/player.html

Configuring Small-Scale Routing Protocols Between PE and CE Routers

http://www.cisco.com/web/learning/le31/le46/cln/qlm/CCIP/mpls/configuring-small-scale-routing-protocols-2/player.html

Understanding MPLS Traffic Engineering Concepts

http://www.cisco.com/web/learning/le31/le46/cln/qlm/CCIP/mpls/understanding-mpls-te-concepts/player.html

AnyConnect for Samsung Android has Arrived!

Cisco AnyConnect Secure Mobility Client is the first 3^rd party (and only SSL) VPN client available for Samsung Android devices.

Customers may download the Cisco AnyConnect Secure Mobility Client directly from the Android Market.

Supported Devices:

Galaxy S model GT-I9000 (Gingerbread Maintenance Release)

Galaxy S model SC-02B (Gingerbread Maintenance Release)

Galaxy S II model GT-I9100

Galaxy S II model SC-02C

AnyConnect is also supported on Tab 7 running Android 2.3.3+ or Galaxy Tabs 8.9 and 10.1 running Android 3.0+.

Software Access:

https://market.android.com/details?id=com.cisco.anyconnect.vpn.android

Users Guide:

http://www.cisco.com/en/US/docs/security/vpn_client/anyconnect/anyconnect24/android-user/guide/android-acug.html

Release Notes:

http://www.cisco.com/en/US/docs/security/vpn_client/anyconnect/anyconnect24/release/notes/rn-ac2.4-android.html

Licensing and Infrastructure Requirements:

AnyConnect for Android requires Cisco Adaptive Security Appliance (ASA) Boot image 8.0(4) or later.

For licensing questions and evaluation licenses, please contact ac-mobile-license-request (AT) cisco.com and include a copy of "show version" from your Cisco ASA.

If you already have an Essentials or Premium ASA license, you may use the automated license request tool at:

https://tools.cisco.com/SWIFT/Licensing/PrivateRegistrationServlet?FormId=717

The ASA requires an AnyConnect Mobile license (L-ASA-AC-M-55XX=), as well as either an AnyConnect Essentials (L-ASA-AC-E-55XX=) or AnyConnect Premium Clientless SSL VPN Edition (L-ASA-AC-SSL-YYYY=) license, where XX is the last two digits of your ASA model number and YYYY is the number of simultaneous users. AnyConnect Mobile and Essentials licenses are enabled per ASA, there is no per user charge for either of these licenses.

H.225 RAS Signaling: Gatekeepers and Gateways

RAS Gatekeeper Discovery

Thisis the processes by which H.323 terminals/gateways discover their zone gatekeepers Automatic Gatekeeper Discovery:

• If an H.323 endpoint does not know its gatekeeper, then it can send a Gatekeeper Request (GRQ). This is a UDP datagram addressed to the well-known destination port 1718 and transmitted in the form of an IP multicast with the multicast group address 224.0.1.41.
• One or several gatekeepers can answer the request with either a positive Gatekeeper Confirmation (GCF) message or a negative Gatekeeper Reject (GRJ) message. A reject message contains the reason for the rejection and can optionally return information about alternative gatekeepers. Auto discovery enables an endpoint to discover its gatekeeper through a multicast Gatekeeper Request (GRQ) message. Because endpoints do not have to be statically configured for gatekeepers, this method has less administrative overhead. A gatekeeper replies with a GCF or GRJ message. A gatekeeper can be configured to respond only to certain subnets.
  Note: A Cisco IOS gatekeeper always replies to a GRQ with a GCF/GRJ message. It never remains silent.

If a gatekeeper is not available, the gateway periodically attempts to rediscover a gatekeeper. If a gateway discovers the gatekeeper has gone off-line, it ceases to accept new calls and attempts to rediscover a gatekeeper. Active calls are not affected.

This table defines the RAS gatekeeper discovery messages:

Gatekeeper Discovery
GRQ (Gatekeeper_Request)	A message sent by endpoint to gatekeeper.
GCF (Gatekeeper_Confirm)	A reply from gatekeeper to endpoint which indicates the transport address of the gatekeeper RAS channel.
GRJ (Gatekeeper_Reject)	A reply from gatekeeper to endpoint that rejects the endpoint's request for registration. Usually due to gateway or gatekeeper configuration error.

RAS Registration and Unregistration

Registration is the process by which gateways, terminals, and/or MCUs join a zone and inform the gatekeeper of their IP and alias addresses. Registration occurs after the discovery process. Every gateway can register with only one active gatekeeper. There is only one active gatekeeper per zone.

The H.323 gateway registers with an H.323 ID (email ID) or an E.164 address. For example:

• EmailID (H.323 ID): gwy-01@domain.com
• E.164 Address: 5125551212

This table defines the RAS gatekeeper registration and unregistration messages:

Gatekeeper Discovery
RRQ (Registration_Request)	Sent from an endpoint to a gatekeeper RAS channel address.
RCF (Registration_Confirm)	A reply from the gatekeeper that confirms endpoint registration.
RRJ (Registration_Reject)	A reply from the gatekeeper that rejects endpoint registration.
URQ (Unregister_Request)	Sent from endpoint or gatekeeper to cancel registration.
UCF (Unregister_Confirm)	Sent from endpoint or gatekeeper to confirm an unregistration.
URJ (Unregister_Reject)	Indicates that the endpoint was not preregistered with the gatekeeper.

Interdigit Timeout (T302)

This is a small explanation about Interdigit Timeout.
Interdigit timeout is called T302 within CUCM.

To change the interdigit timeout value go to
System => Service Parameters
Select the CUCM server and the “Cisco CallManager” Service.
Once there do search with “CTRL-F” to open up the find window in your  browser.
Then search for “302″
This will bring you down to the “T302 Timer” settings field.
Enter the desired value in milliseconds and then click “Save”.
When you are done restart the CCM service.

Navigate to Cisco Unified Serviceabillity
Tools => Control Center – Feature Services
Now select the server then click the radio button for Cisco CallManager  and click on “Restart”.

From CUCM in regards to the T302 timer:

T302 Timer :
This parameter specifies an interdigit timer for sending the SETUP ACK  message. The timer restarts each time Cisco CallManager receives a  digit.
When this timer expires, CUCM routes the dialed digits. For exact timer  definitions, refer to the Q.931 specification.
This is a required field.
Default: 15000
Minimum: 3000
Maximum: 75000

All Units are in msec.

New Features for Cisco Unity Connection Version 8.5

Features and Benefits

New Features for Cisco Unity Connection Version 8.5

• Unified messaging with Microsoft Exchange 2010: • Voice messages are synchronized with the Exchange inbox. • MWI and message status are synchronized. • Secure, private messages, mobile client, and calendar integration for Exchange 2010 are all supported. • You can enable unified messaging for specific users or all users.

• The solution offers a new Web 2.0 Inbox Client that is based on HTML v5.0. • The Cisco Unity Connection Web Inbox uses Representational State Transfer-based application programming interfaces (APIs) for functions, it is backward-compatible with HTML v4.0, and it can be deployed as a widget or gadget.

• Virtualization and platform enhancements: • The solution offers a new 1000-user virtualization overlay. • Support for Cisco UCS B200 M2, UCS C210 M2, UCS C200 M2, and the Cisco 7825-I5 Media Convergence Server is introduced with Cisco Unity Connection 8.5 and later.

• Application and database audit logging allows you to track configuration changes to the Cisco Unity Connection system in separate audit log files. • This feature reports configuration changes for Cisco Unity Connection Administration, Cisco Unity Connection Serviceability, Cisco Unified Serviceability, Real-Time Monitoring Tool (RTMT), Cisco Personal Communications Assistant, the command-line interface (CLI), user authentication events for Cisco Unity Connection clients that use the Representational State Transfer APIs, and API calls for clients that use the Cisco Unity Connection Provisioning Interface. • The database logging feature reports changes to the Cisco Unity Connection database.

• IPv6 support: • Cisco Unity Connection 8.5 supports IPv6 addressing with Cisco Unified Communications Manager (7.1(2) or later) phone system integrations using Skinny Client Control Protocol (SCCP) and Session Initiation Protocol (SIP). • The addressing mode is configurable by port group.

• Messaging enhancements: • Messaging files are shredded for secure deletion. • Message recording expiration guarantees voice messages cannot be listened to after they reach a set expiration date. • Message aging alerts are supported.

• Conversation enhancements: • The Cisco Unity Connection system can announce before playing a message when it has been sent to multiple recipients. • You can customize message conversations to allow you to listen to all the recipients of the message. • You now can toggle between touch-tone and speech-recognition conversations.

• API enhancements: • The Cisco Unity Connection Provisioning Interface has been expanded to include access for individual users. • The Cisco Unity Connection Messaging Interface has been expanded to allow for access to secure messages.

• Bulk Administration Tool enhancements: • With this tool you can create and update multiple alternate first and last names for contacts and users. • You can create and update multiple alternate names for Distribution List objects and passwords. • You now can update personal identification numbers (PINs). • You can create and update standard and closed transfer rule settings for users.

• Run Now functions have been added to task management.

• Call-routing rules now allow for routing to different conversation styles.

• The solution supports Cisco Unified SIP Proxy integration.

• The solution offers new support for Lightweight Directory Access Protocol (LDAP) integrations (AD 2008 and AD-LDS).

• The limit for VPIM locations has been increased to 50.

• The solution supports more languages (Catalan, Chinese-Hong Kong, and Norwegian).

• A new ViewMail for Outlook plug-in provides unified messaging support for Outlook 2007 and 2010 and IMAP support for Outlook 2003.

Message Access from the TUI

• You can play and process messages (repeat, reply, forward, delete, save, mark as new, hear day or time stamp, or skip to the next message).

• You can reverse, pause, or fast forward messages during playback.

• You can control volume and speed during message playback.

• You can pause or resume during message recording.

• You can address messages to multiple recipients.

• With the message locator, you can search for messages by caller ID, name, or extension in saved messages.

• You can record messages and mark them as regular, urgent, private, or secure.

• You can record messages and request a return receipt.

• You can record a live conversation with a caller and have the recording sent to your mailbox.

• You can switch between spelling name and extension when addressing a message.

• With live reply, you can immediately reply to messages from other users.

• You can access email messages over the phone using the text-to-speech (TTS) feature (for Microsoft Exchange 2003, 2007, and 2010).

• When TTS is enabled, a conversation tells you if the message has attachments; when an attachment is in a playable or readable format, the attachment is played or read.

• You can view, listen, respond to, and play back messages using the Cisco Unified Communications Widget for Visual Voicemail on Cisco Unified IP Phones. Learn more about Visual Voicemail at:http://www.cisco.com/en/US/prod/collateral/voicesw/ps6882/ps9156/at_a_glance_c45-487475.pdf.

• You can access Microsoft Exchange calendar through speech or the TUI. • You can browse the calendar and accept, decline, or cancel an Outlook appointment.

• If you inadvertently disconnect while sending a new message, replying to, or forwarding a message, and if the message has at least one recipient or a recording, Cisco Unity Connection can save the message as a draft and allow you to return to finish the message on a subsequent call.

• You can review and recall messages sent over a period of time.

• When you hang up or your call is disconnected, bookmarks allow you to call back into Cisco Unity Connection and resume listening to messages without losing your place.

Speech-Enabled Messaging*

• Speech Connect for Cisco Unity Connection, a speech-enabled Automated Attendant for the enterprise, allows you to connect quickly with your colleagues using only your voice (available with Cisco Unity Connection v7.1.3 and later).

• You can speak your voicemail password.

• You can speak dates and times.

• You can use speech commands to play and process messages (play, record, reply, forward, delete, save, etc.).

• You can use speech commands to edit and manage your personal greetings.

• You can use speech commands to address messages to private distribution lists.

• You can use speech commands such as pause, resume, speed up, slow down, skip ahead, and skip back to provide rich and granular control of messages and prompts.

• Speech-enabled directory handlers allow outside callers to use voice commands to reach Cisco Unity Connection users.

• You can temporarily use touch tones to change setup options, and then return to speech-recognition mode.

• A speech command tutorial is available.

• You can customize speech-enabled directory handler greetings.

* Speech-enabled messaging is available for U.S. English only.

Call-Transfer Rules

• You can define rules to route incoming calls by caller.

• You can define rules to route incoming calls by time of day.

• You can define rules to route incoming calls by your calendar free or busy status (Microsoft Exchange only).

End-User Features

• If a call is dropped while you are recording a message, Cisco Unity Connection saves a draft message and you can continue recording where you left off during your next session.

• You can customize message-notification options, manage personal greetings, or change passwords with Cisco Unity Connection Assistant (the Cisco web browser-based personal administrator).

• You can select the conversation type: Full or brief prompts.

• You can record and then address a message, or address and then record a message.

• You can record a message for future delivery.

• You can record up to five personal greetings (alternative, busy, internal, off hours, or standard).

• You can manage an alternative greeting, set the expiration date or time, notify users when an alternative greeting is set, or require callers to listen to the full alternative greeting.

• You can forward calls directly to an alternative greeting (or other personal greeting) without ringing the phone.

• You can specify an after-greeting action; after a user greeting, callers can leave a message, sign in, or hang up, or they can be sent to call handlers, directory handlers, interview handlers, or other users.

• You can use flex stack to specify the order in which messages are presented over the phone: by urgency and then by last in, first out (LIFO) or first in, first out (FIFO).

• You can create private distribution lists and address messages to them through the TUI or GUI.

• You can provide message notification for new messages through devices such as Simple Mail Transfer Protocol (SMTP), Short Message Service (SMS), text pagers, and phone destinations.

• With a cascade message-notification feature, you can send additional notification types if a message is not retrieved.

• You can send notifications for messages from a particular user or phone number.

• You can select whether message counts are announced; totals, saved, and new counts are available.

• You can specify whether Cisco Unity Connection announces a transferred call.

• You can perform a supervised transfer for individual alternate contact numbers.

• You can view and play back messages using Visual Voicemail on Cisco Unified IP Phones. You can use soft keys on Cisco Unified IP Phones to access all messages, new messages, or messages from a specific subscriber or outside caller.

• You can use a Really Simple Syndication (RSS) reader to retrieve voice messages.

• You can perform a “live reply” to someone who left a message from an external telephone.

• With ViewMail for Microsoft Outlook (VMO) and ViewMail for IBM Lotus Notes (VMN) plug-ins, you can compose, reply to, forward, play, rewind, or pause messages directly from within the Outlook or Notes email client.

• You can compose, reply to, and forward messages by using IMAP clients.

• Through calendar integration with Cisco Unified MeetingPlace® 7.0, you can join a meeting that is in progress, hear a list of participants for a meeting, send a message to the meeting organizer or participants, and set up an immediate meeting.

• You can dispatch a message to a group, with the message being assigned to the first member of the group to listen to the message. When the message is assigned, it is deleted from all other users’ inboxes and becomes a normal message in the assignee’s mailbox.

• You have flexibility with support for partitions, search spaces, and search scopes.

• You can receive and forward fax messages through integration with the Cisco Fax Server.

• You can customize subject lines for messages received in any visual client that displays the subject message, such as an IMAP or RSS client.

• You can use a single phone number for both voice calls and fax transmissions.

• With the Voice Message Store and Forward feature, administrators, on a per-user basis, can forward voice messages to an external mailbox, making it easier for you to access voice messages on a mobile device.

System Administration Overview

• Cisco Unity Connection supports digital networking for up to 100,000 users within an enterprise and up to 20 servers or active-active cluster server pairs, including cross-server login, cross-server transfer, and cross-server live replay.

• Cisco Unity Connection is scalable to 250 ports and 20,000 users per server. Refer to the Cisco Unity Connection Supported Platforms List for details at: http://www.cisco.com/en/US/products/ps6509/products_data_sheets_list.html.

• High-availability support is achieved through an active-active redundancy configuration, which also supports up to 500 ports in the server pair.

• You can use advanced Cisco Unity Connection to Cisco Unity networking to allow both solutions to be networked together transparently.

• Cisco Unity Connection supports the synchronization of user information using LDAP with Microsoft Active Directory 2000, 2003, and 2008; Sun One; Sun iPlanet; and Netscape Directory Server, enhancing your deployment and administrative options.

• Cisco Unity Connection allows for separation of an active-active pair across data centers (geospatial separation), providing greater deployment options for the enhanced reliability of high availability across the WAN.

• Cisco Unity Connection supports Voice Profile for Internet Messaging Version 2 (VPIMv2), which allows networking of up to 10 Cisco Unity, Cisco Unity Express, or third-party voicemail systems, allowing users on each of these systems to transparently reply to, forward, and exchange voice messages.

• Phone-system integrations include any phone system that provides a serial data link (Simplified Message Desk Interface [SMDI], Message Center Interface [MCI], or Message Digest Algorithm 110 [MD110] protocol) to the master PBX IP media gateway (PIMG) unit (serial integration through analog PIMG or T1 IP media gateway [TIMG] units). • Use TIMG units for in-band integration with Avaya Definity G3. • Use TIMG units for in-band integration with Avaya S8500 and S8700.

• Cisco Unity Connection integrates with Cisco Unified Communications Manager and Cisco Integrated Services Routers using QSIG.

• Cisco Unity Connection integrates with Cisco Unified Communications Manager and leading traditional telephone systems, even simultaneously (using the PIMG or TIMG).

• Cisco Unified Communications Manager 4.1(3) and higher, Cisco Unified Mobility Advantage, and Cisco Unified Mobile Communicator are supported.

• Cisco Unity Connection natively supports SIP proxy servers, designated SIP phones and clients, and SIP-capable access gateways.

• Cisco Unity Connection provides a browser-based system administration console and tools for easy installation and maintenance.

• City and Department fields are available for administratively defined contacts.

System Administration Features

• Alternate extensions are configurable by the system administrator or user.

• Alternate key mappings for message retrieval can help you transition from traditional voicemail systems.

• Custom keypad mapping allows administrators to create TUIs for specific user needs.

• Automatic gain control provides consistent message volume playback levels.

• Handlers provide building blocks for Automated-Attendant and intelligent call-routing functions. • Call handlers accept calls, play recorded prompts, route calls, and accept messages. • Directory handlers manage the way that callers search the directory. • Interview handlers collect and record input from callers.

• You can customize directory handlers with a voice greeting.

• You can configure per-user message-handling actions to determine how messages of specific types are handled in the system, such as “accept the message”, “reject the message”, or “relay the message”.

• Caller ID is supported.

• Call screening is configurable.

• Class of service (CoS) controls user access to features.

• Administrators can create users individually or in bulk.

• Administrators can import users from Cisco Unified Communications Manager.

• Messages are day and time stamped.

• You can perform a directory search by spelling a username; you can enter up to 24 letters.

• You can log in to the TUI without entering your ID.

• Representational State Transfer-based APIs for provisioning and messaging allow integrations with existing corporate provisioning tools or messaging clients.

• Cisco Unity Connection 8.5 supports IPv6 addressing with Cisco Unified Communications Manager (7.1(2) or later) phone system integrations using SCCP and SIP. The addressing mode is configurable by port group.

• Encrypted SCCP, Secure Real-Time Transport Protocol (SRTP), and Transport Layer Security/SRTP (TLS/SRTP) for SIP facilitates Cisco Unified Communications Manager integration.

• SIP support includes the following: • TLS/SRTP: Cisco Unified Communications Manager SIP trunk integrations support authentication and encryption of the Cisco Unity Connection voice messaging ports. • Keypad Stimulus Protocol (KPML): For Cisco Unified Communications Manager SIP trunk integrations, administrators can configure the integration to send dual tone multifrequency (DTMF) keystrokes in the Real-Time Transport Protocol (RTP) media stream (in-band) or in a SIP message (out-of-band). • Port multiplexing: SIP integrations (such as for PIMG, TIMG, or Cisco SIP Proxy Server) can share the same SIP port on the Cisco Unity Connection server.

• Simple Network Management Protocol (SNMP) Versions 1, 2, and 3 are supported.

• Event logging is supported.

• Full mailbox warning is supported.

• You can create folders within a mailbox for inbox, deleted items, sent items, and draft items.

• Installation is simple and quick.

• A list of observed holidays is configurable.

• You can configure how Cisco Unity Connection handles messages that are interrupted by disconnected calls.

• MWI is supported, including enhanced MWI that displays a constant message count on certain Cisco Unified IP Phones.

• Multiple administrative levels allow you to control access to pages in the system administration GUI by CoS (read, modify, or delete rights).

• Music on hold (MOH) is supported.

• Nondelivery or delivery receipt reason details are presented in the GUI inbox.

• You can specify the public distribution lists to which new users will be added.

• Restriction tables are configurable.

• You can exclude return receipts.

• The system schedule is configurable.

• Self-enrollment allows you to set your password, record your voice name, and specify your directory listing.

• A status monitor allows for real-time administrator status of telephone ports, reports in progress, and system configuration.

• System broadcast messages for officewide announcements are supported.

• System greetings are configurable.

• The system offers 12- and 24-hour clock support for time stamps.

• The system time clock adjusts automatically for Daylight Savings Time.

• A TUI greetings administrator (Cisco Unity Connection Greetings Administrator) is supported.

• LDAP directory integration allows users to be quickly imported, synchronized, and authenticated within the directory.

• You can create up to nine mailbox stores in addition to the default mailbox store that is created when Cisco Unity Connection is installed.

• You can simulate abbreviated extensions by using prepended digits for call handlers and user mailboxes.

Security

• A host intrusion prevention system, the Cisco Security Agent standalone agent, protects Cisco Unity Connection servers from worm and virus attacks; an optional Cisco Security Agent management console is available.

• Password and PIN security policy options to enforce expiration, complexity, reuse, and lockout are supported.

• Call-restriction tables to prevent toll fraud are supported.

• Security event logging and reports of failed login and account lockouts to help prevent unauthorized PIN use are supported.

• Secure, private messaging prevents the playing of private messages accidentally forwarded outside the enterprise.

• A message aging policy for secure messages automatically deletes all secure messages that are older than the specified number of days.

• Message aging policies can be set on a per-user basis.

• Secure RTP and signaling encryption provides for secure communication between Cisco Unity Connection and Cisco Unified Communications Manager.

• A user telephone PIN reset feature in Cisco Unity Connection Assistant reduces help-desk calls and operating expenses.

• Support for Secure HTTP (HTTPS) provides for secure web access to Cisco Unity Connection and allows for playback of secure messages within Microsoft Outlook.

Voice Quality

• G.722 and Internet Low Bitrate Codec (iLBC) voice codecs are supported (advertised or “on the line”). G.711 mu-law, G.711 a-law, and G.729 are also supported.

• System-level recording is available for linear pulse code modulation (PCM), Global System for Mobile Communications (GSM) 6.10, G.711 mu-law, G.711 a-law, G.729a, and G.726 through system-based transcoding resources.

Reports

• Call Handler Traffic Report

• Distribution Lists Report

• Events Report

• Outcall Billing Report

• Port Usage Report

• Users Report

• User Message Activity Report

• System Configuration Report

• Transfer Call Billing Report

• User Access Activity Report

• User Lockout Report

• Message Traffic Report

• Port Activity Report

• Mailbox Store Report

• Dial Plan Report

• Dial Search Scope Report

• For a full list and description of reports, refer to the Cisco Unity Connection System Administration Guide at:http://www.cisco.com/en/US/products/ps6509/prod_maintenance_guides_list.html.

Localization

The Cisco Unity Connection TUI, end-user GUI, and TTS engine are available in the following languages: • Arabic (no TTS) • Catalan • Chinese (Hong Kong, Mandarin TUI with simplified and traditional Chinese GUI, simplified Mandarin TTS, but no traditional Mandarin TTS) • Czech • Danish • Dutch • English (U.S., U.K., and Australian) • English TTY • French (European and Canadian) • German • Greek • Hungarian • Italian • Japanese • Korean • Norwegian • Polish • Portuguese (Brazilian and European) • Russian • Spanish (European and Latin American) • Swedish • Turkish (no TTS)