All About Cisco Multicast MOH Configuration


  • Overview
  • Configuration Check
  • Troubleshooting & Common Problems



Configuration Check

Make sure that you have the IP Voice Media Streaming App turned “On” in order for your server to start streaming the Multicast packets outbound toward the gateway.


Make sure that you have your MOH file loaded in to each and every CUCM server that you are using in the cluster. This is a bit counterintuitive because you think you are configuring the server to play the MOH file that you are loading. Instead we do this simply to be able to select the MOH file to play whenever the phone is configured.

Here we verify how long our MOH file is as well as verifying its “In Use”


This page allows us to make the configuration change for Multicasting as well as verify what codecs are supported with the stream that you’ve installed.


This page shows us how many MOH servers we can configure as well as where exactly they are registered to. Many times customers don’t know that these must show up as “Registered with …” in order to work properly. Be sure to check this and then reset these as needed.


MRG setup as well as the Multicast checkbox that you must enable.


MRGL setup that allows us to place our MRG inside the selected resource groups in order for us to be able to access our MOH server and file correctly.


This is the phone configuration page and the locations that you must configure if you want to access an MOH server/file. You don’t need an MRGL here if its taken care of on the Device Pool.


This is the line configuration page and the locations that you could possibly configure if you want to access an MOH


This is the Device Pool where I’ve configured an MRGL (optional), but only if its not taken care of at the Phone Configuration page.


This is the MOH server configuration where I enable Multicasting and then define how I will configure the multicast IP on the gateway as well as the port number.


This is the chart that shows you how the IP’s and ports change depending on the previous Slides configuration.


This is the gateway configuration that I’ve done, and you see where I’ve added the “Pim sparse-dense-mode” command that is needed.


Below is the configuration setup for multicast routing that you must enable throughout the entire network.


You will need to place this command on the gateway in order for the MOH to begin streaming outbound from the gateway to the PSTN/Outbound Interfaces as desired.


Troubleshooting &Common Problems Common Problems

  • I hear TOH (Tone-On-Hold) and not able to hear my MOH file that I expect
  • Verify Audio Source and Multicast Flag are set correctly on the traces, if they are then this is most likely a gateway issue.
  • I just hear silence and my file isn’t playing
  • Verify Audio Source and Multicast Flag are set correctly on the traces, if they are then this is most likely a gateway issue.
  • Verify Regions and Codecs being used, if H.323 remember that Dial-Peer 0 uses G.729 automatically so be sure to specify the correct codec if needed.

Below is the command that you can run from the gateway that will show you the MOH file that is configured to play which must match up to the file that is expected to be streaming. You can also see the Multicast IP address involved with the command listed. The IP address is the Loopback address. You can use either the loopback or a physical interface for your setup.


This is the command that you run from the SSH session on the CUCM servers. The Following command will spit the output of the Multicast MOH servers that are configured and what their port/IP address is. We are incrementing on the IP address for my setup which means that the IP addresses will Increment here as well. For each different IP address you involve a different codec negotiated.


The following command will be input on the MOH gateway that we’ve involved. The output of this command will display the active MOH sessions on that particular GW as desired. From this command we can see the Multicast IP address, RTP Port, Packet count (which should be incrementing), call ID, Codec negotiated, and Incoming interface being used


The command ran on the gateway here is “debug MGCP Packets”and from that output we can see where the multicast IP address is detected and then used to stream all relative information as needed.








  • Verify through the CUCM traces first that you have your AudioSource ID and Multicast Flag enabled.
  • Proceed to the gateway, verify the codec information and gateway configuration commands are set correctly. If you are receiving silence try using another file you’ve verified to work or speak with the MS team.
  • Traces to obtain:CM traces set to “Detailed” and pull after making a test callI PVMS traces set to detailed to confirm that its working correctly Debug MGCP packets (if MGCP)Debug voip CCAPI inout Show ccm-manager music-on-hold (whenever you enable the call)

#moh-server, #multicast-ip-address, #multicast-moh

Cisco UCM User Licensing

Originally posted on Bsoft Bangalore:


  • UCM Licensing High Level design
  • Phone classification (a subset)
  • License Tier classification
  • Screen shots of the Licensing UI pages
  • Troubleshooting tips
  • Feature documentation

UCM Licensing High Level Design


Phone Classification

  • Tin: analog, Cisco 3905, Cisco VGC phone, Cisco VGC Virtual phone
  • Copper: Cisco 6901, Cisco 6911, Cisco 6921, CUC-RTX
  • Bronze: Most phones
  • Silver: Cisco IP Communicator, Cisco IP Personal Communicator, IMS integrated mobile, Unified Client Services Framework
  • Gold: Cisco Dual Mode for Android, Cisco Dual Mode for iPhone, Cisco Unified Mobile Communicator,
  • Telepresence: Carrier-Integrated Mobile

License Tiers

Essential Basic Enhanced Standard Premium Professional Telepresence
Phones (number) 1 1 1 2 6 10 1
Phone Type
Tin Y Y Y Y Y Y
Copper Y Y Y Y Y
Bronze Y Y Y Y
Silver Y Y Y
Gold Y Y
Tele-presence Y

Licensing States on UCM

View original 466 more words

Collaboration update information videos

TP & VIDEO Everywhere With Video Collaboration – YouTube

Cisco Collaboration launch March 2014 – new units – YouTube

Cisco Partner Summit 2014 Global General Session Demo: TelePresence SX10 – YouTube

Cisco Partner Summit 2014 Collaboration Breakout Demo: TelePresence MX700 – YouTube

Dynamic Video Meetings with SpeakerTrack 60 – YouTube

Collaboration Summit 2013 Keynote Demo: Cisco TelePresence MX300 G2 – YouTube



Cisco BE6000 Overview – YouTube

Experience Ease-of-Use with Cisco Business Edition 6000 – YouTube

Fundamentals of the Unified Communications BE 6000 – YouTube

Cisco IP Phone 7800 Series – YouTube

Cisco DX650: User Experience – YouTube



Cisco Jabber – IM Presence Voice and TelePresence on ANY DEVICE ANYWHERE! – YouTube

Cisco Jabber Guest Explained in Detail – YouTube

Jabber Specialist Training – Presentations and Recordings

Microsoft Competitive Material


CCIE Collaboration Lab

Lab Equipment:

  • Cisco Unified Computing System (UCS) C460 Rack Server
  • Cisco 3925 Integrated Services Routers Generation 2 (ISR G2)
  • Cisco 2921 Integrated Services Routers Generation 2 (ISR G2)
  • ISR G2 Modules and Interface Cards
    • – 1-Port 3rd Gen Multiflex Trunk Voice/WAN Int. Cards – T1/E1
    • – Cisco High-Density Packet Voice Digital Signal Processor Modules (PVDM3)
    • – Cisco Service Ready Module 710 Service Module with Cisco Unity Express
    • – 4-port Cisco Gigabit EtherSwitch 10/100/1000BASE-TX autosensing EHWIC with POE
  • Cisco Catalyst 3750-X Series Switch
  • Cisco Unified IP Phones 7965 and 9971
  • Cisco Jabber for Windows
  • Cisco Jabber Video for Cisco Telepresence*

*In backbone, no candidate configuration required

Software Versions
Any major software release which has been generally available for six months is eligible for testing in the CCIE Collaboration Lab Exam.

  • Cisco Unified Communications Manager 9.1
  • Cisco Unified Communications Manager Express 9.1
  • Cisco Unified Contact Center Express 9.0
  • Cisco Unified Communications Manager IM and Presence 9.1
  • Cisco Unity Connection 9.1
  • Cisco Unity Express 8.6
  • All routers use IOS version 15.2(4) M Train
  • Cisco Catalyst 3750 Series Switches uses 15.0(2) Main Train

Network Interfaces

  • Fast Ethernet
  • Frame Relay

Telephony Interfaces

  • T1/E1

Cisco Press Resources

 CCIE Collaboration Reading List:

     Cisco IP Telephony (CIPT) (Authorized Self-Study Guide), 2nd Edition

     Cisco Voice Gateways and Gatekeepers

     Fax, Modem, and Text for IP Telephony

     SIP Trunking

     Cisco Unified Presence Fundamentals

     Cisco Unity Connection

     Securing Cisco IP Telephony Networks

     Troubleshooting Cisco IP Telephony

Cisco Solutions Reference Network Design (SRND) and Design Guides

     Cisco Unified Communications System 9.x SRND

     Cisco Unified Contact Center Express Solution Reference Network Design Release 9.0(1)

     Design Guide for Cisco Unity Connection Release 9.x

     Cisco Unified CallManager Express Solution Reference Network Design Guide

Administration and Deployment Guides

     Cisco Unified Communications Manager Documentation Guide, Release 9.0(1)

     Cisco Unified CCX Administration Guide, Release 9.0(1)

     System Administration Guide for Cisco Unity Connection Release 9.x

     Deployment Guide for IM and Presence Service on Cisco Unified Communications Manager, Release 9.0(1)

Cisco Unified Communications Manager Express System Administrator Guide

Coming soon in Bsoft Bangalore 

#30, 1st floor,1st Main,BTM 2nd stage
Bennergetta Road,Bangalore,India

Mobile: 09886623909
Phone : 080 4146 5262

CCIE Collaboration | 400-051

Cisco Collaboration Infrastructure

  • UC Deployment Models
  • User Management
  • IP Routing in Cisco Collaboration Solutions
  • Virtualization in Cisco Collaboration Solutions
    • UCS
    • VMware
    • Answer Files
  • Wireless in Cisco Collaboration Solutions
  • Network Services
    • DNS
    • DHCP
    • TFTP
    • NTP
    • CDP/LLDP
  • Power over Ethernet
  • Voice and Data VLAN
  • IP Multicast
  • IPv6

Telephony Standards and Protocols

  • SCCP
    • Call Flows
    • Call States
    • Endpoints types
  • MGCP
    • Call Flows
    • Call States
    • Endpoints types
  • SIP
    • Call Flows
    • Call States
    • SDP
    • BFCP
  • H323 and RAS
    • Call Flows
    • Call States
    • Gatekeeper
    • H.239
  • Voice and Video CODECs
    • H.264
    • ILBC
    • ISAC
    • LATM
    • G.722
    • Wide band

Cisco Unified Communications Manager

  • Device Registration and Redundancy
  • Device Settings
  • Codec Selection
  • Call Features
    • Call Park
    • Call Pickup
    • BLF Speed Dials
    • Native Call Queuing
    • Call Hunting
    • Meet-Me
  • Dial Plan
    • Globalized Call Routing
    • Local Route Group
    • Time of Day Routing
    • Application Dial Rules
    • Digit Manipulations
  • Media Resources
    • TRP
    • MoH
    • CFB
    • Transcoder/MTP
    • Annunciator
    • MRG/MRGL
  • CUCM Mobility
    • EM/EMCC
    • Device Mobility
    • Mobile Connect
    • MVA
  • CUCM Serviceability and OS Administration
    • Database Replication
    • CDR
    • Service Activation
    • CMR
  • CUCM Disaster Recovery
  • ILS/URI Dialing
    • Directory URI
    • ISL Topology
    • Blended Addressing
  • Call Admission Control
    • RSVP
    • SIP Pre-conditions
  • SIP and H323 Trunks
    • SIP Trunks
    • H.323 Trunks
    • Number Presentation and Manipulation
  • SAF and CCD
  • Call Recording and Silent Monitoring

Cisco IOS UC Applications and Features

    • SCCP Phones Registration
    • SIP Phones Registration
    • SNR
  • SRST
    • CME-as-SRST
    • MGCP Fallback
    • MMOH in SRST
  • CUE
    • AA
    • Scripting
    • Voiceview
    • Web Inbox
    • MWI
    • VPIM
  • IOS Based Call Queuing
    • B-ACD
    • Voice huntgroups
    • Call Blast
  • IOS Media Resources
    • Conferencing
    • Transcoding
    • DSP Management
  • CUBE
    • Mid-call signaling
    • SIP profiles
    • Early/Delayed offer
    • DTMF interworking
    • Box-to-Box Failover/Redundancy
  • Fax and Modem Protocols
  • Analog Telephony Signaling
    • Analog Telephony Signaling Theories (FXS/FXO)
    • Caller ID
    • Line Voltage Detection
    • THL Sweep
    • FXO Disconnect
    • Echo
  • Digital Telephony Signaling
    • Digital Telephony Signaling Theories (T1/E1, BRI/PRI/CAS)
    • Q.921 and Q.931
    • QSIG
    • Caller ID
    • R2
    • NFAS
  • IOS Dial-Plan
    • Translation Profile
    • Dial-peer matching logics
    • Test Commands
  • IOS Accounting

Quality of Service and Security in Cisco Collaboration Solutions

  • QoS: Link Efficiency (e.g. LFI, MLPPP, FRF.12, cRTP, VAD)
  • QoS: Classification and Marking
    • Voice vs Video Classification
    • Soft Clients vs Hard Clients
    • Trust Boundaries
  • QoS: Congestion Management
    • Layer 2 Priorities
    • Low Latency Queue
    • Traffic Policing and Shaping
  • QoS: Medianet
  • QoS: Wiress QoS
  • Security: Mixed Mode Cluster
  • Security: Secured Phone Connectivity
    • VPN Phones
    • Phone Proxy
    • TLS Proxy
    • 802.1x
  • Security: Default Security Features
  • Security: Firewall Traversal
  • Security: Toll Fraud

Cisco Unity Connection

  • CUCM and CUCME Integration
  • Single Inbox
  • MWI
  • Call Handlers
  • CUC Dial-plan
  • Directory Handlers
  • CUC Features
    • High Availability
    • Visual Voicemail
    • Voicemail for Jabber
  • Voicemail Networking

Cisco Unified Contact Center Express (UCCX)

  • UCCX CTI Integration
  • ICD Functions
  • UCCX Scripting Components

Cisco Unified IM Presence

  • Cisco Unified IM Presence Components
  • CUCM Integration
  • Cisco Jabber
  • Federation
  • Presence Cloud Solutions
  • Group chat and Compliance

Cisco Press Resources

Cisco Solutions Reference Network Design (SRND) and Design Guides

Administration and Deployment Guides


All about SIP

Session Initiation Protocol (SIP)
SIP is a signaling protocol widely used for voice and video.
Traditionally if your enterprise wants to connect to PSTN could, you would need BRIs, PRIs, PSTN gateways.
With SIP Trunking service you can avoid all the Network PSTN connections.

What is SIP Trunk:
Internet telephony service provider (ITSP) offer SIP trunk as a service for communication between enterprise PBX and PSTN.
IP Phone——-CUCM———–SIP Trunk————-ITSP——–PSTN
Each SIP trunk can have multiple voice session based on enterprise needs.
G.711 – 17 calls over T1
G.729a – 45 calls over T1
SIP trunk is not limited to voice, it can also help enterprise to setup instant messaging, real-time presence, video, etc.
For more details refer to RFC 3261

Benefits of having SIP Trunk:

  1. Like I said no need to invest in PSTN gateway and voice cards.
  2. Off course low cost compared to traditional PRI T1.
  3. Reach out to the world on cost of local call.
  4. Easy installation and maintenance.
  5. Optimal use of bandwidth as data and voice run on same connection.
  6. Move away from T1/E1 capacity limitations of 23/30 channels.
  7. SIP trunk normalization and transparency
  8. Up to 16 destination IP addresses per trunk
  9. Automated hunting from a primary PBX to a secondary
  10. Automated failover to PSTN with full trunk

For you to deploy SIP in your enterprise there are two things which needs to be there.

  1. SIP enabled PBX
  2. SIP enabled edge device

Bandwidth Utilization:
It is always good idea to reserve 27 Kbps with G.729  per call and 84Kbps for G.711

Minimum Bandwidth Requirement
Codec Voice Bit Rate
G.711 64 Kbps
G.729 8 Kbps


Cisco Recommendations:

Application H.323 MGCP SIP Preferred
Voice Mail Control of Individual Ports MGCP
Configuration Dial Peer Based Centralized in CUCM Dial Peer Based MGCP
Load on CUCM Least MGCP
Q.SIG Tunneling Only between PBXs Supported MGCP
Video H.320 ISDN H.320 ISDN H323/SIP
Fax & Modem Pass-through, T.38 Pass-through, T.38 Pass-through, T.38 H323/SIP/MGCP
Port Density High Density Cards MGCP
Redundancy Range of options with dial-peers Range of options with dial-peers H.323/Sip
Security IPSec and SRTP IPSec and SRTP TLS and SRTP SIP
Voice XML Supported Supported H.323/SIP


 Centralized SIP Trunk Design Limitations

MoH Centralized MoH Limited to 50
Central Site Device Pool Multiple Device Pools for Devices on Datacenters
Non Ported DIDs Requires a Different Call Flow and Different Call Routing
FAX Not Supported on SIP Trunk; Handled by Site GW
SRST Limited Access via FXO, PRI for Medium/Large Site
DTMF SIP Trunk and Check IP Phone

Basic SIP Trunk configuration in CUCM

  1. Step 1:
    1. Create a SIP profile (optional).
    2. Create a SIP trunk security profile (optional)
    3. Create a SIP trunk.
    4. Configure the destination address.
    5. Configure the destination port.
  2. Step2: Associate the SIP trunk to a Route Pattern or Route Group.
  3. Step3: Configure SIP timers, counters, and service parameters, if necessary.
  4. Step4: Reset the SIP trunk

Common SIP Requests

  1. REGISTER – to register a phone or line with a SIP Server
  2. INVITE – to set-up a call
  3. CANCEL- to cancel a call set-up
  4. BYE – to terminate a call

Common SIP Responses

  1. 100 trying
  2. 180 ringing
  3. 200 OK
  4. 401 not authorized
  5. 404 destination not found
  6. 486 busy

SIP Trunks vs. H.323 Trunks (Inter cluster)

H.323 SIP
Annex M1 Features / Q.SIG Tunneling NO NO
Signal Authentication No YES
Media Encryption YES YES
GK Support YES NO
SIP Proxy Support NO YES
iLBC and G.Clear Support No YES
G.722 Support YES YES
Multicast MoH YES YES
SIP Subscribe/Notify, Publish-Presence NO YES
Path Replacement NO NO
Call Completion to Busy Subscriber NO NO
Call Completion No Reply NO NO
Message Waiting Indicator (On/ Off) No YES
Alerting Name NO YES


SIP Trunking byChristina Hattingh, Darryl Sladden, ATM Zakaria Swapan Published by Cisco Press.

Cisco Unified Communications System 8.x SRND
BRKUCC-2006 – SIP trunk design and deployment
BRKCCT-2030 – SIP based Architectures for Cisco Contact Center Solutions & Collaboration
BRKUCC-2735 – SIP Trunk Design and Deployment Playbook for the Enterprise
BRKUCC-2450 – Planning for SIP trunking and dial plan centralization with SME


All about UC 500

Technical Enablement Collateral Helps Partners


This document includes:

  1. Links to documents and Materials you need to be famaliar with
  2. Technical Enablement Labs that show you how to configure basic functions
  3. TEL Template

This is a community document open for editing as you see fit.


Links to Documents


For any Partner who wants to build a successful practice deploying UC 500 (the new UC 540/UC 560 or the legacy UC 520), there are certain materials you need to become famaliar with and as well some actions you need to take to have your Technicians and Engineers feel comfortable building and deploying UC 500.


Small Business University
Select Certification <– for both AM and SE


First Look SBCS UC540 2.0 Lab
Distance Learning Opportunity for Channel Partners <– for both AM and SE


Smart Designs
Design and implementation notes to show typical deployments <— for the SE


Integrating the UC 500 System with an Existing Customer’s Firewall


SBCS Feature Reference Guide (CCA configurable)

No longer requires login; accessible to partners and customers. <– for both AM and SE


Administration guide for CCA



The CCA Release Notes give you additional information about whats new as well as any known issues with each release

The CCA Prerequisite Checks must be performed on the PC you plan to run CCA on and will save you days of wasted time



Platform Reference Guides <– for the SE





make sure you understand the SKUs supported and the capacity of the system as defined in these great guides


Quick Pricing Tool (QPT) <— for the AM and SE (reduce Quoting time from Days to Minutes)



Use the Site Survey <— for AM & SE

Cisco Steps to Success for VoIP – UC500 SMB Version

Collect what you need to build the system (much of it in a staging environment) before you deploy on site.


SBSC System Test information (how it was tested, which code was used, what the results were)


Third Party Applications to be understood:



In addition to emmedded applications like SNR, WEBEX, TCV, IMAP Mail Integration, Video telephone, learn about third party integrations


Possible Milestones you can achieve as a measure of your ability to deploy.

  • Select Certification and identification of key individuals in for roles and responsibilities
  • get NFR Gear (UC500, ESW switches, Phones, AP541s, which you can demo to your customers)
  • Training touches – attend the virtual trainings and perform the TELs below in this document
  • Build your lab
  • Perform Demos


SBCS User Guides (including phones):


Get the Latest CCA Code:

Let the Latest UC500 Bundle:

Please take a look at the UC 500 Software Pack Roadmap posted at

Office Manager (OM) Information and SW

Office manager Installation Guide



Technical Enablement Labs (TEL) and Videos (VOD)

As SBCS moves forward, it does so with the UC 500 SWP, a given CCA version and the Office Manager (OM) version that work together to deliver new features and functions. This TEL lab area is updated by partner demand on the support community, many times to answer discussion threads that we feel should be answered more completely with lab demonstration. While we try to obsolete older labs to the bottom, there are still some older labs not yet replaced and so you should always consult the CCA OLH and Admin Guide as well as latest Smart Designs (which normally lag behind TELs but become more officially supported docs).


When we Announced CCA 3.0 and Office Manager 1.2, we delivered two Partner Webcasts and the .ppt used are here:


CCA 3.0 (new capabilities) Demonstrated (Webcast ppt from 1/13/2011)

Office Manager 1.2 technical Webcast/demo for partners 1/20/2011


Initial Maintenance and Configuration with CCA

Creating and Managing Customer Sites with CCA TEL


SW Upgrade: SWP 8.1.0 with CCA 3.0 TEL


Configuring a UC500 Demo System with CCA 3.0 TSW and Bulk User and Phone Import


UC 500 Localization using CCA 3.0 SW Upgrade Utility VOD


TSW for Bulk User and Phone Data Import using BulkDataImport.xls in CCA 3.0 VOD


Upload and Download files to UC 500 Flash using CCA 3.0 VOD


Upgrading or Installing a New License on a UC540/560 using CCA 3.0 TEL


Adding a VLAN to UC 500 TEL


Features and Functions Labs and VODs


The Remote Teleworker Router Configuration with CCASPA525G_SSL_Remote_TEL_19.pdf


SPA525G Remote SSL VPN Teleworker to UC 500


Users And Extensions Assignment in CCA 3.0 with Hold Alert and Speed Dial TEL Connectivity Status Diagnostics and Recovery with CCA VOD


Multiple WAN IP to LAN Static NAT Mappings with CCA 3.0 VOD


CCA Tele-worker Phone Support MTP Configuration VOD


Live Record Configuration and Beep Adjustments with CCA 3.0 TEL


Removing Live Record Beep Tones with CCA 3.0 VOD


Configuring Combined Paging Groups with CCA TEL


Paging Group Configuration with CCA VOD


SKYPE for SIP Configuration with CCA TEL


Configuring Remote Access to Administraton via Telephoine (AvT) for your AA with CCA TEL


Auto Attendant Script Customization (Alternate Greeting and Dial by First Name) with CCA TEL


Assigning FXO ports as “CO Lines” and assigning them to phones


Manual Override to AA TOD Schedule using Night Service


Configuring the UC500 to work with Call Accounting and Call Recording from ISI


SA500 Configuration with CCA TEL


Configuring and Operating FXO Trunk Groups with CCA TEL


Hybrid Key System Configuration with CCA


Webex Phone Connect configuration with CCA 3.0


Lab 3: UC540 EZVPN Configuration

Lab 4: SIP Trunk Configuration

Lab 6: SA500 Security Applicance in front of a UC500 with SIP trunking

Lab 7: B-ACD (Basic ACD) Configuration

Lab 8: Auto Attendant and Night Service

Lab 9: Multisite using CCA 2.1 Multisite Manager

Lab 10: Protect Link Gateway Service on SA 500 in front of UC540

Lab 11: Smart Application: TimeCardView

Lab 12: Smart Application – IMAP Unified Messaging

Lab 14: Unified CallConnector (UCC)

Lab K: Traffic Shaping and Max Connections (CAC),

Lab 15: SR520-T1 Configuration with CCA

Lab 16: Video Monitoring xVC2300 IP Cameras from SPA525G Phone

Lab 17: Smart Call Connector Attendant Console Operator

Lab 22: Disable Cisco IP Phone WEB Server access on UC500

LAB 24: NTP on SBCS (UC500 & SBCS)

Lab 25: TheOffice Demo configured using CCA 3.0 (Updated)

How to DEMO the OFFICE SBCS – Demo Guide

The actual DEMO Performed (54 minutes run time)

Lab 26: SNR Smart Application Configuration and Operation on UC500 with CCA

Lab 28: UC560 Voice Mail Expansion using upgrade of external CUE Flash to 8G

Lab 30  UC500  Access List Manager – An example on how to restrict Guest Network access

NOW OBSOLETE Labs Archive (based on older SWP and CCA version)

LAB 1: UC540 Initial Connection, Discovery, SW Upgrade, Phone Load Management, and Licensing

Lab 13: Smart Application – Live Record

Lab 20: Cisco Phone Firmware Upgrade (79xx models – manual)

Lab 23: Paging Groups on UC500

Lab 18: Auto Attendant custom script configuration for ‘no action transfer

Lab 21: SMTP Notification of UC500 Voice Mail to EMAIL

Lab 27: Configuring Alternate Greeting on UC500 Auto Attendant

Lab 2: UC540 Telephony Setup Wizard in Staging (staging without phones)

Lab 5: UC540 FXO PSTN Trunk Group Configuration