Oct
05

INE is happy to announce that we now have all 21 Modules of our new CCIE Voice Deep Dive completed --115 hours of recorded class-on-demand style video (no breaks or dead-air in the recordings - that's 115 hours of actual learning!)-- completed and ready for your consumption!

As we mentioned in a previous post, The author and poet Maya Angelou said “Words mean more than what is set down on paper. It takes the human voice to infuse them with deeper meaning.”. Well that is certainly what we have attempted to do with the CCIE Voice Deep Dive self-paced Class on Demand series – that is to bring the human instructional voice element to infuse deeper meaning to what is already fantastic Cisco Documentation. Anyone that has set out and determined to undertake the task of studying for and ultimately passing any CCIE Lab exam, knows that at some point during your studies, the words on paper (Cisco Docs, RFCs, books) – while a absolute phenomenal source of information – can at times seem to loose their impact. Perhaps you have been studying too long, read one too many docs, have the time pressure of your family and friends waiting for you to return to be a part of their life, or perhaps you are just starting out on your adventure and don’t know where to begin. Whatever stage you are at or whatever the case may be, it is certainly helpful to have a tutor and mentor there beside you at times, assisting you in understanding what each complex technology’s documentation is trying to teach you, in possibly a deeper and more insightful way than you can manage on your own.

For each complex topic we have held (or will soon hold) an online class where we dive down deep and explore all the concepts, practical application, and troubleshooting associated with each technology topic. The general format for each Class-on-Demand Deep Dive module spends between 4-7 hours on the given topic for that day, and during that time follows this outlined training methodology:

  • Collectively discuss and teach all concepts involved in the technology
  • Whiteboard concepts to further deepen every participant’s understanding
  • Define a specific set of tasks to be accomplished
  • Demonstrate how the tasks and concepts are implemented and properly configured
  • Test the configuration thoroughly
  • Vary the configuration to understand how different permutations effect the outcome
  • Debug and trace the working configuration to understand what should be seen
  • Break the configuration and troubleshoot with debugs and traces to contrast from the working set

Before we go on with the 21 module outline, here are a few demos of this Deep Dive series:

Demo 1: Module 10 :: Dial Plan :: Globalization Prezi - Theory and Reasons :: Runtime 1 hr

Demo 2: Module 10 :: Dial Plan :: Inbound Calling Party Localization :: Runtime 30 mins

Demo 3: Module 12 :: CUBE :: Conforming to ITSP Reqs: SIP Header Conversions :: Runtime 51 mins

Demo 4: Module 13 :: Unified Mobility :: Mobile Connect Access Lists and Exclusivity :: Runtime 20 mins

Here is the outline for the complete Deep Dive series:

Network Infrastructure

Module 1 :: Network Infrastructure, RSVP CAC, and LAN & WAN Quality of Service :: Runtime 10.5hrs

  • NTP
  • VLANs
  • TFTP
  • DHCP
  • Multicast (Infrastructure)
  • LAN QoS - including:
  • Catalyst 3560/3750 Classification and Marking
  • Catalyst 3560/3750 Conditional Trust
  • Catalyst 3560/3750 Ingress Interface Mapping
  • Catalyst 3560/3750 Ingress Interface Queuing
  • Catalyst 3560/3750 Ingress Interface Expedite Queue
  • Catalyst 3560/3750 L2 CoS to L3 DSCP Mapping
  • Catalyst 3560/3750 Egress Interface Mapping
  • Catalyst 3560/3750 Egress Interface Queuing
  • Catalyst 3560/3750 Interface Queue Memory Allocation
  • Catalyst 3560/3750 Egress Queue-Set Templates
  • Catalyst 3560/3750 Weighted Tail Drop (WTD) Buffer Allocation
  • Catalyst 3560/3750 Egress Interface Expedite Queue
  • Catalyst 3560/3750 Egress Interface Sharing
  • Catalyst 3560/3750 Egress Interface Shaping
  • Catalyst 3560/3750 Scavenger Traffic Policing
  • CUCM RSVP-Based Locations for Call Admission Control
  • WAN QoS Classification
  • WAN QoS Low Latency Queuing (CBWFQ-PQ)
  • WAN QoS Traffic Shaping
  • WAN QoS Frame-Relay Fragmentation

Unified Communications Manager

Module 02 :: CUOS GUI and CLI Admin :: Runtime 3.6 hrs

  • CUCM WebUI: Service Activation and Stop/Start/Reset
  • CUCM WebUI: Bulk Administration Tool (Import/Export, Phone Reports, etc)
  • CUCM WebUI: DB Replication Status
  • CUCM WebUI: Trace Files
  • CUOS CLU: TFTP Files Management
  • CUOS CLU: Status and Hostname
  • CUOS CLU: DB Replication Assurance
  • CUOS CLU: DB Replication Repair and Cluster Reset
  • CUOS CLU: Trace Files
  • CUOS CLU: RIS DB Search
  • CUOS CLU: Performance Monitor (PerfMon)
  • RTMT: Trace Files
  • RTMT: Performance Monitor (PerfMon)

Module 03 :: CUCM System and Phone - SCCP and SIP Fundamentals :: Runtime 4.4 hrs

  • CUCM Services
  • UC Servers and Groups
  • Date/Time with NTP Reference
  • Regions and Codecs
  • Location-Based Call Admission Control
  • SRST References
  • Device Pools
  • System Parameters
  • Enterprise Parameters
  • Phone Button Templates
  • Softkey Templates
  • SCCP Phone Basics
  • SIP Phone Basics

Module 04 :: Users, Credentials, Multi-Level Roles and LDAP Internetworking :: Runtime 3.6 hrs

  • CUCM User Credentials and Policies
  • LDAP Synchronization for CUCM and Unity Connection
  • LDAP Authentication for CUCM and Unity Connection
  • CUCM End Users
  • CUCM User Roles
  • CUCM Multi-Level Administration
  • CUCM Device/Phone/Line User Association
  • UCCX and CUP Basic Users

Module 05 :: Call Features - In-Depth :: Runtime 5.3 hrs

  • SCCP and SIP Phone Display
  • Phone Firmware
  • Phone Logging
  • Ring Settings
  • Basic and Advanced Call Forwarding Display
  • Auto-Answer Options
  • CallBack (Camp-On)
  • Intercom
  • Advanced Call Hold Options
  • Call Park
  • Directed Call Park
  • Advanced Call Park Settings
  • Call Pickup
  • Group Call Pickup
  • Other Call Pickup
  • Directed Call Pickup
  • Call Pickup Attributes
  • Shared Line
  • Barge and cBarge (Conference Barge)
  • Privacy
  • Built-In IP Phone Bridge

Module 06 :: Media Resources - MTPs, Conf Bridges, Annunciator and Music on Hold :: Runtime 5.6 hrs

  • IOS Software MTP
  • IOS Conference Bridge
  • IOS Transcoding
  • Media Preference and Redundancy
  • Meet-Me Conferencing
  • Ad-Hoc Conferencing
  • Annunciator
  • Unicast Music on Hold
  • Traditional Multicast Music on Hold
  • Alternate Multicast Music on Hold

Module 07 :: Expert Gateways & Trunks :: Runtime 5.9 hrs

  • ISDN Switch Types and Advanced CNAM options
  • ISDN Information Elements
  • SIP Trunks - Fundamental and Advanced Options
  • H.323 Gateways - Fundamental and Advanced Options
  • MGCP Gateways - Fundamental and Advanced Options

Module 08 :: Expert H.323 Gatekeeper :: Runtime 7.1 hrs

  • Provisioning IOS H.323 Gatekeeper
  • Registering CUCM with H.323 Gatekeeper
  • Registering CUCME with H.323 Gatekeeper
  • Routing Calls from CUCME to CUCM via Gatekeeper in Multiple Zones with Dynamic E.164 Aliases
  • Routing Calls from CUCM to CUCME via Gatekeeper in Multiple Zones with Multiple Tech Prefixes
  • Routing Calls from CUCME to CUCM via Gatekeeper in Multiple Zones with Multiple Tech Prefixes
  • Routing Calls from CUCME to CUCM via Gatekeeper in Multiple Zones with Static E.164 Aliases
  • Routing Calls from CUCM to CUCME and Back via Gatekeeper in One Zone with One Tech Prefix
  • Gatekeeper Call Admission Control
  • Routing Calls from CUCM to CUCME and Back via Alternate Gatekeeper Clustering in Multiple Zones with Multiple Tech Prefixes using GUP

Module 09 :: Dial Plan - Line Device Approach and the Not-So-Basic Fundamentals :: Runtime 7 hrs

  • Class of Service: Calling Search Spaces and Partitions
  • Gateways, Route Groups, Local Route Groups/Device Pools
  • Route Lists and Standard Local Route Groups
  • Route Patterns and Translation Patterns
  • Digit Manipulation: Calling & Called Party Transformations and IOS Dial Peers
  • Private Line Automatic Ringdown (PLAR)

Module 10 :: Dial Plan - Globalization & Localization of both the Calling and the Called Numbers, and with Mapping the Global Number to the Local Variant :: Runtime 6.3 hrs

  • Inbound PSTN Calls (Ingress from PSTN, Egress to Phones): Calling Party Globalization :: GW Incoming Calling Party Settings
  • Inbound PSTN Calls (Ingress from PSTN, Egress to Phones): Calling Party Localization :: Phone Calling Party Transformations
  • Outbound PSTN Calls (Ingress from Phones, Egress to PSTN): Called Party Globalization :: PSTN Patterns - a.k.a. "Translation Patterns are the *New* Route Patterns"
  • Outbound PSTN Calls (Ingress from Phones, Egress to PSTN): Called Party Localization :: Digit Manipulation: Calling & Called Party Transformations and IOS Voice Translation Rules & Dial Peers
  • Mapping the Global Number to the Local Variant :: + Dialing and One-Button Missed Call DialBack

Module 11 :: Dial Plan - Unlocking the Full Potential of Globalization & Localization :: Runtime 4.2 hrs

  • System & User Speed Dials and Corporate Directory
  • Call Forward on Unregister
  • Automated Alternate Routing Made So Simple
  • Multiple Backup Gateways for Every Site using only Standard Local Route Group
  • National and International Tail End Hop Off (TEHO) Made Easy
  • Globalized Call Routing using H.323 ICTs

Module 12 :: Dial Plan - Cisco Unified Border Element (CUBE) with SIP Normalization :: Runtime 6.5 hrs

  • SIP Trunk to SIP ITSP for PSTN Call Routing
  • Conforming to ITSP Reqs: Various SIP-Attributes
  • Conforming to ITSP Reqs: SIP Header Conversions
  • Advanced Call Admission Control Mechanisms with CUBE
  • Skype SIP Trunk for Branch2 Site
  • Testing Supplementary Features

Module 13 :: Unified Mobility - Getting the Most out of Single Number Reach and Direct Inward System Access :: Runtime 6.8 hrs

  • Mobile Connect Basics
  • Mobile Connect Ring Schedule
  • Mobile Connect Localization
  • Mobile Connect Access Lists and Exclusivity
  • Mobile Connect Interaction with Local Route Group
  • Mobile Voice Access - Inbound Call Recognition and Display
  • Mobile Voice Access and Direct Inward System Access (DISA)
  • Mobile Connect Mid-Call Features - Supplementary Services

Module 14 :: Device Mobility & Extension Mobility - What They Have in Common and When To Use Each :: Runtime 4 hrs

  • Device Mobility - Between Sites but Within a Country
  • Device Mobility - Between Sites and Between Countries
  • Extension Mobility
  • Device and Extension Mobility and TEHO Interactions

Unified Communications Manager Express

Module 15 :: CUCME System and Phone Basics - SCCP and SIP :: Runtime 5 hrs

  • IOS DHCP
  • IOS Clock and Network Time
  • IOS TFTP Server
  • SIP CME Server Setup
  • SIP CME Phone and DN Setup
  • SCCP CME Server Setup
  • SCCP CME Phone and DN Setup
  • CME Directory Services
  • SCCP CME Server Redundancy
  • Endpoint Registration with External SIP Proxy Server
  • CME Templates for SCCP and SIP Phones and DNs
  • CME Phone Customization

Module 16 :: CUCME Dialplans, Class of Restrictions (COR) & Media Resources :: Runtime 5.6 hrs

  • PSTN Dialing from CME
  • IOS Voice Translation Rules in CME
  • Load Balancing Calls in CME
  • Class of Restrictions for CME
  • Multicast Music on Hold for CME
  • IOS Transcoding for CME
  • IOS Hardware Conference Bridge for CME
  • Multicast Broadcast-Paging in CME
  • MTP and DSP Resources in CME
  • Speed Dials in CME

Module 17 :: CUCME Advanced Call Features :: Runtime 4.2 hrs

  • Shared Lines and Feature Ring with SCCP Phones
  • Shared Lines with Barge and Privacy for SCCP Phones
  • Intercom for SIP and SCCP Phones
  • Night Service Bell for SCCP Phones
  • Call Park for SIP and SCCP Phones
  • Call Blocking for SIP and SCCP Phones
  • CallerID Blocking for SIP and SCCP Phones
  • Call Transfer and Forwarding for SIP and SCCP Phones

Module 18 :: CUCME Call Coverage and Survivable Remote Site Telephony :: Runtime 3.5 hrs

  • CME as SRST Fallback Mode (SIP, SCCP and MGCP Fallback)
  • Traditional SRST Fallback Mode (SIP, SCCP and MGCP Fallback)
  • 4-Digit Reachability from CUCM & SRST While in Fallback Mode
  • Call-Coverage - Call Pickup Groups
  • Call-Coverage - Basic-Automatic Call Distribution (B-ACD)
  • Call-Coverage - SIP and SCCP Hunt Groups and IOS Hunting

Unified Contact Center Express

Module 19 :: Unified Contact Center Express – Integration, CSQ Provisioning and Custom Scripting :: Runtime 5.1 hrs

  • UCCX Setup and Integration and Troubleshooting Being "Locked-Out"
  • CSQ Setup with Preferential Agent Choice
  • Basic Custom Scripting – Examination of UCCX Editor and “Simple Queuing.aef” Default Script
  • Basic Custom Scripting – Day of Week & Time of Day
  • Basic Custom Scripting – Reroute to Voicemail or Proceed to Queue
  • Basic Custom Scripting – MoH in Queue
  • Basic Custom Scripting – How Many Times Through Queue with Option to Go to Voicemail
  • Basic Custom Scripting – Nested Queues for More Available Agents
  • Basic Custom Scripting – Agent-Based Routing
  • Basic Custom Scripting – Testing and Debugging

Unified Messaging

Module 20 :: Messaging - Unity Connection & Unity Express :: Runtime 2.4 hrs

  • Unity Connection
  • Unity Express

Unified Presence

Module 21 :: CUCM Presence and Cisco Unified Presence Server - Integration and Client Usage :: Runtime 4 hrs

  • CUCM-Only Presence with Subscribe CSS
  • CUCM-Only Presence with Presence Groups
  • CUPS & CUCM Integration and CUPC Personal Communicator Provisioning
  • CUPS and IP Phone Messenger (IPPM)

And for those that couldn't get enough of this trailer for this series the first time, here it is again:

Sep
10

A while back, Cisco began consolidating all of the CCIE lab equipment used by candidates when sitting to write their practical lab exam. Most of the lab hardware now resides in San Jose, California US, with only the Storage and Wireless awaiting movement. While most of the CCIE tracks' practical lab examinations are able to be completely self-contained inside a single rack, the pesky Voice exam remains an abnormality with the need for hardware IP phones at the testing site where the candidate may sit for the exam.

Having the IP phones in a completely separate location -- and therefore seemingly an entirely different L3 IP subnet -- would seem to present a major challenge for candidates attempting to test certain configuration tasks such as Multicast Music on Hold, many QoS mechanisms, SRST, and even smaller things such as CDP discovery and DHCP. So how is Cisco able to get away with having phones at a remote location (5000 miles or more in some instances), and yet still allow candidates to configure and then properly test what they critically need to?

Layer 2 Tunneling - that's how. It's the only way to make phones "look" via CDP like they are connected to the same switch that they students are configuring on their rack - even if this couldn't be further from the truth (pun intended). This also allows everything else to function and work as perfectly as if the phone actually were on the same physical L2 wire, but alas, they most certainly are not.

Now to do this, it requires a good investment in a good number of additional routers and switches to make it all work, since after all there are a number of physical L2 LANs per candidate rack, and many more than simply one rack, not only in San Jose, but also at each remote testing site.

INE is very happy to announce that -- in both a strong desire to provide the exact same environment as the real lab, and also our response to listening to student requests from bootcamps this year -- we have put forth the necessary investment in both capital and man-hours to setup the necessary infrastructure to allow every Voice class that we run to be identical to taking a seat in the actual CCIE Voice lab exam. When you take a seat in any one of INE's Voice classes, that your 2 CorpHQ 7961 IP phones, your 1 Branch1 7961 IP phone, and your 2 Branch2 7961 IP phones will both appear to their respective site L2 switches, and function, just as if they were directly connected to your rack. It's as if we are now transporting each Voice rack of equipment with us no matter where we travel. Oh - did I mention that? This will be the standard no matter where you sit an INE Voice class - London, Seattle, Chicago, RTP, and anywhere else that we may have an on-site class requested and put together (some possibilities at the moment are Dubai, Bangalore, and Sydney).

On that note, we also just released our new training schedule with dates and expanded locations for 2011. Only dates through March are posted on the site at the moment, but the rest should be posted up either today or Monday (13 Sept).

So just to recap what you get when you train with INE for your CCIE Voice exam:

  • 5  7961 Hardware IP Phones on your desk in front of you, plus 1  7960 PSTN phone
  • Every 7961 IP phone functions exactly as it would as if it were physically and directly connected to its site's layer 2 switch (and just as it actually is in the real CCIE lab exam)
  • An instructor who has helped hundreds of people become CCIEs and who has been developing training materials and teaching the technologies needed to pass the CCIE Voice exam for over 5 years now

It is also worth updating everyone that we have released a few more CCIE Voice Deep Dive modules (scroll to bottom of linked page for each module details), and next week we will be completing two additional modules (both on different forms of Mobility) to round out our complete section relating to the CUCM server. So currently we are up to modules 1-12 and we have over 70 hours of class-on-demand style training with extremely detailed walk-throughs of every configuration and debug/trace in CUCM. Once we finish up the last two modules on CUCM, we will be moving onto CME, and you can expect the same level of detail on absolutely every aspect of that platform as well!

Happy Studies!

-Mark Snow

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