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Book Review: I'm the Boss of Me: A Guide to Owning Your Career by Jeanne Beliveau-Dunn

By David Holman
icon
I'm the Boss of Me:
A Guide to Owning Your Career
by Jeanne Beliveau-Dunn
iconI was asked if I'd like to review this book a few months ago.  At first I was hesitant.  It's not typically what I write about on VoiceCerts.com, but in the end I felt that it could have value to everyone here.  Written by a Cisco Systems Executive, I was curious how someone from my industry would approach career development.

So, I accepted a complimentary copy of the book in exchange for my honest review.  (standard disclaimer)

You are probably thinking, "oh boy, another self-help book".  Well, don't run away.  In order to be successful, we need to develop a cycle of continuous learning.  Perhaps you enjoy studying for technical certifications and soaking up as much technospeak as you can - that's great!   How will you navigate your career, though?  Do you have the soft skills to be successful?  We've all had different bosses along the way and each guided us in some way.  Love them or hate them, they all impact us in ways that we cannot control.  The only constant is YOU - and YOU are the focus of this book.  Congratulations, I bet you didn't know you were famous, did you?

Table of Contents


Chapter 1 - You Are Who You Are Because of Your Choices
Chapter 2 - You Are a Work in Progress - Challenge Yourself to Get Better
Chapter 3 - Do You Know What You Want from Your Career and Life?
Chapter 4 - Vision Strategy Execution
Chapter 5 - Becoming Your Vision:  Aligning Your Behavior
Chapter 6 - Becoming Your Vision II:  Articulating Your Brand and Building Your Bench
Chapter 7 - Attitude, Altruism and Altitude
Chapter 8 - Myths and Murders:  Getting Past Roadblocks and Barriers
Chapter 9 - Planning and Making Choices
Chapter 10 - Self-Leadership 2.0:  Success in the Face of Obstacles

Review


As I read this book, I really found myself doing a lot of self-discovery along the way.  As a 46-year old engineer, I've been fortunate to have had a wide range of experiences in my career, both good and bad.  At times, I've caught myself on auto-pilot just going through the motions (admit it, you've been there).  What I personally found valuable about this publication is how it presents a framework within which the author leads you on a productive trip of positive self development.

Who are you?  Where do you want to go?  How are you going to get there?  How do you identify career pitfalls?  What is your personal brand?  How do you choose supportive mentors in my "bench"?  How do you develop the soft skills necessary to be successful?

The author shares her proven strategies for addressing all of these questions while sharing many of her personal experiences along the way.  It's a light read that I believe everyone can find benefit within.  Personally, as I face my own midlife (crisis), I feel a greater sense of clarity now about who I am and where I am going.  Jeanne Beliveau-Dunn's career-building lessons are a useful addition to anyone's toolbox.

About the Author


Jeanne Beliveau-Dunn is a Chairman and COO of the IoT Talent Consortium, Vice President and General Manager, Cisco Systems.


Just Released - CIPTV1, 3rd Edition - 300-070 CCNP Collaboration Exam Learning Guide

By David Holman
icon
By Akhil Behl, Berni Gardiner,
Joshua Samuel Finke
From CiscoPress.com:icon

Now fully updated for Cisco’s new CIPTV1 300-070 exam Implementing Cisco IP Telephony and Video, Part 1(CIPTV1) Foundation Learning Guide is your Cisco® authorized learning tool for CCNP® Collaboration preparation. Part of the Cisco Press Foundation Learning Series, it teaches essential knowledge and skills for building and maintaining a robust and scalable Cisco Collaboration solution.

The authors focus on deploying the Cisco Unified Communications Manager (CUCM), CUCM features, CUCM based call routing, Cisco IOS Voice Gateways, Cisco Unified Border Element (CUBE), and Quality of Service (QoS).

They introduce each key challenge associated with configuring CUCM, implementing gateways and CUBE, and building dial plans to place on-net and off-net calls using traditional numbered dial plans and Uniform Resource Identifiers (URIs). They show how to implement conferencing and other media resources, and prepare you to apply QoS features for voice and video.

Each chapter opens with a topic list that clearly identifies its focus, ends with a quick-study summary of key concepts, and presents review questions to assess and reinforce your understanding. The authors present Cisco best practices, and illustrate operations and problem solving via realistic examples.

This guide is ideal for all certification candidates who want to master all the topics covered on the CIPTV1 300-070 exam.

The official book for Cisco Networking Academy’s new CCNP CIPTV1 course includes all new Learning@ Cisco CIPTV1 e-Learning course content:
  • Covers CUCM architecture, deployment models, and tradeoffs
  • Walks through bringing CUCM online, deploying endpoints, and setting up users
  • Explains how to create a solid IP Phone foundation for advanced services
  • Covers dial plan elements, design, and implementation
  • Reviews key call routing elements
  • Explains digit manipulation
  • Shows how to control user access
  • Discusses audio/video resources and videoconferencing
  • Covers QoS tools and preferential call handling
  • Explains external connections via Cisco IOS Voice Gateways and CUBE
  • Streamlines review with clear summaries, assessment questions, and objectives

Book Review: CIPTV2 (300-075) Implementing Cisco IP Telephony and Video, Part 2 - CIsco Press Foundation Learning Guide

By David Holman
CIPTV2, 300-075 Guide
Alex Hannah, Akhil Behl
Cisco Press
iconRecently, I wrote a post entitled, "Easily Upgrade CCNP Voice to CCNP Collaboration".  When I drafted that title, I was thinking about the number of steps involved to migrate a CCNP Voice to a CCNP Collaboration certification.  In my mind, this was indeed "easy" - it's just one single exam (300-075)!  I immediately received feedback that this exam was not "easy" in any way - in fact, it was difficult to study for it due to the lack (at the time) of a Cisco Press authorized self-paced learning tool.  There is a tremendous amount of information available - but how to pull it all together to really prepare effectively?
4.5 out of 5
Recommended
Self-Study Resource

So, like many others, I was eager for the release of the the official self-study foundation learning guide "Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2)" by Cisco Press on March 24, 2016.

I had already spent hours scouring the web to find resources that I thought would help me prepare for this exam (CIPTV2 300-075) based on the exam blueprint that had been released by Cisco.   However, even though I found tons of information on the topics listed, I still felt as though the blueprint was broad enough that it would be difficult to prepare fully for each topic listed.   Simply put, what will I be tested on?

Cisco Press states that "as an Authorized Self-Study Guide, this book fully reflects the content of Cisco's official CIPTV2 course."  As long as the 300-075 exam fully reflects the content of the official CIPTV2 course, I'll be in good shape.

Read on for my full review of "Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2)" in specific preparation to take the exam that migrates my CCNP Voice to CCNP Collaboration, 300-075. Also, to be fully transparent, this product was received complimentary as part of the Cisco Press Reviewer Program

Book Contents: Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2) 


Closely aligning with the stated blueprint of the 300-075 exam, the book dutifully covers the stated exam objectives with the following chapters:
  1. Cisco Collaboration Solution Multisite Deployment Considerations
  2. Understanding Multisite Deployment Solutions
  3. Overview of PSTN and Intersite Connectivity Options
  4. URI-Based Dial Plan for Multisite Deployments
  5. Remote Site Telephony and Branch Redundancy Options
  6. Cisco Collaboration Solution Bandwidth Management
  7. Call Admission Control (CAC) Implementation
  8. Implementing Cisco Device Mobility
  9. Cisco Extension Mobility
  10. Implementing Cisco Unified Mobility
  11. Cisco Video Communication Server & Expressway Deployment
  12. Deploying Users and Local Endpoints in Cisco VCS Control
  13. Interconnecting Cisco Unified Communications Manager and Cisco VCS
  14. Cisco Unified Communications Mobile and Remote Access
  15. Cisco Inter-Cluster Lookup Service (ILS) and Global Dial Plan Replication (GDPR)
  16. Cisco Service Advertisement Framework (SAF) & Call Control Discovery (CCD)
  17. Answers Appendix

Goals and Methods


To quote the publication, "The most important goal of this book is to provide you with knowledge and skills in Cisco Collaboration solution, with a focus on deploying the CUCM, Cisco TelePresence VCS, Cisco Expressway Series Solution and associated Cisco Collaboration solution features."

However, the obvious goal of the book is to prepare the reader to pass the CIPTV2 (300-075) exam as part of the CCNP Collaboration certification and it does so by teaching the reader to understand the topics so that the end result will be a more skill Cisco Collaboration professional.  I feel that this is a very important concept.  The true value of a certification is not the certification itself - but the knowledge obtained in the pursuit.   The book stays true to this tenet and as it steps through each chapter.

Chapter Test Question
Each chapter contains practice exercises on the topics and test questions at the end of each chapter. The test questions are an excellent way of determining if you've understood the information presented in the chapter.  Don't move on to the next chapter until you fully and completely understand the test questions!

Better yet, your learning is not confined by the information within the book either!  If you'd like to dig even deeper into a specific topic, there are online references listed at the end of each chapter for a truly deep dive.

Example Chapter References

Impressions


Much of the difficulty found within the 300-075 CIPTV2 exam can be explained by the large scope of material involved.  It's difficult and time consuming to gain depth in all topics.

Example Note Box
However, the book does an outstanding job of rendering the information down into dense nuggets of information that allows the reader to consume knowledge as efficiently as reasonably possible.  You won't find a lot of fluff, the information presented is done so for a reason.  To get maximum benefit from the book, I recommend that you've either previously learned the CIPTV1 material (or the similar in the CCNP Voice certification) from the CCNP Collaboration blueprint.

While this may seem subtle, the prolific use of the "Note" box is a personal favorite of mine and calls attention to and clarifies the most important concepts. Pay close attention to these boxes for important information that will likely be reflected in some manner on the official test.

You'll also find tons of the requisite technical illustrations and diagrams as well.  Logical flowcharts, examples of GUI screens and plenty of network diagrams are included throughout.

Example Illustration
From a purely selfish point of view, I needed to obtain the knowledge needed to pass the 300-075 exam for two reasons.  Firstly, I need to update my knowledge beyond what was presented on the CCNP Voice certification.  Secondly, I need to renew my certification and obtain the CCNP Collaboration certification.  I believe that the book succeeds in providing the necessary framework that I needed to meet both of my goals.

The material is logically presented, supported with visual aids where it makes sense and builds upon the CIPTV1 (and larger CCNP Voice) curriculum in a methodical manner.  The online references are a tremendous resource if you desire additional reinforcement.

Final Thoughts


I'm not a stranger to Cisco Press self-study books and have heavily used self-study in my learning approach for various tests and certifications over the years.  Alex Hannah and Akhil Behl do not disappoint and have successfully created the high quality resource that I expected.  This book is simply the best self-study material available to prepare for the CIPTV2 300-075 exam.

Utilize the written chapters, verify with the test questions and go deeper with the online references. This approach will guide you as you prepare to take the exam if you truly commit to learning the material, but more importantly, it will make you a better Cisco Collaboration professional.   Many tools are provided so that you can be successful in your pursuit.

My largest criticism of this product comes at the lack of a testing engine or any sample tests.  While the chapters contain sample test questions, I personally find great benefit from the iteration and validation available in previous Cisco Press products that have included a testing engine.

Finally, the big question.  Did the book succeed in helping me pass my exam?   Yes!  I passed the exam today on my 2nd attempt.  Make no bones about it - this is a tough exam.   (Perhaps the toughest Cisco exam I've ever taken)

Now that I've passed, I can give you some more details on the approach that I used.

  1. Read the CIPTV2 book from Cisco Press.  I'm not going to lie to you - It's not Stephen King or Dean Koontz.  There are times that I read a paragraph and then read it again.  And then again. Don't just gloss over the words without serious contemplation of what they mean.  If you say to me, "but Dave, I'm already solid on some of these chapters and time is short", then you need to focus hard on your weaknesses.  Read those chapters thoroughly - they build a great foundation to build upon.  
  2. Next, you need to get online and read the documents from which Cisco is likely to pull exam questions.  I suggest the following for each major technology on the exam.   This may seem a bit like spitting into the wind, but you can't expect all exam questions to come from the Cisco Press literature or the official CIPTV2 class.  
  • Implementation Guides
  • Configuration Guides
  • TechNotes
  • SRNDs
  • Administration Guides
  • Troubleshooting Guides
  • Don't overlook the online references provided in the book at the end of each chapter - these are gold!
  • Finally, get your hands dirty in a lab, if you have that available.  If not, Cisco's dCloud could be an option.  If that is not an option, you can find videos in various places to help get more exposure to the technologies involved.  
  • Easily Upgrade CCNP Voice to CCNP Collaboration

    By David Holman
    icon
    CIPTV2 by Alex Hannah
    iconI've noticed that a lot of people with the CCNP Voice certification do not entirely understand how it relates to the CCNP Collaboration certification or how it can be used to easily obtain the CCNP Collaboration certification.

    Cisco has thankfully provided a migration path that let's the effort used to get your CCNP Voice not go to waste.  It's straightforward and only 1 test!  So, don't wait and slap a new paint job on that slightly dull CCNP Voice and drive off the lot with a new CCNP Collaboration!

    If you didn't complete the CCNP Voice certification previously, check out the CCNP Collaboration Migration Tool.  You may be surprised to see that it's not as hard as you thought!

    Why get the CCNP Collaboration Certification if I have CCNP Voice?


    The CCNP Voice certification is categorized as a "Retired Certification" by Cisco.  It does not encompass all of the current technologies that are included in the CCNP Collaboration certification and as a result does not hold the value that the CCNP Collaboration certification contains.

    If you have the CCNP Voice certification (and it's not expired), then you are in luck.  There is a fairly easy path to obtain the CCNP Collaboration, which is one of Cisco's currently available CCNP certifications.   Keep reading for details...

    Ok, so how do I upgrade my CCNP Voice to a CCNP Collaboration?


    First, make sure that your CCNP Voice certification is not expired.

    Next, pass 300-075 CIPTV2 (Implementing Cisco IP Telephony and Video, Part 2) and you will have earned your CCNP Collaboration certification!   Now that was easy, right?   BAM!

    Read my full review of the Cisco Press Official CIPTV2 Self-Study Foundation Guide (300-075).

    Cisco Certification Tracker - CCNP Collaboration


    What are the exam topics in 300-075 CIPTV2?


    In short, this exam covers several topics related to video and features added to CUCM 9.x or later.

    Per Cisco, "The Implementing Cisco IP Telephony & Video, Part 2 (CIPTV2) v1.0 exam is a 75 minute 55-65 question assessment that tests candidates seeking CCNP Collaboration on their ability for implementing a Cisco Unified Collaboration solution in a multisite environment. It covers Uniform Resource Identifier (URI) dialing, globalized call routing, Intercluster Lookup Service and Global Dial Plan Replication, Cisco Service Advertisement Framework and Call Control Discovery, tail-end hop-off, Cisco Unified Survivable Remote Site Telephony, Enhanced Location Call Admission Control (CAC) and Automated Alternate Routing (AAR), and mobility features such as Device Mobility, Cisco Extension Mobility, and Cisco Unified Mobility. The exam also describes the role of Cisco Video Communication Server (VCS) Control and the Cisco Expressway Series and how they interact with Cisco Unified Communications Manager."


    Exam Topics - 300-075 CIPTV2
    Check out this detailed PDF for the full list of 300-075 exam topics.


    Study resources for 300-075 CIPTV2?

    1. Cisco Press Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2)
    2. Cisco Collaboration Solutions Design Guidance <- Great Top Level Collection
    3. Cisco CIPTV2 Study Material
    4. Cisco Collaboration Study Group @ The Cisco Learning Network
    5. Cisco Collaboration 9.x Solution Reference Network Designs
    6. Cisco TelePresence VCS Configuration Guides
    7. Cisco TelePresence VCS Document Collection
    8. H.320 Gateway to H.323 Gatekeeper Video Call Flow

    Call Classification: OnNet vs OffNet

    By David Holman
    Call Classification in Cisco Unified Communications Manager (CUCM) is the practice of labelling a call to be either "OnNet" or "OffNet".   This is also a topic that you should understand well if you are pursuing CCNA Collaboration or CCNP Collaboration certifications.  You could go into quite a bit of depth when researching the various features and interactions related to Call Classification, but my goal with this post is provide some useful reference material and get your studies off in the right direction.

    Why is Call Classification important?   It has direct impact on the features listed below:

    • Directory Number Call Forward Settings:  Forwarding paths are configured differently for OnNet calls (Ex., Forward Busy Internal) and OffNet calls (Ex., Forward Busy External).  Internal calls can be forwarded to a manager or team member while OffNet calls are preferred to forward to voicemail when the party is unavailable.
    • Drop Conference When No OnNet Party Remains:  A toll fraud feature, this requires that an internal party be on a conference to ensure that the conference stays open.  
    • Toll Fraud Control (Block OffNet to OffNet Transfer):  Another toll fraud feature, this prevents you from forwarding your friend's calls to their family member in Costa Rica.  I'm sure that she's a nice lady, but your company shouldn't be paying for your friend's calls.  

    Call Classification Configuration:

    1. Route Pattern Configuration, Cisco Unified Communications Manager Administration Guide
    2. Gateway Configuration, Cisco Unified Communications Manager Administration Guide
    3. Trunk Configuration, Cisco Unified Communications Manager Administration Guide
    4. Service Parameter Configuration, Cisco Unified Communications Manager Administration Guide
    5. Conference Bridges, Cisco Unified Communications Manager System Guide

    Additional interactions related to Call Classification:

    • Call Data Records (CDR):  Many reports will break data down by OnNet/Internal or OffNet/External calls.   The value of these reports will depend upon understanding exactly what types of calls could contribute to each sum.  
    • Dialed Number Analyzer (DNA):   DNA will reference the call classification when possible and is a useful tool for troubleshooting and testing various call flows.  
    • Bulk Administration:   Call Classification is a configurable value when using this tool to administer trunks and gateways.

    Definition:  OnNet vs OffNet


    "OnNet" typically refers to calls made within your own network of telephony devices.   Within a CUCM cluster of devices, this typically means calls between devices that are configured by the cluster itself.   

    "OffNet" conversely refers to calls made to devices outside of your local network of devices.  For example, if you are calling from your cluster to any external PSTN device (like a cell phone) - it's an OffNet call.

    So, somebody is going to ask - "But what if my company has multiple clusters?"  That's a very good question!  The trouble with this question is that it depends!  Calls between clusters could be either OffNet or OnNet depending on how you've designed the telephony network.  

    Perhaps, these are entirely different business units that function quite independently of one another?  I would love to have some readers share their experiences on this topic in the comments below. 

    Toll Fraud Considerations


    Drop Conference When No OnNet Party Remains


    The system drops the active conference when the last on-network party in the conference hangs up or drops out of the conference. Cisco Unified Communications Manager releases all resources that are assigned to the conference.

    Be thoughtful when enabling this feature!  To use the additional functionality that advanced ad hoc conferencing provides, Cisco recommends that you set this service parameter to "Never." Any other setting can result in unintentional termination of a conference.

    Block OffNet to OffNet Transfer


    The Cisco Unified Communications Manager clusterwide service parameter "Block OffNet to OffNet Transfer" allows administrators to prevent users from transferring external calls to another external number. This parameter specifies values as "True" or "False". 

    Setting the parameter to "True" blocks external calls from being transferred to another external device. The default value specifies False. You modify the "Block OffNet to OffNet Transfer" service parameter by using the Service Parameter Configuration window.

    When a user tries to transfer a call on an OffNet gateway or trunk when the service parameter "Block OffNet to OffNet Transfer" is set to "True", a message displays on the user phone to indicate that the call cannot be transferred.

    Cisco UC Call Recording with MediaSense

    By David Holman
    Cisco's MediaSense, introduced in 2011, is their solution for multimedia capture, streaming and recording.

    This article provides a brief overview of the solution, describes the MediaSense deployment models and gives information on how to improve upon the free user interface that Cisco provides with MediaSense at no additional cost.  Cisco also provides extensive design, installation and upgrade, configuration, maintenance and troubleshooting resources online.  While this article is focusing specifically on call recording and the "Search and Play" interface, all characteristics and features of MediaSense can be explored in the design guide.

    Let's start with some call recording history first…

    History of Cisco UC Call Recording


    Before 2011, the only way to record phone calls in the Cisco IP telephony environment was to use one of several 3rd party products. Most of these solutions used “SPAN-based” recording.  When SPAN (Switched Port Analyzer) is enabled on a switch port (or VLAN), the switch copies all network packets to another port where a recording solution detects the VoIP packets stores them as audio files.

    Some also supported a more efficient "audio forking" approach provided by Cisco through the use of bridge functionality provided within the IP phone itself.  Most Cisco IP phones contain the embedded conference bridge (“built-in bridge or BiB”), which, with proper CUCM configuration, will fork the audio streams of the phone call (for example, one for the agent voice and one for the customer voice) to the recording server.

    Built-in-Bridge (BiB) and SPAN Recording


    Introduction of Cisco MediaSense


    Near the end of 2010, Cisco entered the call recording market and performed 3 great steps:

    1. Cisco introduced the MediaSense product that captures audio-streams duplicated by Cisco IP phone built-in bridge;
    2. At the same time Cisco released the version 15.2 of IOS (software that runs on Cisco routers) with media forking feature that duplicates the phone call RTP streams to Cisco MediaSense for recording;
    3. Cisco positioned MediaSense as the recording platform with API to 3rd party apps.  MediaSense works on the network layer capturing and storing the phone calls but you need to turn to 3rd party vendors for business applications that work with call recordings, like speech analytics, quality management, agent training, etc.
    Cisco MediaSense now supports the capture of media streams forked via:

    • Cisco IP phones (models with built-in bridge) – to record employees’ Cisco phones;
    • Cisco Unified Border Element (application run on Cisco IOS) – to record 3rd party SIP devices and external-to-external phone calls.
    Cisco MediaSense Recording


    Unfortunately, from the users’ perspective, MediaSense only provides a simple “Search and Play” web-interface that is not as feature-rich as most desire:

    Cisco MediaSense Search-n-Play User Interface


    The "Search and Play" interface is a free web-application provided by Cisco (with the source code) as an example of how to use the Cisco MediaSense API – this password-protected web-page allows user to perform basic search and play call recordings.

    Cisco MediaSense API Enables Innovation


    As a result of the API that Cisco makes available, most call recording software vendors now support integration with Cisco MediaSense to provide clients with more convenient user interfaces to access the call recording archive with features like:

    • search recordings by date, time, phone number, employee’s name and client name;
    • indication of a call direction (incoming/outgoing);
    • logical grouping of records in case of call routing, putting on hold or parking and organizing a conference with several participants;
    • email notification about new recorded call;
    • search and playback interface on Cisco IP phone;         
    • option to playback recordings during a phone call (all call participants will hear the recording).

    Free Call Recording UI For Cisco MediaSense


    To provide additional features beyond what Cisco provides "out of the box" with MediaSense, one Cisco Solution Partner offers a free interface to Cisco MediaSense that contains all of the features listed above.  

    Additionally, their commercial “Aurus PhoneUP” application suite contains both a Cisco MediaSense interface and a complete call recording solution for Cisco UCMand BE 6000/7000.

    Most Useful Cisco UC Resources (A Crowdsourcing Effort)

    By David Holman
    We all have our own bookmarks, favorites and "hidden Internet gems" that we have found indispensable in doing our jobs.  I'm calling out to all of the voice engineers to share what you find to be the most useful resources on the net!

    Just share in a comment and I'll categorize below.  Thanks ahead of time for your contribution!

    All Unified Communications Products

     

    Call Control

    1. Cisco Collaboration Systems Documentation : Full matrix of systems documentation back to UC version 6.1.  SRND, configuration notes, system release notes, etc.
    2. Cisco UC Integration for Microsoft Lync 9.7(4) Administration Guide :  Detailed guide for integration of Lync into Cisco UC.  Don't attempt without using this guide!

    Communication Gateways

    1. Cisco IOS Voice Troubleshooting and Monitoring Guide : Very comprehensive list of Cisco troubleshooting guides that covers all aspects of IOS Voice.  If you aren't sure where to start - Start Here!
    2. Voice Translation Rules : If you don't make voice translation rules frequently, the syntax can be impossible to remember.  Consider this page one of your best friends.
    3. Cisco IOS 15.0M Configuration Guides : Extensive list of configuration guides specific to IOS 15.0M broken into many categories, including "Voice and Video", "QoS" and "Security, Services and VPN". 

    Unified Communications Applications

    1. Cisco Unified Communications Manager IM & Presence Configuration Guides : Good starting point when doing any configuration for IM & Presence.
     

    Collaboration Management and Licensing

    1. Cisco Prime Collaboration End User Guides : Versions 9.0 through current (10.x)

    Telephony Extensions

    1. Cisco Emergency Responder DocWiki : Lots and lots of troubleshooting guides to assist with a variety of CER-related issues.

    Voice Servers

    1. Cisco MCS-78XX Boot Error Codes : Quick list of common boot errors that can happen when booting 78XX hardware.

    Virtualization

    1. DocWiki - Virtualization for Cisco Unified Communications Manager (CUCM) : This is the motherload starting page for all things related to virtualizing CUCM.  Contains VM configuration requirements all versions back to CUCM 8.0(2).  

    Miscellaneous

    1. Recognizing and Categorizing Symptoms of Voice Quality Problems : This is the document you wish every user would read before they report that their voice quality is "bad".  Defines a vocabulary that can be used to describe quality problems AND includes sound recordings to demonstrate each variation.

     

    Essential Voice Engineer Tools

    1. TranslatorX : A troubleshooting tool that allows you to quickly parse through Cisco Unified Communications Manager or Cisco Unified Border Element trace files and search for Q.931, H.225, SCCP (Skinny), MGCP, or SIP messages.[submitted by Nick Barnett]
    2. Notepad++ : A free source code editor and Notepad replacement that supports several languages.[submitted by Nick Barnett]
    3. Agent Ransack : A free software program for finding files on your PC or network drives.[submitted by Nick Barnett]
    4. WinMerge : An Open Source differencing and merging tool for Windows. WinMerge can compare both folders and files, presenting differences in a visual text format that is easy to understand and handle.[submitted by Nick Barnett]
    5. Wireshark : The world's foremost network protocol analyzer. It lets you see what's happening on your network at a microscopic level. It is the de facto (and often de jure) standard across many industries and educational institutions. [submitted by Nick Barnett]

      CCIE Collaboration Written Self-Evaluation Tool

      By David Holman
      Self-Evaluation Tool
      As you study for the CCIE Collaboration Written exam, determining "what" to study can be daunting sometimes given the number of items listed in the blueprint.  I mean, where do you start?

      I've always thought that you should target your weakest CCIE Collaboration topics first.  That way, you'll never be completely clueless about a given question (or task) on the lab.  But, how do you determine what your weak topics are?  And how do you know when your weakest topics aren't the weakest anymore?


      Self-Evaluation - How Prepared Am I for the CCIE Collaboration Written?


      I've created the spreadsheet below to help you grade your progress on each topic as you progress through your studies.  Feel free to copy it and use it for your own purposes.  


      Download Version 2 - CCIE Collaboration Written Self-Evaluation Tool

      Note on Version 2: I've uploaded a new version of the spreadsheet that makes it easier to read and improves the mastery calculations (using each sections weighted importance).  The updated calculations should give you a more accurate measure of your knowledge level and readiness for the exam based on the section weights that Cisco has assigned.  Also, values in underlined red are averages of all subtopics in that group - formulas are protected to keep you from overwriting accidentally.

      Within the spreadsheet, you are prompted to grade your knowledge level for each subtopic with a value from zero through five.  Zero represents no knowledge at all, while five indicates subject-matter expert level knowledge.  In my mind, it's actually quite difficult to achieve a value of five as I reserve that for truly elite knowledge - are you really a SME for this topic?  Be honest!

      There's no magic value that says - "You're now ready to take the exam!"  Instead use this tool as a way to supplement your other study resources and perform some very granular self-evaluation to assist in determining topics that need additional attention before taking the CCIE Collaboration Written exam.

      CCIE Collaboration Written Self-Evaluation Tool
      CCIE Collaboration Written Self-Evaluation Tool

      Suggested Mastery Levels:
      • 0  -  No knowledge or experience at all
      • 1  -  Junior Engineer - no hands-on experience
      • 2  -  Junior Engineer - some hands-on experience, basic ability to configure
      • 3  -  Engineer - moderate experience, average configuration and troubleshooting
      • 4  -  Senior Engineer - extensive experience,  advanced configuration and troubleshooting
      • 5  -  Senior Engineer - subject matter expert, elite configuration and troubleshooting ability

       

      Section Mastery Levels


      I believe scoring a 4 or 5 on any given subtopic requires extraordinary study, experience and effort.  Personally, I reserve these scores for true strengths that I feel separate myself from my peers.  Remember, achieving a "five" in a subtopic is reserved for true subject mastery.  That said, anything above 60% mastery in a given topic is exceptional! 

      The far right of each row in the spreadsheet calculates a value called "Total Mastery Points" using your relative mastery levels for each blueprint section (and the importance given to the section by Cisco).  Remember, there's no magic number here to show that you are ready for the exam.  Instead I suggest that you instead re-evaluate yourself every couple of weeks and use this value as an indicator of your progress towards the ultimate goal.  Good luck!

        CCIE Collaboration Written Self-Evaluation Tool Mastery Points
        CCIE Collaboration Written Self-Evaluation Tool Mastery Points

        Final Thoughts

        I hope that this tool helps you focus on your studies.  I know that I've spent way too much time tweaking it and fixing formulas, etc.!  If you have any additional ideas on how to make this a better tool - or if you have one of your own that you'd like to share - please use the comments below!

        14 Fun Facts about CUCM 10.x Hunt Pilots and Native Call Queuing

        By David Holman
        she loves call queuing
        She Loves Call Queuing
        Lately, I've had a lot of talk about how the new features in CUCM 9.x/10.x related to hunting and native call queuing can enhance user experience within existing call flows.  But, nothing in life is free.

        Cisco provides many limits and recommendations related to the use of hunt pilots and native call queuing that could potentially blow your super-fantastic-end-of-the-world design out of the water.  Much of what is listed below would not be learned by just playing around with the new call queuing feature - it's very much worth the time to read through the "additional information" link at the bottom of this post.

        Below is a list of the fun facts related to native call queuing and hunt pilots in CUCM 10.x that I've been able to gather thus far.


        14 Fun Facts about CUCM 10.x Hunt Pilots and Native Call Queuing

        1. Call queuing is not supported with broadcast algorithm hunt lists.
        • This makes sense in the conventional sense of a contact center that typically delivers the call to the longest available user.  
        • The calls are required to be distributed to individuals based on the availability status of the user using a top down, circular or longest available algorithm.
      • You can configure a maximum of 25 hunt pilots per hunt list in Call Queuing. If you exceed this limit, the queue status will not be displayed.
        •  Plenty for 99% of the applications out there.  I suppose you could find a place where this is a limiting factor - but then again, this isn't a replacement for UCCE.
      • If no line members answer a call, then that caller will not be placed in queue. The call is routed to a new destination, or disconnected, based on the setting under "When no hunt members answer, are logged in, or registered" .
        • I assume that if a single line member does not answer, it will re-queue the caller in the way that a typical RONA would usually be handled. 
        • I also assume that the maximum wait time for the hunt pilot is the ultimate trigger here.
      • Calls will be placed in queue only if all members are busy.
        • Makes sense, if they aren't busy they should be able to take the call immediately.
      • If a line member does not answer a queue-enabled call, that line member is logged off the hunt group only if the setting "Automatically Logout Hunt Member on No Answer" is selected on the line group page.
        • This is a good way to limit the impact to callers of someone that has walked away from the desk or left for the day in an "available" state.  I recommend that "Automatically Logout Hunt Member on No Answer" always be checked and ensure that users are trained quite clearly on how this works.
      • Queuing-enabled hunt pilot calls can only be received by line members one call at a time. Two queuing-enabled hunt pilot calls cannot be offered to a line member (no matter what the busy trigger is set to). This does not limit a line member to only receive calls directly to their DN or from non-queuing hunt pilots.
        • On the surface this may seem like a limitation, but in a contact center-like environment you wouldn't want to have a line member bothered with multiple calls at the same time (typically).  
        • This is one for the lab, but I wonder if non-queuing-enabled hunt pilots could deliver calls to a line member if they've already taken a queuing-enabled call (and their busy trigger will allow it).
        • Furthermore, could a non-queuing hunt pilot configured to broadcast all users be used as an overflow for when a queuing hunt pilot did not find an answer? 
      • If the hunt list has multiple line groups then these line groups need to have the same setting for Automatically Logout Hunt Member on No Answer.
        • Unsure what happens if you don't comply, but I can see this being a common configuration error.
      • All Hunt options need to be set to Try Next Member, then Try Next Group in the hunt list. 
        • Again, unsure what happens when you don't comply.
      • H323 Fast Start does not support Call Queuing. 
        • I'll need to find more clarification on this.  It can have a bearing if you have call handling groups located in various spokes from the CUCM hub.  Will it still deliver calls, just not in fast start mode?
      • Cisco Unified Communications Manager does not support Unified Mobility with Call Queuing.
        • This policy seems to have been done with the caller in mind.  Remote team members can be reached mobile using more conventional coverage path techniques.
      • A single Unified CM Subscriber supports a maximum of 100 hunt pilots with call queuing enabled per node.
        • This is both a limiting factor and a management headache in very large hub-spoke enterprises.  Requires being very mindful of how hunt pilots are spread throughout the cluster.
      • The maximum number of simultaneous callers in queue for each hunt pilot that you can configure ranges from 1-100 (default 32).  
        • Should be more than enough for most environments.  If you're pushing this limit, consider the more conventional contact center products.
      • The maximum wait time in queue for each hunt pilot that you can configure ranges from 0-3600 seconds (default 900). 
        • A maximum of one hour in queue is for the truly determined customer.
      • An increase in the number of hunt lists can require you to increase the dial plan initialization timer that is specified in the Unified CM service parameters. Cisco recommends you to set the dial plan initialization timer to 600 seconds if you have 1500 hunt lists configured.  
        • Very large clusters beware.

        How to Configure Native Call Queuing in CUCM

        This great video from Paul Stryer, Cisco Architecture Systems Engineer, does a great job illustrating how to configure CUCM's native call queuing feature.  He also demonstrates with test calls in his lab, which really help to reinforce the concepts.



        For additional information about how CUCM Native Call Queuing functions, I highly recommend you check out the limitations outlined in the Features and Services Guide for Cisco Unified Communications

        CUCM (BE) SQL Query: Exposing Call Forwarding History

        By David Holman
        So, you're spending a perfectly good business day browsing the latest Apple rumor websites when your phone rings.  It's Anthony from "Flying Saucer Pizzeria" across town - and he's not happy.  He reports that he's been getting random calls from your company's main phone number.  The troubling part of this report is that Anthony says that he's talking to confused customers that called your company and don't know why they are talking to the best Pizza joint (555-9000) in town.  Perhaps karma?  No, not likely.  Though, pizza does sound pretty good right now!

        Instead, like a good voice engineer, you reference the CDR records on your CUCM server and find out that someone has forwarded their line (1000) to the pizza shop.  Perhaps, they thought this would be funny?  You take a quick look at the DN configuration in CUCM and no longer find the evidence you wanted.  It's not call forwarded?

        What to do now?  Well, it turns out that (the Cisco BE version of) CUCM maintains a list of the last five numbers to which a DN was forwarded AND when each was configured!  To my knowledge this information isn't available via the GUI, but you can get to it via a well-formed SQL query.   

        The Cisco Business Edition (BE) CUCM database has a table called "callforwardhistorydynamic".  See information about the fields contained within this table at the bottom of this post.  Pay close attention to the "Description" in the Data Dictionary Table - if it begins with "Cisco BE", it will only be available in the Cisco BE version.  There are exactly 28 tables in the 8.6(1) version of the Data Dictionary that are specific to the Cisco BE version.  In the 9.1(1) CUCM Data Dictionary, the number of tables specific to Cisco BE has grown to 40!

        Let's get right to the fun part! 

        CUCM SQL Query Example:  Exposing Call Forwarding History


        Query Composition

        select n.dnorpattern, cfhd.dnorpattern, cfhd.datetimestamp
        from numplan as n
        inner join callforwardhistorydynamic as cfhd on cfhd.fknumplan=n.pkid
        where n.dnorpattern = 1000


        Syntax

        From the CLI of the CUCM, enter the command below.

        admin:  run sql select n.dnorpattern, cfhd.dnorpattern, cfhd.datetimestamp from numplan as n inner join callforwardhistorydynamic as cfhd on cfhd.fknumplan=n.pkid where n.dnorpattern = 1000

        Output

        dnorpattern dnorpattern datetimestamp
        =========== =========== =============
        1000        5551900     1397782752
        1000        5559000     1397786930  <-
        1000        5551213     1397786943
        1000        5559000     1397786952  <-
        1000        5551215     1397786961


        This output proves it!  The DN was forwarded to the pizza joint on two different occasions!

        Interpretation

        First column:  The DN being targeted, in this case, "1000".
        Second column:  The number to which 1000 was forwarded.
        Third column:  The date and timestamp when the forwarding was done.

        Translating datetimestamp

        1. Launch Microsoft Excel.
        2. In cell A1 type the datetimestamp from column 3. 
        3. In cell A2, paste the formula =A1/86400+DATE(1970,1,1)
        4. Right-click on cell A2 and select format cells.
        5. Under the Number tab select Time where the format is 3/14/98 1:30 PM.
        6. The result is the actual time in readable format (Universal Time)
        1397786930 4/18/14 2:08 AM
        1397786952 4/18/14 2:09 AM

         

        CUCM 8.6 Data Dictionary Table:  callforwardhistorydynamic

        callforwardhistorydynamic from CUCM 8.6 Data Dictionary

        Additional CUCM SQL Information


        CUCM SQL Series: A Series:  From UCGuerrilla.com, this is really a fantastic starting point for anyone trying to get into CUCM SQL and understand how the tables are structured.
        CUCM SQL Series: Listing Call Forward All Settings:   From UCGuerrilla.com, great examples of how to query call forward all values from the database.
        Running SQL Queries on CUCM 6x/7x Using AXL SOAP Toolkit Part 1:  From NetCraftsmen.net, this article has a wonderful explanation of the table structures in the CUCM database and the different ways that they relate to one another.  I can't give @ucguerrilla enough credit, his articles on CUCM SQL queries are the best on the net!

        User Management - CCIE Collaboration Written Study Guide

        By David Holman
        ccie collaboration blueprint topic
        This post will serve as a study guide for the specific CCIE Collaboration Written blueprint topic, User Management, from the larger blueprint section, Cisco Collaboration Infrastructure.


        [Full CCIE Collaboration Written Blueprint]

        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

        Collaboration 9.x SRND Highlights: User Management


        This section contains information published within the Cisco Collaboration 9.x SRND (and is therefore not my own work).  I feel that it is very likely that questions on the CCIE Collaboration written exam related to "User Management" will have direct references to the linked, quoted and paraphrased information below.  It is highly suggested to read the actual SRND document closely as you prepare for the CCIE Collaboration Written Exam.

        LDAP Directory Integration (28 pages, Cisco Collaboration 9.x SRND)

        Directories are specialized databases that are optimized for a high number of reads and searches, and occasional writes and updates. Directories typically store data that does not change often, such as employee information, user policies, user privileges, and group membership on the corporate network.

        Directories are extensible, meaning that the type of information stored can be modified and extended. The term directory schema defines the type of information stored, its container (or attribute), and its relationship to users and resources.

        The Lightweight Directory Access Protocol (LDAP) provides applications with a standard method for accessing and potentially modifying the information stored in the directory. This capability enables companies to centralize all user information in a single repository available to several applications, with a remarkable reduction in maintenance costs through the ease of adds, moves, and changes.
        Various Requirements for Directory Integration
        Various Requirements for Directory Integration

        By default, all users are provisioned manually in the publisher database through the Unified CM Administration web interface.

        Cisco Unified CM has two types of users:
        • End users — All users associated with a physical person and an interactive login. This category includes all Unified Communications users as well as Unified CM administrators when using the User Groups and Roles configuration (equivalent to the Cisco Multilevel Administration feature in prior Unified CM versions).
        • Application users — All users associated with other Cisco Unified Communications features or applications, such as Cisco Attendant Console, Cisco Unified Contact Center Express, or Cisco Unified Communications Manager Assistant. These applications need to authenticate with Unified CM, but these internal "users" do not have an interactive login and serve purely for internal communications between applications.

        Unified CM provides an LDAP Query Filter to optimize directory synchronization performance. Cisco recommends importing only those directory user accounts that will be assigned to Unified Communications resources in each individual cluster. When the number of directory user accounts exceeds the number supported for an individual cluster, filtering must be used to select the subset of users that will be associated on that cluster. The Unified CM synchronization feature is not meant to replace a large-scale corporate directory.

        In many cases, a unique search base is all that is needed to control which accounts are synchronized. When a unique search base is not available, a custom LDAP filter might be required. When any mechanism is used to limit the accounts imported into Unified CM, the default directory lookup configuration will list only those directory entries that exist in the Unified CM database.
        Default Behavior for User-Related Operations for Unified CM
        Default Behavior for User-Related Operations for Unified CM

        Video Tutorial(s): User Management


        Active Directory/LDAP Integration for UCCaaS Overview (Anson Garcia)


        CUCM LDAP Integration (academytech)

        If you'd like to suggest a useful video, please use the comment section below.

        Additional Resources: User Management


        If you'd like to suggest additional resources, please use the comment section below.

        Final Thoughts: User Management


        This blueprint topic has many possible interpretations.  It could be applied to many different features throughout all of the UC Collaboration space.  For example, BOTH managing and integrating the corporate user directory AND configuring aspects of a user within CUCM/CUC are valid possibilities.  However, it does fall under the larger blueprint category of "Cisco Collaboration Infrastructure" which seems to exclude items such as user configuration and customization.

        As such, I've decided to try to focus on the most likely interpretations of what this could mean in the blueprint and subsequently land on the written exam.  The topic of "LDAP Directory Integration" does not seem to obviously fall under any other blueprint topic, so I'm going to believe that the blueprint authors intended this to fall under the larger umbrella of "User Management".

        There are "additional considerations" for Microsoft Active Directory for both Synchronization and Authentication.  These feel like prime targets for questions that could be on the CCIE Collaboration Written exam, I'd recommend being especially familiar with these sections.

        Ultimately, if you think this topic means something different or a major piece is notably missing from the description - please use the comment section below.

        Take your time and remember that anything in the SRND is fair game for the written exam!

        UC Deployment Models - CCIE Collaboration Written Study Guide

        By David Holman
        ccie collaboration blueprint topic
        This post will serve as a study guide for the specific CCIE Collaboration Written blueprint topic, UC Deployment Models, from the larger blueprint section, Cisco Collaboration Infrastructure.

        [Full CCIE Collaboration Written Blueprint]

        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

        Collaboration 9.x SRND Highlights: UC Deployment Models


        This section contains information published within the Cisco Collaboration 9.x SRND (and is therefore not my own work).  I feel that it is very likely that questions on the CCIE Collaboration written exam related to "UC Deployment Models" will have direct references to the linked, quoted and paraphrased information below.  It is highly suggested to read the actual SRND document closely as you prepare for the CCIE Collaboration Written Exam.

        Collaboration Deployment Models (50 pages, Cisco Collaboration 9.x SRND)

         Typical UC Deployment Models


        Where the Unified Communications and Collaboration services, their associated endpoints, gateways, border controllers, media resources, and other components are all located on a single high speed LAN or MAN.
        Campus UC Deployment Model


        Where the Unified Communications and Collaboration services are located in a central campus site or data center, but the endpoints, gateways, media resources, and other components are distributed across multiple remote sites that are interconnected by a QoS-enabled WAN.
        Multisite Centralized Processing UC Deployment Model


        Where multiple campus and/or centralized deployments are interconnected over a QoS-enabled WAN.
        Multisite Distributed Processing UC Deployment Model

        Video Tutorial(s): UC Deployment Models


        If you'd like to suggest a useful video, please use the comment section below.

        Additional Resources: UC Deployment Models


        If you'd like to suggest additional resources, please use the comment section below.

        Final Thoughts: UC Deployment Models


        This is a very large topic from the SRND and you can be certain that you will get at least one question about it on the CCIE Collaboration Written exam, probably multiples.  Fifty pages in the Collaboration 9.x SRND are dedicated to this one singular topic.  My advice when approaching this topic is to review the SRND section above very closely and look for small details that you may have not have remembered from CCNP Voice studies.

        Also, be on the lookout for details that may have changed in this 9.x version of the Collaboration SRND.  I didn't go back and compare previous versions of the SRND, but it feels to me that there is a stronger emphasis on virtualized hardware deployments.  Also, notice the additional use of terminology when describing the basic deployment models: it is also known as Single Site, Multisite/Centralized and Multisite/Distributed.

        Take your time and remember that anything in the SRND is fair game for the written exam!

        First Attempt with SQL Commands: Search and Change Speed Dials via CUCM CLI

        By David Holman
        Run SQL
        Ok, so I recently was asked how to locate all devices in a cluster that had a specific speed dial number configured.  It was a hypothetical question for a very large enterprise cluster with thousands of users.  We very quickly boiled it down to two possibilities.
        1. Export all phones to excel and search for the speed dial
        2. Use an SQL command to interrogate the CUCM's database directly.
        Number 2 sounded more fun to me.  (Note:  I had to giggle just a little bit when I typed that, after all, men never really grow up.)

        After some research in the great Internet, it was quickly obvious that William Bell (@ucguerrilla) was one of the top experts on this topic in the small world of Cisco voice engineers.  I found some examples on his website that almost did exactly what I wanted - but not quite.  That's OK though, I am game for a challenge.  I spoke with him briefly via Twitter and he gave me some great ideas that I just had to test out.  Btw, he has a very good article to get the juices flowing here.

        I also hypothesized that once I had the list of devices with a given speed dial, I might want to change them!  This could be a very real operational problem if an organization, external vendor or other critical contact changes their phone number. 

        So, I proceeded to test on a lab CUCM to see if I could find success with what HAD TO BE a fairly simply request.  After some problems with typos and syntax, I did find success!  (and a new found attraction to finding new ways to leverage this powerful method of manipulating the CUCM database!)

        See the actual commands below.  Thanks again to William for his expertise!

        STEP 1:  Search all devices for a specific speed dial number

        This command will list all devices that have a given speed dial configured. In this case, the 314 number.

        admin:run sql select d.name, d.description, sd.speeddialindex, sd.label, sd.speeddialnumber from speeddial as sd inner join device as d on sd.fkdevice=d.pkid where speeddialnumber like '13145551212'

        Output below (shows only one match):

        name            description     speeddialindex label      speeddialnumber
        =============== =============== ============== ========== ===============
        SEP009C028CFA48 Baba Booey 1000 1              Cell Phone 13145551212  

        STEP 2:  Change all speed dials from one number to another

        This command will change all speed dial entries from one number to another number (explicit match). Note: After the command is run, all phones that had the speed dial changed will reset.

        DISCLAIMER:  CHANGING SPEED DIAL IN THIS MANNER WILL RESET THE PHONE IMMEDIATELY.

        admin:run sql update speeddial set speeddialnumber = '12125551212' where speeddialnumber = '13145551212'
        Rows: 1

        It seemed like the command worked, but let's make sure!

        STEP 3:  Confirm that the correct changes have been made

        This command will display all speed dials for ALL devices. It’s not a very useful command in an enterprise, but it can be manipulated with additional “where” qualifiers. In this case, I just wanted to confirm that step 2 above changed only the speed dial desired. Note: it is the same as the command from step 1, just without the “where” qualifier.

        admin:run sql select d.name, d.description, sd.speeddialindex, sd.label, sd.speeddialnumber from speeddial as sd inner join device as d on sd.fkdevice=d.pkid order by d.name, sd.speeddialindex

        Output below:

        name            description       speeddialindex label      speeddialnumber
        =============== ================= ============== ========== ===============
        SEP009C028CFA48 Baba Booey1000   1              Cell Phone 12125551212    <-
        SEP009C028CFA48 Baba Booey1000   2              TAC        18005532447    
        SEP009C028CFA48 Baba Booey1000   3              Zebar      13091234567    
        SEP009C028CFA49 David Holman 1002 1              Cell Phone 16141234567    
        SEP009C028CFA49 David Holman 1002 2              Baba       15851234567    
        SEP009C028CFA49 David Holman 1002 3              Baba 2     16141234567


        Notes: Interfacing UCCE with CUCM via JTAPI

        By David Holman
        UCCE
        Unified Contact Center Enterprise uses JTAPI prolifically to communicate with Unified Communications Manager (CUCM).  There are many nuances to this communication.  Below is a compilation of information I was able to collect regarding various aspects of JTAPI from the UCCE 9.x SRND and other online sources.  I've interjected additional explanation or clarification as I found necessary.  Note, this is not a complete collection, but I hope that it does make it easier to understand JTAPI's role in communication within the UCCE environment.

        Interfacing UCCE with CUCM via JTAPI


        From the UCCE 9.x SRND:

        In order for JTAPI communications to occur between Unified CM and external applications such as Unified CCE and Unified IP IVR, a JTAPI user ID and password must be configured within Unified CM. Upon startup of the Unified CM PIM or on startup of the Unified IP IVR, the JTAPI user ID and password are used to sign in to Unified CM. This sign-in process by the application (Unified CM PIM or Unified IP IVR) establishes the JTAPI communications between the Unified CM cluster and the application. A separate JTAPI user ID is required for each Unified IP IVR server (and Unified CM PIM). In a Unified CCE deployment with one Unified CM cluster and two Unified IP IVRs, three JTAPI user IDs are required: one JTAPI user ID for Unified CCE and two JTAPI user IDs for the two Unified IP IVRs. The best practice is one PG User for each PG Pair.

        The Unified CM software includes a module called the CTI Manager, which is the layer of software that communicates through JTAPI to applications such as Unified CCE and Unified IP IVR. Every node within a cluster can execute an instance of the CTI Manager process, but the Unified CM PIM on the PG communicates with only one CTI Manager (and thus one node) in the Unified CM cluster. The CTI Manager process communicates CTI messages to and from other nodes within the cluster.

        For example, suppose a deployment has a Voice Gateway homed to node 1 in a cluster, and node 2 executes the CTI Manager process that communicates to Unified CCE. When a new call arrives at this Voice Gateway and needs to be routed by Unified CCE, node 1 sends an intra-cluster message to node 2, which sends a route request to Unified CCE to determine how the call is routed.

        Each Unified IP IVR also communicates with only one CTI Manager (or node) within the cluster. The Unified CM PIM and the two Unified IP IVRs from the previous example can communicate with different CTI Managers (nodes) or they can all communicate with the same CTI Manager (node). However, each communication uses a different JTAPI user ID. The JTAPI user ID is how the CTI Manager tracks the different applications.

        When the Unified CM PIM is redundant, only one side is active and in communication with the Unified CM cluster. Side A of the Unified CM PIM communicates with the CTI Manager on one Unified CM node and Side B of the Unified CM PIM communicates with the CTI Manager on another Unified CM node. The Unified IP IVR does not have a redundant side, but the Unified IP IVR does have the ability to fail-over to another CTI Manager (node) within the cluster if its primary CTI Manager is out of service. 

        UCCE Signaling
        The JTAPI communications between the Unified CM and Unified CCE include three distinct types of messaging:
        1. Routing control Routing control messages provide a way for Unified CM to request routing instructions from Unified CCE.
        2. Device and call monitoring Device monitoring messages provide a way for Unified CM to notify Unified CCE about state changes of a device (phone) or a call.
        3. Device and call control Device control messages provide a way for Unified CM to receive instructions from Unified CCE on how to control a device (phone) or a call.
        A typical Unified CCE call includes all three types of JTAPI communications within a few seconds. When a new call arrives, Unified CM requests routing instructions from Unified CCE. For example, when Unified CM receives the routing response from Unified CCE, Unified CM attempts delivery of the call to the agent phone by instructing the phone to begin ringing. At that point, Unified CM notifies Unified CCE that the device (phone) has started ringing and that notification enables the agent’s answer button on the desktop application. When the agent clicks the answer button, Unified CCE instructs Unified CM to make the device (phone) go off-hook and answer the call.

        In order for the routing control communication to occur, Unified CM requires the configuration of a CTI Route Point. A CTI Route Point is associated with a specific JTAPI user ID, and this association enables Unified CM to know which application provides routing control for that CTI Route Point. Directory (Dialed) Numbers (DNs) are then associated with the CTI Route Point. A DN is associated to a CTI Route Point that is associated with Unified CCE JTAPI user ID. This enables Unified CM to generate a route request to Unified CCE when a new call to that DN arrives.

        In order for the phones to be monitored and controlled, they also must be associated in Unified CM with a JTAPI user ID. Starting with Unified CM 8.0, when using Extension Mobility or Extension Mobility Cross Cluster, it is possible to associate an Extension Mobility device profile instead. In a Unified CCE environment, the IP phones or the corresponding Extension Mobility device profiles are associated with Unified CCE JTAPI user IDs. When an agent logs in from the desktop, the Unified CM PIM requests Unified CM to allow the PIM to begin monitoring and controlling that phone. Until the login has occurred, Unified CM does not allow Unified CCE to monitor or control that phone. If the device or the corresponding Extension Mobility device profile has not been associated with a Unified CCE JTAPI user ID, then the agent login request fails.

        Support for Extension Mobility Cross Cluster (EMCC) is provided in Release 8.0. The Unified CCE PIM phone registers to the local Unified CM after Extension Mobility login and it looks like an agent situated across a WAN. The Unified CCE peripheral is managing the agent devices based on the Extension Mobility profile rather than on a phone device in the Application User on Unified CM. For more details, see the Cisco Unified Communications Solution Reference Network Design Guide (SRND).

        Because EMCC is now supported, you can associate Extension Mobility devices by two methods; either by device or by user profile. The best practice is to associate the Extension Mobility profile to the CCE Application User on UC Manager.

        Because the Unified IP IVR also communicates with Unified CM using the same JTAPI protocol, these same three types of communications also occur with the Unified IP IVR. Unlike Unified CCE, the Unified IP IVR provides both the application itself and the devices to be monitored and controlled.

        The devices that Unified CCE monitors and controls are the physical phones. The Unified IP IVR does not have physical ports like a traditional IVR. Its ports are logical ports (independent software tasks or threads running on the Unified IP IVR application server) called CTI Ports. For each CTI Port on the Unified IP IVR, there needs to be a CTI Port device defined in Unified CM.

        Unlike a traditional PBX or telephony switch, Unified CM does not select the Unified IP IVR port to which it sends the call. Instead, when a call needs to be made to a DN that is associated with a CTI Route Point that is associated with a Unified IP IVR JTAPI user, Unified CM asks the Unified IP IVR (through JTAPI routing control) which CTI Port (device) handles the call. Assuming the Unified IP IVR has an available CTI Port, the Unified IP IVR responds to the Unified CM routing control request with the Unified CM device identifier of the CTI Port that is going to handle that call.

        The following scenarios are examples of device and call control performed by the Unified IP IVR.

        • When an available CTI Port is allocated to the call, a Unified IP IVR workflow is started within the Unified IP IVR. When the Unified IP IVR workflow executes the accept step, a JTAPI message is sent to Unified CM to answer the call on behalf of that CTI Port (device). When the Unified IP IVR workflow wants the call transferred or released, it again instructs Unified CM on what to do with that call.
        • When a caller releases the call while interacting with the Unified IP IVR, the Voice Gateway detects the caller release and notifies Unified CM by using H.323 or Media Gateway Control Protocol (MGCP), which then notifies the Unified IP IVR by using JTAPI. When DTMF tones are detected by the Voice Gateway, it notifies Unified CM via H.245 or MGCP, which then notifies the Unified IP IVR through JTAPI.

        In order for the CTI Port device control and monitoring to occur, the CTI Port devices on Unified CM must be associated with the appropriate Unified IP IVR JTAPI user ID. If you have two 150-port Unified IP IVRs, you have 300 CTI ports. Half of the CTI ports are associated with JTAPI user Unified IP IVR 1, and the other half of the CTI ports are associated with JTAPI user Unified IP IVR 2.

        While Unified CM can be configured to route calls to Unified IP IVRs on its own, routing of calls to the Unified IP IVRs in a Unified CCE environment is done by Unified CCE (even if you have only one Unified IP IVR and all calls require an initial IVR treatment). Doing so ensures proper Unified CCE reporting. For deployments with multiple Unified IP IVRs, this routing practice also allows Unified CCE to load-balance calls across the multiple Unified IP IVRs.

        Design Warning:  SIP, JTAPI and DTMF


        From the UCCE 9.x SRND:

        Unified IP IVR is notified of caller entered digits (DTMF input) by way of JTAPI messages from Unified CM. Unified IP IVR and Unified CM do not support mechanisms to detect in-band DTMF digits. In deployments with SIP Voice Gateways or SIP phones that support only in-band DTMF (or are configured to use in-band DTMF In accordance with RFC 2833), Unified CM must invoke an MTP resource to convert the in-band DTMF signaling to out-of band signaling so that the Unified IP IVR can be notified of caller entered digits.
        Therefore, in environments that include SIP phones or gateways, it is necessary to provision sufficient MTP resources. Keep this in mind if the phones need to interact with Unified IP IVR. Likewise, CTI ports do not support in-band DTMF (RFC 2833). The Mobile Agent feature relies on CTI ports, so MTP resources are required when in-band DTMF (RFC 2833) is negotiated. SIP trunking is supported using CVP deployments with Cisco IOS gateways (IOS GW) and Unified Border Elements (CUBE).

        Additional UCCE 9.x SRND Information About JTAPI in the UCCE Environment

        1. Unified CM and CTI Manager Design Considerations
        2. Configuring the UCCE PG for CTI Manager Redundancy
        3. Unified IP IVR Design Considerations
        4. Peripheral Gateway Design Considerations
        5. Unified CCE Failure Scenarios
        6. Cisco Remote Silent Monitoring

        Notes: Cisco Unified Contact Center Enterprise (UCCE) Components

        By David Holman
        UCCE
        First in a likely series of UCCE posts coming from my first pass through the UCCE 9.x SRND, this post will focus on the various Cisco Unified Contact Enter Enterprise (UCCE) components.  The following is a mashup of information from the SRND to be used as a quick UCCE "cheat sheet".

        Cisco Unified Contact Center Enterprise Components


        The Unified CCE solution consists primarily of four Cisco software products:
        • Unified Communications Infrastructure: Cisco Unified CM via JTAPI
        • Queuing and Self-Service: Cisco Unified IP Interactive Voice Response (Unified IP IVR) or Cisco Unified Customer Voice Portal (Unified CVP).  
          • The Unified IP IVR provides prompting, collecting, and queuing capability for the Unified CCE solution.  Unified IP IVR does not provide call control as Unified CVP does because it is behind Unified CM and under the control of the Unified CCE software by way of the Service Control Interface (SCI). When an agent becomes available, the Unified CCE software instructs the Unified IP IVR to transfer the call to the selected agent phone. The Unified IP IVR then requests Unified CM to transfer the call to the selected agent phone.

            Unified IP IVR is a software application that runs on Cisco MCS Servers. You can deploy multiple Unified IP IVR servers with a single Unified CM cluster under control of Unified CCE.

            Unified IP IVR has no physical telephony trunks or interfaces like a traditional IVR. The telephony trunks are terminated at the Voice Gateway. Unified CM provides the call processing and switching to set up a g.711 or G.729 Real-Time Transport Protocol (RTP) stream from the Voice Gateway to the Unified IP IVR. The Unified IP IVR communicates with Unified CM through the Java Telephony Application Programming Interface (JTAPI), and the Unified IP IVR communicates with Unified CCE through the Service Control Interface (SCI) with a VRU Peripheral Gateway or System Peripheral Gateway. 
          • Unified Customer Voice Portal (Unified CVP) is a software application that runs on industry-standard servers such as Cisco Media Convergence Servers (MCS). It provides prompting, collecting, queuing, and call control services using standard web-based technologies. The Unified CVP architecture is distributed, fault tolerant, and highly scalable. With the Unified CVP system, voice is terminated on Cisco IOS gateways that interact with the Unified CVP application server using VoiceXML (speech) and H.323 or SIP (call control).

            The Unified CVP software is tightly integrated with the Cisco Unified CCE software for application control. It interacts with Unified CCE using the Voice Response Unit (VRU) Peripheral Gateway Interface. The Unified CCE scripting environment controls the execution of building-block functions such as play media, play data, menu, and collect information. The Unified CCE script can also invoke external VoiceXML applications to be executed by the Unified CVP VoiceXML Server, an Eclipse and J2EE-based scripting and web server environment. VoiceXML Server is well suited for sophisticated and high-volume IVR applications and it can interact with custom or third-party J2EE-based services. These applications can return results and control to the Unified CCE script when complete. Advanced load balancing across all Unified CVP solution components can be achieved by Cisco Content Services Switch (CSS) and Cisco IOS Gatekeepers or Cisco Unified Presence SIP Proxy Servers.
        • Contact Center Routing and Agent Management: Unified CCE. The major components are CallRouter, Logger, Peripheral Gateway, and the Administration & Data Server/Administration Client.
          • CallRouter: Makes all routing decisions on how to route a call or customer contact. Often just referred to as the “Router” in the context of Unified CCE components.
          • Logger: The database server that stores contact center configuration data and temporarily stores historical reporting data for distribution to the data servers.
          • CTI Object Server (CTI OS): CTI interface for Agent Desktops.
          • Peripheral Gateway (PG): Interfaces to various peripheral devices, specifically to Unified CM, VRU (Unified IP IVR or Unified CVP), or Multichannel products (EIM and WIM for email and chat). The PG includes one or more Peripheral Interface Managers (PIMs) for the specific device interfaces.
          • Agent PG: PG that has a Unified CM PIM.
          • Unified CM Peripheral Interface Manager (PIM): Part of a PG that interfaces to a Unified CM cluster by using the JTAPI protocol.
          • VRU PIM: Part of a PG that interfaces to the Unified IP IVR or Unified CVP through the Service Control Interface (SCI) protocol.
          • MR PIM: Part of a PG that interfaces to call center Multimedia products, specifically EIM and WIM for email and chat.
          • CTI Server: Part of the PG that interfaces to CTI OS and provides an open CTI interface, which is useful for some server-to-server communications.
          • Network Interface Controller (NIC): Interfaces to the public switched telephone network (PSTN), which enables Unified CCE to control how the PSTN routes a call.
          • Administration & Data Server: Configuration interface and real-time and historical data storage (for example, for reporting). There are several different deployment models described later in this chapter.
          • Administration Client: Configuration interface. This component has a subset of the functionality of the Administration & Data Server. It is a lighter weight deployment because it does not require a local database and it is deployed to allow more places from which to configure the solution.
          • Cisco Unified Intelligence Center (Unified Intelligence Center): Provides web browser-based real-time and historical reporting. Unified Intelligence Center also works with other Cisco Unified Communications products.
        • Agent Desktop Software: Cisco Agent Desktop, Cisco Toolkit Agent Desktop (CTI OS), Cisco Finesse, or integrations with third-party customer relationship management (CRM) software through Cisco Unified CRM Connector.


        Acronyms: Cisco Unified Contact Center Enterprise (UCCE)

        By David Holman
        UCCE
        As the first step of my pursuit to improve my overall understanding of the Cisco Unified Contact Center Enterprise (UCCE) product, I've committed to reading the UCCE 9.x SRND cover to cover.  That's right, 398 pages of pure unadulterated gleeful knowledge absorption.  At least that is how exciting I'm trying to convince myself it will be.

        As I progress, I'm likely to create a series of UCCE-related posts in the coming weeks.  I'll use it as a way to reinforce my learning and perhaps share a few things that I learn along the way. 

        I'm going to get the series started by focusing on the list of acronyms in the SRND.  Yes, it's quite an impressive list. Many of them were already familiar to me due to my experience with other aspects of the Cisco UC product line.  As I proceed, I will use this acronym post as a way to link content and deepen understanding.

        I'd also welcome anyone to contribute to the list - this will be a living and changing document!


        Cisco UCCE SRND 9.x Acronyms

        A

        ACD     Automatic call distribution
        AD      Active Directory
        ADSL    Asymmetric digital subscriber line
        AHT     Average handle time
        ANI     Automatic Number Identification
        APG     Agent Peripheral Gateway
        AQT     Average queue time
        ARM     Agent Reporting and Management
        ASA     Average speed of answer
        ASR     Automatic speech recognition
        AVVID   Cisco Architecture for Voice, Video, and Integrated Data
        AW      Administrative Workstation
        AWDB    Administrative Workstation Database


        B

        BBWC    Battery-backed write cache
        BHCA    Busy hour call attempts
        BHCC    Busy hour call completions
        BHT     Busy hour traffic

        BIB     Built-In-Bridge
        BOM     Bill of materials
        bps     Bits per second
        Bps     Bytes per second


        C

        CAD     Cisco Agent Desktop
        CC      Contact Center
        CCE     Contact Center Enterprise
        CG      CTI gateway
        CIPT OS Cisco Unified Communications Operating System
        CIR     Cisco Independent Reporting
        CMS     Configuration Management Service
        ConAPI  Configuration Application Programming Interface
        CPE     Customer premises equipment
        CPI     Cisco Product Identification tool
        CRM     Customer Relationship Management
        CRS     Cisco Customer Response Solution
        CSD     Cisco Supervisor Desktop
        CSS     Cisco Content Services Switch

        CSS     Calling Search Space
        CSV     Comma-separated values
        CTI     Computer telephony integration
        CTI OS  CTI Object Server

        CUCM    Cisco Unified Communications Manager 
        CUSP    Cisco Unified SIP Proxy
        CVP     Cisco Unified Customer Voice Portal


        D

        DCA     Dynamic Content Adapter
        DCS     Data Collaboration Server
        DES     Data Encryption Standard
        DHCP    Dynamic Host Configuration Protocol
        DID     Direct inward dial
        DiffServ Differentiated Services
        DMP     Device Management Protocol
        DMZ     Demilitarized zone
        DN      Directory number
        DNP     Dialed Number Plan
        DNS     Domain Name System
        DSCP    Differentiated Services Code Point
        DSL     Digital subscriber line
        DSP     Digital signal processor

        DTAL    Direct Transfer Across Line, MLAM required
        DTMF    Dual Tone Multi Frequency


        E

        ECC     Expanded Call Context
        EIM     E-mail Integration Manager

        H

        HA WAN  Highly available WAN
        HDS     Historical Data Server
        HSRP    Hot Standby Router Protocol


        I

        ICCS    Intra-Cluster Communication Signaling
        ICM     Cisco Unified Intelligent Contact Management
        IDF     Intermediate distribution frame
        IDS     Intrusion detection system
        IntServ Integrated services

        IOS     Internetwork Operating System
        IP      Internet Protocol
        IPCC    Cisco IP Contact Center
        IPM     Internetwork Performance Monitor
        IPPA    Unified IP Phone Agent
        ISN     Internet service node
        IVR     Interactive voice response
        IXC     Inter-exchange carrier


        J

        JTAPI   Java Telephony Application Programming Interface
        JAL     Joint Access Line, MLAM required.

        K

        kb      Kilobits
        kB      Kilobytes
        kbps    Kilobits per second
        kBps    Kilobytes per second


        L

        LAMBDA  Load Adaptive Message-Base Data Archive
        LAN     Local area network
        LCC     Logical contact center
        LDAP    Lightweight Directory Access Protocol
        LEC     Local exchange carrier
        LAA     Longest available agent
        LSPAN   Local switched port analyzer


        M

        MAC     Media access control
        Mbps    Megabits per second
        MC      Management center
        MCS     Media Convergence Server
        MDF     Main distribution frame
        MDS     Message delivery Subsystem
        MED     Minimum expected delay
        MGCP    Media Gateway Control Protocol

        MLAM    Multi-Line Agent Mode [1]
        MoH     Music on hold
        MR      Media routing
        MRCP    Media Resource Control Protocol
        MTU     Maximum transmission unit


        N

        NAT     Network Address Translation
        NDIS    Network driver interface specification
        NIC     Network interface controller


        O

        OAMP    Operations, Administration, Maintenance, and Provisioning
        OPC     Open peripheral controller
        OS      Object server
        OU      Organizational unit


        P

        PAT     Port address translation
        PerfMon Microsoft Windows Performance Monitor
        PG      Peripheral gateway
        PHB     Per-hop behavior
        PIM     Peripheral interface manager
        PLAR    Private line automatic ringdown
        PPPoE   Point-to-Point Protocol over Ethernet
        Progger Peripheral gateway, router, and logger
        PSPAN   Port switched port analyzer
        PSTN    Public switched telephone network
        PVC     Permanent virtual circuit


        Q

        QoS     Quality of Service

        R

        RAID    Redundant array of inexpensive disks
        RIS     Real-time information server
        Rogger  Router and Logger
        ROI     Return on investment
        RONA    Reroute On No Answer
        RSPAN   Remote switched port analyzer
        RSVP    Resource Reservation Protocol
        RTD     Real-Time Distributor
        RTMT    Real-Time Monitoring Tool
        RTP     Real-Time Transport Protocol


        S

        SAA     Service assurance agent
        SCCP    Skinny Client Control Protocol
        SCI     Service control interface
        SCSI    Small computer system Interface
        SDL     Signal distribution layer
        SE      Systems engineer
        SIP     Session Initiation Protocol
        SLG     Service level goal
        SNMP    Simple Network Management Protocol
        SPAN    Switched port analyzer
        SRND    Solution Reference Network Design
        SRST    Survivable remote site telephony
        SSL     Secure Socket Layer
        SUS     Microsoft Software Update Services


        T

        TAC     Cisco Technical Assistance Center
        TAPI    Telephony application programming interface
        TCD     Telephony Call Dispatcher
        TCP     Transmission Control Protocol
        TDM     Time-division multiplexing
        TES     Task event services
        TFTP    Trivial File Transfer Protocol
        TLS     Transport Layer Security
        TNT     Takeback N Transfer
        TOS     Test other side
        TTS     Text-to-speech


        U

        UDP     User Datagram Protocol
        UI      User interface
        URL     Uniform resource locator


        V

        V3PN    Cisco Voice and Video Enabled Virtual Private Network
        VLAN    Virtual local area network
        VMS     CiscoWorks VPN/Security Management Solution
        VoIP    Voice over IP
        VPN     Virtual private network
        VPNSM   Virtual Private Network Services Module
        VRU     Voice response unit
        VSPAN   Virtual LAN switched port analyzer
        VXML    Voice XML (Extensible Markup Language)


        W

        WAN     Wide area network
        WIM     Web Interaction Manager
        WUS     Microsoft Windows Update Services


        X

        XML     Extensible markup language





        CUCM Special Characters

        By David Holman
        As you are all aware, the list of characters available to use when configuring various fields in CUCM depending upon the field type.  I chose CUCM 8.5 for my sample below as it is what I'm using mostly these days (and will be a quick reference when my memory is shot), but I'm sure some of you can contribute some differences in other versions in the comment section down below.

        Be sure to click the "more" link - it goes to the big and juicy Cisco CUCM System Guide for tons of great details!

        CUCM 8.5 Special Characters [ more ]



        Character

        Description

        Examples

        @

        The at symbol (@) wildcard matches all NANP numbers.

        Each route pattern can have only one @ wildcard.

        The route pattern 9.@ routes or blocks all numbers that the NANP recognizes.

        The following route patterns examples show NANP numbers that the @ wildcard encompasses:

        • 0

        • 1411

        • 19725551234

        • 101028819725551234

        • 01133123456789

        X

        The X wildcard matches any single digit in the range 0 through 9.

        The route pattern 9XXX routes or blocks all numbers in the range 9000 through 9999.

        !

        The exclamation point (!) wildcard matches one or more digits in the range 0 through 9.

        The route pattern 91! routes or blocks all numbers in the range 910 through 91999999999999999999999.

        ?

        The question mark (?) wildcard matches zero or more occurrences of the preceding digit or wildcard value.

        The route pattern 91X? routes or blocks all numbers in the range 91 through 91999999999999999999999.

        +

        The plus sign (+) wildcard matches one or more occurrences of the preceding digit or wildcard value.

        The route pattern 91X+ routes or blocks all numbers in the range 910 through 91999999999999999999999.

        [ ]

        The square bracket ([ ]) characters enclose a range of values.

        The route pattern 813510[012345] routes or blocks all numbers in the range 8135100 through 8135105.

        -

        The hyphen (-) character, used with the square brackets, denotes a range of values.

        The route pattern 813510[0-5] routes or blocks all numbers in the range 8135100 through 8135105.

        ^

        The circumflex (^) character, used with the square brackets, negates a range of values. Ensure that it is the first character following the opening bracket ([).

        Each route pattern can have only one ^ character.

        The route pattern 813510[^0-5] routes or blocks all numbers in the range 8135106 through 8135109.

        .

        The dot (.) character, used as a delimiter, separates the Cisco Unified Communications Manager access code from the directory number.

        Use this special character, with the discard digits instructions, to strip off the Cisco Unified Communications Manager access code before sending the number to an adjacent system.

        Each route pattern can have only one dot (.) character.

        The route pattern 9.@ identifies the initial 9 as the Cisco Unified Communications Manager access code in an NANP call.

        *

        The asterisk (*) character can provide an extra digit for special dialed numbers.

        You can configure the route pattern *411 to provide access to the internal operator for directory assistance.

        #

        The octothorpe (#) character generally identifies the end of the dialing sequence.

        Ensure the # character is the last character in the pattern.

        The route pattern 901181910555# routes or blocks an international number that is dialed from within the NANP. The # character after the last 5 identifies this digit as the last digit in the sequence.

        \+

        A plus sign preceded by a backslash, that is, \+, indicates that you want to configure the international escape character +.

        Using \+ means that the international escape character + is used as a dialable digit, not as a wildcard.

        For more information on this character, see the "Using the International Escape Character +" section.



        Field

        Valid entries

        Call Park Number/Range

        [ ^ 0 1 2 3 4 5 6 7 8 9 - ] X * #

        Calling Party Transform Mask

        0 1 2 3 4 5 6 7 8 9 X A B C D * # +

        Called Party Transform Mask

        0 1 2 3 4 5 6 7 8 9 X A B C D * # +

        Caller ID DN (Gateways and Trunks)

        0 1 2 3 4 5 6 7 8 9 X * # +

        Directory Number

        \+ [ ^ 0 1 2 3 4 5 6 7 8 9 - ] + ? ! X * # +

        Directory Number (Call Pickup Group Number)

        0 1 2 3 4 5 6 7 8 9

        External Phone Number Mask

        0 1 2 3 4 5 6 7 8 9 X * # +

        Forward All

        0 1 2 3 4 5 6 7 8 9 * # +

        Forward Busy

        0 1 2 3 4 5 6 7 8 9 * # +

        Forward No Answer

        0 1 2 3 4 5 6 7 8 9 * # +

        Meet-Me Conference Number

        [ ^ 0 1 2 3 4 5 6 7 8 9 - ] X * #

        Prefix Digits

        0 1 2 3 4 5 6 7 8 9 A B C D * # +

        Prefix DN (Gateways and Trunks)

        0 1 2 3 4 5 6 7 8 9 * # +

        Route Filter Tag Values

        [ ^ 0 1 2 3 4 5 6 7 8 9 - ] X * #

        Route Pattern

        [ ^ 0 1 2 3 4 5 6 7 8 9 A B C D - ] + ? ! X * # + . @ \+

        Translation Pattern

        [ ^ 0 1 2 3 4 5 6 7 8 9 A B C D - ] + ? ! X * # + . @ \+

        Hunt Pilot

        [ ^ 0 1 2 3 4 5 6 7 8 9 A B C D - ] + ? ! X * # + . @ \+

        Bonus Question


        Can anyone tell me why "A B C D" are valid characters when configuring a route pattern in CUCM 8.5?

        Answer:  [ source ]

        There are 16 DTMF digits (arranged as four columns by four rows). In addition to the standard 12 keypad digits—0 to 9, *, and #—an additional four digits form an extra fourth column of digits called simply A, B, C, and D. Because the ABCD digits are unavailable on a normal phone keypad, you are unlikely to ever come across these for normal phone calls. They are used occasionally by voice mail systems to operate an intersystem exchange of voice messages between separate voice mail systems using a standard called Analog Message Interchange Standard (AMIS). 

        Some security-type phones also use the ABCD digits for initial negotiation. You may also see these used where there is a need to create telephone numbers that cannot be directly dialed from a phone keypad. One example of this is if you want to create nondialable phone numbers for intercoms. You normally place an intercom call by pressing a button specifically configured for intercom (this works somewhat like a speed-dial button), so you do not need to be able to enter individual dialed digits.

        Codec Overview: Portmanteaux, Conjugate Structures and Lots of Code Excitement

        By David Holman
        I have to admit something.  I have short-term memory when it comes to studying certain topics in
        preparation for the CCIE Voice or (now) the CCIE Collaboration certification.  Codecs and DSPs are full to the rim with bit rates, acronyms and trickery that always have me scrambling to confirm my fleeting memory of them.  So, I'm embarking to try to make more sense of it all.  Here we go!

        Definition of "Codec"


        In a general sense, the word "codec" is a portmanteau of "coder-decoder" or "compressor-decompressor".  As it pertains to voice technologies, it simply defines a method for converting analog voice signals to a digital form and then back to audio so that a remote party hears a rendition that closely approximates the original analog voice input.  (i.e,. audio compression format)

        Codecs that reduce quality in order to achieve compression are considered "lossy".  This is an important concept when it comes to considering how to design network bandwidth requirements between call parties.  The ideal codec will consume minimal network bandwidth AND provide excellent quality ( minimal loss ).


        Technical Details of ITU-T Audio Compression Formats

         

        Audio Codec Compression Algorithms


        • Pulse Code Modulation (PCM):  In a PCM stream, the amplitude of the analog signal is sampled regularly at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps.
        • Adaptive Differential Pulse Code Modulation (ADPCM): Adaptive differential pulse-code modulation (ADPCM) is a variant of differential pulse-code modulation (DPCM) that varies the size of the quantization step, to allow further reduction of the required bandwidth for a given signal-to-noise ratio.
        • Low-Delay Code Excited Linear Prediction (LDCELP): Delay of the codec is only 5 samples (0.625 ms). The linear prediction is calculated backwards with a 50th order linear predictive coding filter. The excitation is generated with gain scaled VQ. The essence of CELP techniques, which is an analysis-by-synthesis approach to codebook search, is retained in LD-CELP. The LD-CELP however, uses backward adaptation of predictors and gain to achieve an algorithmic delay of 0.625 ms.
        • Conjugate-Structure Algebraic-Code-Excited Linear-Prediction (CS-ACELP): A voice compression algorithm defined in ITU-T G.729,ACELP improves on CELP through the algebraic expression, rather than the numeric description, of each entry in the codebook. ACELP yields quality that is considered to be as good as ADPCM, but requiring bandwidth of only 8 kbps, which yields a compression ration of 8:1. 
        • Multi-Pulse, Multi-Level Quantization (MP-MLQ):  Used by the high-rate coder portion of the ITU-T G.723.1 standard.

         

        Codec Mean Opinion Scores


        Each codec provides a certain quality of speech. The quality of transmitted speech is a subjective opinion of the listener. The mean opinion score (MOS) is a scale from 1 (bad) to 5 (excellent) used to rate the average quality.  Notice in the scale below the relationship between "Bit Rate" and "Compression Delay". These are important considerations when designing a voice network.

        Compression Method Bit Rate (kbps) MOS ScoreCompression Delay (ms)
        G.711 PCM 644.10.75
        G.726 ADPCM 323.851
        G.728 LD-CELP 163.613 to 5
        G.729 CS-ACELP 83.9210
        G.729 x 2 Encodings 83.2710
        G.729 x 3 Encodings 82.6810
        G.729a CS-ACELP 83.710
        G.723.1 MP-MLQ 6.33.930
        G.723.1 ACELP 5.33.6530

        Codec DSP Complexity


        Some codecs require additional DSP processing power to handle the required computations of the codec. High complexity codecs will result in fewer calls per DSP. 

        • Medium Complexity:  G.711 (a-law and u-law), G.726, G.729ab (G.729a Annex B)
        • High Complexity:  G.728, G.723, G.729, G.729b (G.729 Annex B)
        Thanks to @ciscovoicedude for pointing out the following as published on their DSP Calculator:

        Low Complexity Codecs: G.711, Fax Passthrough, Modem Passthrough, and Clear-Channel
        Medium Complexity Codecs: G.726, G.729A, G.729AB, G.722, and T.38 Fax Relay
        High Complexity Codecs: G.729, G.729B, G.723, G.728, iLBC, GSM-AMR, T.38 SG3 Fax, and Modem Relay

        In Summary


        That's a good start for now!  So many more codec topics that get much more detailed and are better left for another post later.  

          Cisco Changing Position on CCIE Voice to CCIE Collaboration Transition

          By David Holman
          In response to the criticism generated by their original announcement related to the "evolution" of the CCIE Voice certification to the CCIE Collaboration certification, Cisco has backpedaled some with this admission that they are listening and will be changing their stance:

          "We are listening to the feedback from our valued CCIE community, and will be adjusting the CCIE Collaboration requirements. As a quick preview of the evolution of the CCIE Collaboration certification, a current holder of the CCIE Voice designation will now be able to migrate to a CCIE Collaboration credential by taking the CCIE Collaboration written exam only. We appreciate all of the great feedback and patience of the community while we update our webpages to reflect this change. We will be communicating further details about this modification as soon as possible."

          This is a great announcement for all CCIE Voice folks out there!
          ❌