Subscribe: CCIE Voice Study Guide
Preview: CCIE Voice Study Guide - CCIE Collaboration | CCIE Voice | CCNP Voice | CCNA Voice

Cisco voice blog with info related to voice and collaboration products and certifications: CCNA Voice, CCNP Voice, CCIE Voice and CCIE Collaboration.

Updated: 2018-04-16T07:40:11.102-04:00


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


I'm the Boss of Me:A Guide to Owning Your Careerby Jeanne Beliveau-DunnI 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, 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 ContentsChapter 1 - You Are Who You Are Because of Your ChoicesChapter 2 - You Are a Work in Progress - Challenge Yourself to Get BetterChapter 3 - Do You Know What You Want from Your Career and Life?Chapter 4 - Vision Strategy ExecutionChapter 5 - Becoming Your Vision:  Aligning Your BehaviorChapter 6 - Becoming Your Vision II:  Articulating Your Brand and Building Your BenchChapter 7 - Attitude, Altruism and AltitudeChapter 8 - Myths and Murders:  Getting Past Roadblocks and BarriersChapter 9 - Planning and Making ChoicesChapter 10 - Self-Leadership 2.0:  Success in the Face of ObstaclesReviewAs 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 AuthorJeanne 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 Akhil Behl, Berni Gardiner,
Joshua Samuel Finke

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


CIPTV2, 300-075 GuideAlex Hannah, Akhil BehlCisco PressRecently, 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 5RecommendedSelf-Study ResourceSo, 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 ProgramBook 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:Cisco Collaboration Solution Multisite Deployment ConsiderationsUnderstanding Multisite Deployment SolutionsOverview of PSTN and Intersite Connectivity OptionsURI-Based Dial Plan for Multisite DeploymentsRemote Site Telephony and Branch Redundancy OptionsCisco Collaboration Solution Bandwidth ManagementCall Admission Control (CAC) ImplementationImplementing Cisco Device MobilityCisco Extension MobilityImplementing Cisco Unified MobilityCisco Video Communication Server & Expressway DeploymentDeploying Users and Local Endpoints in Cisco VCS ControlInterconnecting Cisco Unified Communications Manager and Cisco VCSCisco Unified Communications Mobile and Remote AccessCisco Inter-Cluster Lookup Service (ILS) and Global Dial Plan Replication (GDPR)Cisco Service Advertisement Framework (SAF) & Call Control Discovery (CCD)Answers AppendixGoals and MethodsTo 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 QuestionEach 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 t[...]

Easily Upgrade CCNP Voice to CCNP Collaboration


CIPTV2 by Alex HannahI'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 CollaborationWhat 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 CIPTV2Check out this detailed PDF for the full list of 300-075 exam topics.Study resources for 300-075 CIPTV2?Cisco Press Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2)Cisco Collaboration Solutions Design Guidance <- Great Top Level CollectionCisco CIPTV2 Study MaterialCisco Collaboration Study Group @ The Cisco Learning NetworkCisco Collaboration 9.x Solution Reference Network DesignsCisco TelePresence VCS Configuration GuidesCisco TelePresence VCS Document CollectionH.320 Gateway to H.323 Gatekeeper Video Call Flow[...]

Call Classification: OnNet vs OffNet


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:Route Pattern Configuration, Cisco Unified Communications Manager Administration GuideGateway Configuration, Cisco Unified Communications Manager Administration GuideTrunk Configuration, Cisco Unified Communications Manager Administration GuideService Parameter Configuration, Cisco Unified Communications Manager Administration GuideConference Bridges, Cisco Unified Communications Manager System GuideAdditional 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 ConsiderationsDrop Conference When No OnNet Party RemainsThe 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 conferen[...]

Cisco UC Call Recording with MediaSense


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:Cisco introduced the MediaSense product that captures audio-streams duplicated by Cisco IP phone built-in bridge;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;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 RecordingUnfortunately, 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 InnovationAs 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 c[...]

Most Useful Cisco UC Resources (A Crowdsourcing Effort)


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 ControlCisco Collaboration Systems Documentation : Full matrix of systems documentation back to UC version 6.1.  SRND, configuration notes, system release notes, etc.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 GatewaysCisco 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!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. 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 ApplicationsCisco Unified Communications Manager IM & Presence Configuration Guides : Good starting point when doing any configuration for IM & Presence.  Collaboration Management and LicensingCisco Prime Collaboration End User Guides : Versions 9.0 through current (10.x)Telephony ExtensionsCisco Emergency Responder DocWiki : Lots and lots of troubleshooting guides to assist with a variety of CER-related issues.Voice ServersCisco MCS-78XX Boot Error Codes : Quick list of common boot errors that can happen when booting 78XX hardware.VirtualizationDocWiki - 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).   MiscellaneousRecognizing 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 ToolsTranslatorX : 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] Notepad++ : A free source code editor and Notepad replacement that supports several languages.[submitted by Nick Barnett] Agent Ransack : A free software program for finding files on your PC or network drives.[submitted by Nick Barnett] 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] 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


Self-Evaluation ToolAs 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 ToolNote 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 ToolSuggested Mastery Levels:0  -  No knowledge or experience at all1  -  Junior Engineer - no hands-on experience2  -  Junior Engineer - some hands-on experience, basic ability to configure3  -  Engineer - moderate experience, average configuration and troubleshooting4  -  Senior Engineer - extensive experience,  advanced configuration and troubleshooting5  -  Senior Engineer - subject matter expert, elite configuration and troubleshooting ability Section Mastery LevelsI 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 PointsFinal ThoughtsI 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


She Loves Call QueuingLately, 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 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 w[...]

CUCM (BE) SQL Query: Exposing Call Forwarding History


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 HistoryQuery Compositionselect n.dnorpattern, cfhd.dnorpattern, cfhd.datetimestamp from numplan as n inner join callforwardhistorydynamic as cfhd on cfhd.fknumplan=n.pkid where n.dnorpattern = 1000SyntaxFrom 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 = 1000Outputdnorpattern dnorpattern datetimestamp=========== =========== =============1000        5551900     13977827521000        5559000     1397786930  <-1000        5551213     13977869431000        5559000     1397786952  <-1000        5551215     1397786961 This output proves it!  The DN was forwarded to the pizza joint on two different occasions!InterpretationFirst 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 datetimestampLaunch Microsoft Excel.In cell A1 type the datetimestamp from column 3.  In cell A2, paste the formula =A1/86400+DATE(1970,1,1)Right-click on cell A2 and select format cells.Under the Number tab select Time where the format is 3/14/98 1:30 PM.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:  callforwardhistorydynamiccallforwardhistorydynamic from CUCM 8.6 Data DictionaryAdditional CUCM SQL InformationCUCM SQL Series: A Series:  From, this is really a fantastic starting [...]

User Management - CCIE Collaboration Written Study Guide


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 <

UC Deployment Models - CCIE Collaboration Written Study Guide


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 <

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


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.Export all phones to excel and search for the speed dial 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 numberThis command will list all devices that have a given speed dial configured. In this case, the 314 number.admin:run sql select, 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 anotherThis 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: 1It seemed like the command worked, but let's make sure! STEP 3:  Confirm that the correct changes have been madeThis 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.description, sd.speeddialindex, sd.label, sd.speeddialnumber from speeddial as sd inner join device as d on sd.fkdevice=d.pkid order by, sd.speeddialindexOutput below:name            description       speeddialindex label  &nbs[...]

Notes: Interfacing UCCE with CUCM via JTAPI


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 JTAPIFrom 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 SignalingThe JTAPI communications between the Unified CM and Unified CCE include three distinct types of messaging:Routing control Routing control messages provide a way for Unified CM to request routing instructions from Unified CCE.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.Device and call control Device control messages provide a way for Unified CM to receive [...]

Notes: Cisco Unified Contact Center Enterprise (UCCE) Components


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 ComponentsThe Unified CCE solution consists primarily of four Cisco software products:Unified Communications Infrastructure: Cisco Unified CM via JTAPIQueuing 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 [...]

Acronyms: Cisco Unified Contact Center Enterprise (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 AcronymsAACD     Automatic call distributionAD      Active DirectoryADSL    Asymmetric digital subscriber lineAHT     Average handle timeANI     Automatic Number IdentificationAPG     Agent Peripheral GatewayAQT     Average queue timeARM     Agent Reporting and ManagementASA     Average speed of answerASR     Automatic speech recognitionAVVID   Cisco Architecture for Voice, Video, and Integrated DataAW      Administrative WorkstationAWDB    Administrative Workstation DatabaseBBBWC    Battery-backed write cacheBHCA    Busy hour call attemptsBHCC    Busy hour call completionsBHT     Busy hour trafficBIB     Built-In-Bridge BOM     Bill of materialsbps     Bits per secondBps     Bytes per secondCCAD     Cisco Agent DesktopCC      Contact CenterCCE     Contact Center EnterpriseCG      CTI gatewayCIPT OS Cisco Unified Communications Operating SystemCIR     Cisco Independent ReportingCMS     Configuration Management ServiceConAPI  Configuration Application Programming InterfaceCPE     Customer premises equipmentCPI     Cisco Product Identification toolCRM     Customer Relationship ManagementCRS     Cisco Customer Response SolutionCSD     Cisco Supervisor DesktopCSS     Cisco Content Services SwitchCSS     Calling Search SpaceCSV     Comma-separated valuesCTI     Computer telephony integrationCTI OS  CTI Object ServerCUCM    Cisco Unified Communications Manager CUSP    Cisco Unified SIP ProxyCVP     Cisco Unified Customer Voice PortalDDCA     Dynamic Content AdapterDCS     Data Collaboration ServerDES     Data Encryption StandardDHCP    Dynamic Host Configuration ProtocolDID     Direct inward dialDiffServ Differentiated ServicesDMP     Device Management ProtocolDMZ     Demilitarized zoneDN      Dir[...]

CUCM Special Characters


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 informa[...]

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


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 AlgorithmsPulse 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 ScoresEach 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 ComplexitySome codecs require additional DSP processing power to handle the required computations [...]

Cisco Changing Position on CCIE Voice to CCIE Collaboration Transition


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! (image)

Cisco's CCIE Collaboration Certification Announced


Check your calendars folks!  You have until February 13th, 2014 to pass the CCIE Voice lab before it is retired! 

There's been quite a storm of criticism created following Cisco's announcement of the "new" Cisco CCIE Collaboration Certification AND the retirement of the 10 year old CCIE Voice certification.  This criticism is focused squarely on the decision to retire the CCIE Voice program through what Cisco calls "evolution".

This means that anyone that is currently a CCIE Voice can continue to be so, assuming they continue to pass any of the CCIE written exams every two years.  However, they will not be grandfathered and given the CCIE Collaboration Certification without actually taking the new CCIE Collaboration lab.

Given the relatively minor differences between IE Voice and IE Collaboration (20% maybe), I admit the logic of this change escapes me.  The IE Voice already had elements of collaboration within it - why not simply rename it to more correctly describe the contents of the certification along with some needed updates?

Mark Snow from INE has a very comprehensive review of the announcement that has generated quite a lively number of opinions/comments.  Check it out for some great analysis!

Want to voice your displeasure?  Take it to Twitter:  @LearningatCisco

Important Dates
CCIE Voice Written exam [350-030] is retired after November 20, 2013.
CCIE Voice Lab exam is retired after February 13, 2014.
CCIE Collaboration Written exam [400-051] is born November 21, 2013.
CCIE Collaboration Lab exam is born February 14, 2014.

Cisco IOS Voice Library


I'm not sure if this was recently added or if I've just never happened to find it, but this has to be the SINGLE MOST USEFUL page I've ever found while working on Cisco voice products:

Included in this extensive compilation:
  • Cisco IOS Voice and Telephony Highlights
  • Cisco IOS Command Reference
  • Physical and Virtual Voice Interfaces
  • Troubleshooting
  • Cisco IOS Call Control Technology
  • PBX Interoperability, ISDN, and Trunking
  • Voice Over Layer 2 Protocols
  • Video Applications
  • Fax, Modem and Text Support
  • Mobility Support
  • Cisco IOS Telephony Applications
  • Security
  • Additional Voice Applications
  • Related Documents
  • Tcl IVR and VoiceXML
  • Quality of Service

Click Here To Access >> Cisco IOS Voice Library

Screenshot:  This shows LESS THAN HALF of the information on this page!


IE Voice Alchemy by Kevin Wallace


Kevin Wallace has put out yet more great videos that assist the CCIE Voice student.  Having recently passed the exam himself, he's shared many great strategies that he personally used to pass his CCIE Voice lab exam in this insightful product, "IE Voice Alchemy".

He's made available the following free videos to give you a taste of the full wisdom you can get in the complete product, IE Voice Alchemy.  They're free, but still very useful to CCIE Voice lab students - I've heard from a few I know and they agree.

IE Voice Alchemy - The Bomb Run
allowfullscreen="allowfullscreen" frameborder="0" height="360" src="" width="640">

IE Voice Alchemy - Interview With A Proctor
allowfullscreen="allowfullscreen" frameborder="0" height="360" src="" width="640">

IE Voice Alchemy - Troubleshooting - "The Deal Breaker" allowfullscreen="allowfullscreen" frameborder="0" height="360" src="" width="640">

IE Voice Alchemy - In The Arena allowfullscreen="allowfullscreen" frameborder="0" height="360" src="" width="640">(image)

New Gateway Feature - Toll-Fraud Prevention in IOS 15.1(2)T


This post is a big warning for anyone upgrading a voice gateway to 15.1(2)T or later.  Without additional configuration, all inbound VoIP call setups will be blocked after the upgrade.  Yup.  Blocked.  Additionally, two-stage dialing is no longer enabled by default.

Per Cisco's explanation of the new Toll-Fraud Prevention Feature, a "trusted list" must be configured on the voice gateway so that the sources generating the VoIP call setups will be accepted.

Note:  If you have "session target" defined within dial-peers that you currently use, those calls will be accepted even if no "trusted list" is defined.

Debug Example - Blocked Call 

debug voice ccapi inout
%VOICE_IEC-3-GW: Application Framework Core: Internal Error (Toll fraud call rejected): 
IEC= on callID 3 GUID=F146D6B0539C11DF800CA596C4C2D7EF
000183: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallSetContext:
000184: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/cc_process_call_setup_ind:
>>>>CCAPI handed cid 3 with tag 1002 to app "_ManagedAppProcess_TOLLFRAUD_APP"
000185: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallDisconnect:
Cause Value=21, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)

Disable Toll-Fraud Prevention Feature

If you need to quickly return your router to it's previous functionality after an upgrade, take one of these two paths:
  1. Configure the router to accept incoming call setups from all source IP addresses.
    voice service voip
    ip address trusted list
  2. Disable the toll-fraud prevention application completely.
    voice service voip
    no ip address trusted authenticate

Restore Two-Stage Dialing After Upgrade

If two-stage dialing is required, the following can be configured to return behavior to match previous releases.

For inbound ISDN calls:
voice service pots
no direct-inward-dial isdn
For inbound FXO calls:
secondary dialtone


QoS Simplified: Cisco Catalyst 3560 and Cisco Catalyst 3750


Yet another great video from Kevin Wallace!

Over 1 hour and 45 minutes in length, this video is a fantastic review of QoS for all Cisco CCIE Voice and Cisco CCIE Route/Switch candidates.

From his youtube channel:

"A particularly challenging topic for many CCIE R/S and CCIE Voice candidates is the configuration of Quality of Service (QoS) on a Cisco Catalyst switch, specifically the 3560 for CCIE R/S candidates and the 3750 for CCIE Voice candidates. Fortunately, the QoS architectures of these switches are identical. So, this video, which seeks to simplify QoS theory and configuration topics, is applicable to both CCIE R/S and Voice candidates."

allowfullscreen="" frameborder="0" height="480" src="" width="640">(image)