Feb
22

If you recall from a past article I wrote about "Which Phone Should I Use?” – A Comparison of Models, you may remember from the graphic that support for remotely controlling SIP & SCCP IP phones in CME or SRST was coming approximately around Feb 14. Well, I now have a new graphic to replace that old one (after the jump):


That's right, the very talented and kind folks at Variphy, have just updated their Variphy Insight Remote Phone Control software to include support for SIP and SCCP phones connected to both CME and SRST (either traditional SRST or CME-as-SRST). So now there are literally no phones in our racks that cannot be controlled via our web-based remote phone control client!

Watch the new instructional video here!

Also, I am including just below, the small bit of additional configuration needed to make the Variphy Insight software work with CME for both SIP & SCCP phones. Please note that there is no additional configuration needed to make Variphy Insight work in controlling phones that have fallen back into traditional SRST or CME-as-SRST, due to the fact that they keep their Authentication URL string during fallback, and continue to authenticate via the CUCM (I speak to how exactly this works with INE's racks in the instructional video). Please also note from the below that there is no need for any "ixi" configuration, and I only threw it in below (in shutdown state), to reinforce that principal. In fact, this is what makes it possible to control SIP phones in SRST and CME, since the IXI subsystem currently only supports control over the SCCP CME server and its endpoints.

!
voice register global
url authentication http://177.1.254.3/CCMCIP/authenticate.asp
!
!
username variphy privilege 15 password cisco
ip http server
ip http authentication local
ip http path flash:gui
!
!
ixi transport http
shutdown
!
ixi application cme
shutdown
!
!
telephony-service
url authentication http://177.1.254.3/CCMCIP/authenticate.asp variphy cisco
!



We are in the process of updating all the voice racks in our data center to include this latest version of the client, so be patient as it may take a couple of days to get every rack updated with the new software.

Also stay tuned for some exciting news coming up in the very near future regarding the ability to use all of our Voice rack functionality with NO VPN connection!

Happy Labbing!

-Mark

Feb
12

Mark, I'd just like to say thanks again for the course. It was invaluable and played a huge part in my passing. I passed on my third go, last month just after the course. Without the tips and advice you provided in the bootcamp, I have no doubt I'd still be trying.

- Kevin Lloyd, CCIE Voice #27994

Congratulations Kevin!!

Feb
09

Lab's 1 & 2 from INE's current CCIE Voice Volume II Workbook have been completely re-written from the ground-up, and have now been pushed to all subscribers in their INE Members area.

These two new labs both contain video-based solutions, which walk you -- task-by-task -- through every step of the necessary configuration, along with plenty of live troubleshooting, just as we should expect when sitting for the actual CCIE Voice Lab exam.

For the first two labs alone, we have recorded over forty hours of video-based solutions.

We are in the process completely of re-writing every lab in the Voice Volume II Workbook, and will be releasing them and posting announcements here about each new lab as they are completed. We will also be releasing new CCIE Voice Deep Dives here shortly. So stay tuned for much, much more from INE's Voice program.

Happy Labbing!

Mark

Feb
01

In our series on CCIE Voice Troubleshooting and issues you may encounter during your lab exam, we last looked at problems with European MGCP GWs and Mobile Connect, and today we would like to continue that series by exploring the issues that you might run into in a number of situations related to logins to the Cisco Unified Contact Center Express product.

Perhaps one day you go to login to the UCCX Administration page, and you find that the user that either you, or someone else setup -initially in CUCM and told UCCX to use as it's administrative login- no longer works properly. You try and reset the password in CUCM (or possibly in Active Directory), and then try to login to the UCCX Administration page again, but alas, still no luck to be had. This issue could present itself in a number of situations, perhaps in self-study, during the actual lab exam, or very possibly even in a real-life scenario.

This video from our brand new Volume II lab video solutions will walk you through both the problem and quick solution to get you back into your UCCX, and able to re-link either the existing/should-work administrative user, or choose a new user for all future admin logins.

Happy Labbing,

Mark

Jan
27

The Cisco Unified Communications feature called Mobile Connect (also familiarly referred to as Single Number Reach) is truly a great feature of Unified Communications Manager, and can provide us with many efficiencies both in being able to be reachable just about anywhere, and in being able to be easily identified when placing inbound calls from our mobile phones into the CUCM cluster to our colleagues. As admins, we know that if we wish to have our users place calls from their mobile phones inbound to their colleagues inside the CUCM cluster, that we need to match up all or at least part of their inbound calling party number (CLID) to their CUCM Remote Destination. But what happens when what the carrier is sending CLID digits inbound to our IOS voice gateways that differs significantly from our Remote Destinations in CUCM, especially if we have truly embraced Cisco's push toward true Globalization in v7.0, v8.0 & v8.5?

The fact is that many, if not most European carriers (as well as many more all over the world) send CLID in through an ISDN PRI into the enterprise gateway with a preceding "0" as a courtesy  digit for easy recognition and ease in dialing back out, since this "0" is very commonly used as a carrier-recognized national dialing prefix. If we were speaking of the US and Canada, this "0" we are speaking of would be akin to dialing a "1" prior to the national number. Now in the US and Canada, if a carrier in the US sent CLID into a gateway with a "1" preceding any 10 digit number, this would work fine since the US/Canada country code also happens to be "1". However, the "0" preceding a variable length number is not valid in a true E.164 number format (e.g. If you dialed the phone number from outside of whatever country we were talking about, you would omit that preceding 0 from your dialed digits).

So what are we to do to get our inbound CLID to match our RD's?

That is exactly what we will explore here today in video format, as you watch a very small excerpt from our video-based solutions to one of the many new labs we will have in our new CCIE Voice Volume I & II workbooks.

Click here for the 25 minute video discussion on "The Trouble with European MGCP Gateways and Mobile Connect Inbound Calling Party Matching".

(BTW, if the video starts off with a bit of an echo, just hit CTRL-R to refresh the stream. And then stay tuned to this blog for some very exciting announcements about new formats for video solutions in the very near future)

Happy Labbing,

Mark

Jan
21

I have been asked for many years now, "What is the best phone to use when studying for the CCIE Voice exam?". And while this answer has changed over the years with the different blueprint requirements (with no doubt whatsoever it will change again come next blueprint), the answer for the past year and a half has been a tough one. You see, the actual CCIE Voice lab exam uses Cisco 7965G phones - which you certainly do not need to run out and buy, since all of the same features can be tested using a lesser model, say a 7961 or 2 hardware phone (only difference would be the background image - color and resolution). And certainly, without a doubt, using hardware phones --attached to an INE / GradedLabs Voice Rack using hardware EzVPN through a Cisco IOS router or Cisco ASA-- is hands down, the absolute best way to study.

However, not everyone has the capital outlay needed to achieve such a pristine study environment. In fact, when we at INE got together with another company to offer a discounted rate to students on some 'remote phone control' software --so that students could get the 'next best thing', namely controlling the same hardware phones without purchasing them-- the price still exceeded many of our clients' budgets. They would say (a bit incredulously) to us, "You mean I spent all this money on your great self-paced products, but  now I get surprised to find that I must spend more money just to make your racks work for my lab session??". And you know what? I tend to agree with them. It didn't seem all that fair at all. And then that company decided to raise their prices, which I understand -business is business- it just didn't fit with our needs for our students any longer. So now, after having been approached by every company in this comparison list to use their product, we settled with what we believe is far-and-away the absolute best among them, Variphy Insight, and let our clients use it for free. That's right, it costs you absolutely nothing to use our remote phone control software with our racks. And we have FIVE 7961 phones and one PSTN phone attached to every single CCIE Voice rack.  Not only is their product lightweight by the pure use HTML, but it can be used with any OS.

So what I have done is, after carefully evaluating many possible methods of hard-phone vs. softphone vs. remote phone control software, put together a side-by-side comparison of many of the more popular means of studying for the CCIE Voice exam. This list is by no means exhaustive, however it is a very fair look at each product and what it can offer you in terms of 1) Cost, 2) Features, 3) OS Support, and 4) Ability to Use One Client for everything you need to do in your CCIE Voice self-paced studies. I hope this helps some people, and please comment on things you like, dislike, or would like to see, and we will do our absolute best to accommodate you.

Voice Self-Study Phone Comparison

By the way, we will be releasing complete rewrites with exhaustive video solutions to our Volume I and II workbooks in coming weeks - a few will release in less than 2 weeks.

Oh, and stay tuned for some upcoming announcements soon that will make your self-studying even easier!

Happy Labbing,

Mark

Dec
13

Just wanted to throw out a quick reminder to all of you involved day-to-day with Cisco Unified Communications in some fashion. Tomorrow I will host a free vSeminar on configuring and utilizing Active Directory as a source of LDAP user synchronization and authentication with the Cisco UC architecture servers.

If you still haven't registered, you can do so right up until the webinar begins. To do so, simply click here and fill in your requested information at the bottom of the page.

In case you missed any previous vSeminars, be sure to check out the recent updates here.

Hope to see you tomorrow!

Nov
18

Hi all, after a lot of studying and dedication i passed the Voice lab yesterday on my second attempt!!!! I want to thank a lot to Mark Snow for coaching me during the 2 week bootcamp. He made me feel very confident that I could pass, while also answering all of my questions and always providing clear explanations of all the topics. I also would like to to say that his Deep Dives videos are the best and were a fundamental part of my study and success in passing. I fully recommend the Deep Dive modules and the 2 week bootcamp. Again, thank you a lot for all Mark!!!!

Eduardo Elizondo, CCIE Voice #27511

Congratulations Eduardo!! Share in Eduardo's Success! Get all 17 current Deep Dive modules for FREE (a $1495 value) when you purchase any upcoming 10-Day CCIE Voice Bootcamp by using discount code 27511 at checkout! Be sure to act on this generous offer fast as it will only be available until November 26!!

Nov
02

I took the Voice lab exam last week and passed! I would like to send a BIG thank you to Mark Snow at INE for not only providing excellent training during the two weeks of intense bootcamp, but for taking the time to personally answer all of my questions and provide in-depth explanations in the areas where I was not feeling confident. Mark is an amazing instructor who cares about every student in the class. If you want to make the most of your studies and feel extremely confident before going for your CCIE Voice attempt, I highly recommend attending Internetwork Expert's two week Voice bootcamp classes with Mark Snow!

Mark Holloway, CCIE Voice #27384

Congratulations Mark, CCIE Voice #27384!!

Share in Mark's Success! Save 20% on the upcoming CCIE Voice 10-Day Bootcamp Nov. 29 - Dec. 10 2010 in London, UK when you use discount code 27384! You can also share in his success with 30% off all self-paced material for any track, just use discount code NOV30 at checkout!

Oct
05

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

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

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

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

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

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

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

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

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

Here is the outline for the complete Deep Dive series:

Network Infrastructure

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

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

Unified Communications Manager

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Unified Communications Manager Express

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

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

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

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

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

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

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

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

Unified Contact Center Express

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

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

Unified Messaging

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

  • Unity Connection
  • Unity Express

Unified Presence

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

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

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

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