Posts Tagged ‘call-manager’


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!

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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!!

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Just wanted to update all our CCIE Voice rack renters out there that the new Variphy Remote Phone Control app that I blogged about a few days ago will go live here by the next rack rental session at 3pm PT. Information on how to use it can be found in the first few minutes of this video demonstration of the product.

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We are very pleased to announce that we have reached an agreement with a wonderful company called Variphy that will allow us to give all our remote Voice rack renting clients free remote control access over all of the 7961 phones on our CCIE Voice racks! Variphy Insight is the product, and it is one of the most brilliant inventory control, CDR, phone broadcast and remote phone control software that I have seen in quite some time.

Six months back, we added 3 7961 IP Phones to each rack, and offered a discounted remote control software, but I think that you all will agree that “free” is a slightly more affordable and enticing way of getting you closer to passing your CCIE Voice Lab than “discounted” is.

Oh by the way, we are updating our racks again, and this time adding 2 more 7961 phones to each one. This means you will have 2 7961 phones at the CorpHQ site, 1 7961 phone at the Branch1 site, and 2 7961 phones at the Branch2 site, as well as a PSTN phone – of course.

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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!

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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

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

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

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Many businesses globally – large and small alike – have been converting calls from routing over traditional PSTN carrier trunks – such as E1 & T1 PRI or CAS – to much lower cost, yet still high performance, SIP ITSP (Internet Telephony Service Provider) trunks for years now. INE is no different than your business with regard to this – we have been using SIP trunks in lieu some traditional PSTN calling for years now as well. In fact, in response to a US Federal Communications Commission sub-commitee’s exploration on “PSTN Evolution” in December 2009, a representative from the US carrier AT&T described the traditional circuit-switched PSTN as “relics of a by-gone era”, and said that “Due to technological advances, changes in consumer preference, and market forces, the question is when, not if, POTS service and the PSTN over which it is provided will become obsolete” – source: Reuters [emphasis mine].

The challenge however, becomes that every SIP ITSP carrier has a slightly different way of implementing these sorts of trunks, and each has different provider network equipment that you, the customer, must connect to, and interoperate (properly) with. If you are a large national or multinational business, you may for instance sometimes even connect to two or three different types of provider network equipment, between possibly having multiple contracts with multiple carriers, and even sometimes having to deal with different provider equipment within a single carrier’s network.

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IPMA is yet another well-known CCM application that you may encounter on your CCIE Voice lab exam. While IPMA Proxy mode is clearly a legacy approach to configure this application its still a topic you could see in the lab. Before we discuss the configuration steps, let’s take a quick overview of a simplified model for IPMA Proxy mode operations. Refer to the diagram for IP Phone extension numbers.

IPMA Proxy Mode

The whole purpose of IPMA proxy is to intercept calls going directly to manager’s IP Phone primary line (“1001”), and proceed them using IPMA configurable call routing logic – usually divert calls to the assistant’s so-called Proxy line (“1112”). In other words, calls placed to manager phone, gets re-routed to assistant’s “proxy” line. Of course, manager has some control of the call-routing logic from it’s IP Phone, using special set of softkeys (plus a Web-interface for advanced configurations).

Now the whole idea of Proxy mode is to put the IPMA application in the call routing path between a caller and the manager’s primary extension. To accomplish this goal, manager’s primary line should be isolated into a separate partition – let’s call it PT_MANAGER. No other IP Phone in the system should have direct access to this partition – their respective CSSes should not contain this partition. Let’s name the CSS used by all IP Phones in the system as CSS_DEFAULT.

Now recall that IPMA is a Java application running in Cisco Tomcat server. IPMA uses CTI interface to control various call-routing components in the CallManager. Specifically, a CTI Route Point should be created in the CallManager system, and IPMA application should take control of it. Next, a “wildcard” extension “100X” should be associated with the CTI RP line and placed in partition PT_INTERNAL – the default partition used for all IP Phone lines within the system (Well, the DocCD recommends using a separate partition for the CTI RP – and indeed, this is a more flexible approach. However, for the sake of the configuration speed, it makes sense to use the minimum set of partitions). The “wildcard” extension number is actually used in configurations where many managers with the primary extension numbers in range “100X” should be covered with the IPMA application. If you are providing call coverage for just one manager’s phone, you can use “1001” here. Also, you may want to set the CFNA number to “100X” or “1001” – this will provide call routing backup in a case when IPMA application would happen to fail.

With the above configuration, when any phone in the system calls “1001” – the manager’s primary line, the call gets routed to “100X/PT_INTERNAL”, and eventually hits the IPMA application. At this point, the IPMA application may want to direct the call to the manager’s real line – “1001/PT_MANAGER” – and this is why the CTI RP should have a special CSS assigned, which has access to PT_MANAGER partition. Let’s name this CSS as CSS_IPMA. As a minimum, CSS_IPMA should contain PT_MANAGER and PT_INTERNAL – since the IPMA may need to redirect call to some other internal extension. (Note that “1001/PT_MANAGER” precedes “100X/PT_INTERNAL” when using CSS_IPMA. This order resolves the ambiguity, even in case when one assigns number “1001” to CTI RP).

To complete the picture, recall the proxy line on assistant’s phone. This is where IPMA application direct calls to by default. Since the assistant may need to direct the call back to manager’s phone, this proxy line should be configured to use CSS_IPMA as the Line CSS. With this setup, the proxy number “1112” is placed in PT_INTERNAL partition (so the CTI RP can reach it) and is allowed to call the manager’s primary line directly. Of course, the primary line of assistant’s phone (“1002”) has no special Line CSS configured, and will therefore hit the IPMA application when calling “1001”.

Per the recommended design, you should also create two intercom lines on both manager’s and assistant’s IP Phones. An intercom is simple a line, which has auto-answer with speakerphone turned on. On the opposite side, you just add a speed dial to reach the intercom number. Thus, you need to intercom lines plus two speed dials to accomplish the intercom configuration.

Now let’s move to the actual configuration. While CallManager has a special built-in IPMA Wizard, personally I’d prefer not to mess with it – unless you’re absolutely sure about what you are doing – the wizard will modify your partitions and CSSes, and may do that in the way you don’t expect. Configuring IPMA proxy mode manually takes a little more time, but once you understand it completely, it won’t take that much. Plus, you get full control of your configuration. So it’s a good idea to create your own IPMA configuration checklist, and use it during your practice. Here is how a checklist may look like.

[Call Intercept]

1) Create Partitions & CSSes: PT_MANAGER & CSS_IPMA
2) Create CTI RP, assign extension number “100X/PT_INTERNAL” to it, set CSS_IPMA as the device CSS. You may also use “1001” extension to cover just one manger
2.1) Set CFNA to “100X” or to “1001” if you provide call coverage for just one manager. This will provide call backup if the IPMA application fails.

[IP Phones]

1) Create a new Button Template, say “3+3 7960” to allow more than two lines on an IP Phone. You will need this template for assistant’s phone, to accommodate three lines: primary, proxy and intercom.

2) Configure the Manager’s Phone
2.1) Set Softkey template to “Standard IPMA Manager”
2.2) Configure the primary line in “PT_MANAGER”
2.3) Add an intercom line, “*1001” and a speed-dial to “*1002” to reach the assistant
2.4) Create IPMA IP Phone service & subscribe the IP Phone to it (URL could be found on DocCD)

3) Configure the Assistant’s Phone
3.1) Set Softkey template to “Standard IPMA Assisant”
3.2) Set Button Template to “3+3 7960” (assistant needs extra lines)
3.2) Add a proxy line “1112/PT_INTERNAL” and set the Line CSS to “CSS_IPMA” for this line
3.3) Add an intercom line, “*1002” and a speed-dial to “*1001” to reach the assistant


1) Create a new user named “manager”
1.1) Allow it the use of CTI Application & CTI Super Provider
1.2) Associate this user with manager’s IP Phone

2) Create a new user named “assistant”
2.1) Allow it the use of CTI Application & CTI Super Provider
2.2) Associate this user with assistant’s IP Phone

3) Get back to “manager” user
3.1) Start the Cisco IPMA configuration dialog and disable automatic configuration
3.2) Configure the settings per your setup
3.3) Add a new assistant to the manager
3.4) Configure the assistant, matching proper primary manager’s line against the assistant’s proxy line

[IPMA Application]

1) Choose Service Parameters for Cisco IPMA Application
2) Configure CTI Manager IP Addresses (primary/backup). In simplest case just use your Publisher IP
3) Configure IPMA Application IP Addresses (primary/backup). In simplest case just use your Publisher IP
4) Set the CTI RP name for the IPMA application
5) Restart the Cisco Tomcat Windows Service or go to Tomcat manager interface at http://[IPMA server IP Address]/manager/list and restart the service there


1) Check that manager’s phone has IPMA softkey set on it’s screen
2) Install the Cisco IPMA Console Application and log in there as “assistant”
3) Place a call to manager’s primary line, ensure it get’s routed to the assistant phone, and pick it up from the IPMA console. Forward the call back to manager’s primary line
4) Configure from the manager’s phone to accept all calls and place a call to manager’s primary line once again

Making checklists for complex tasks is a must when preparing to CCIE Voice lab. The above list suggests a simplified manual approach to configure all IPMA application settings, in the order specifically optimized for speed. However, if you are pretty much comfortable with the IPMA Wizard, you can use it for your setup. Just make sure you performed a thorough verification after that.

The final note is about interaction of IPMA proxy mode with the voicemail system. Since we isolate the manager’s primary line in separate partition, we need to make sure MWI CSS is able to access it, in order to light the MWI lamp. Make sure you wont forget about it, since this may cost you some precious points.

Further reading:

Cisco IP Manager Assistant With Proxy Line Support

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