Welcome to the Cisco Networking Professionals Ask the Expert conversation. This is an opportunity to discuss CCIE Voice with Ben Ng. Ben is a Technical Assistance Center (TAC) Customer Support Engineer with the CCIE program at Cisco Systems, Inc. He joined Cisco in March 1999. He has been providing technical and escalation support to Cisco customers on WAN, LAN switching, Multi-Service, and AVVID technologies. Ben is CCIE certified in R&S and Voice. Remember to use the rating system to let Ben know if you have received an adequate response.
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Hi Ben, I have set up extension mobility on CM3.3 When I hit the "services" button on the IP phone it shows me the "login" and "logout" applications I created running the "hotel.aef" and the "hotelout.aef" scripts respectively. However when I hit either option from the IP phone I get some xml/html looking code displayed. Any suggestions as to what I am not doing right? I would be grateful for any suggestions. Thanks in advance.
If you have upgraded CallManager to 3.3, then check this link on how to update Extension Mobility:
Hi Dexter, sorry for the delay in getting back. I hope Partha's post has helped you figuring out the issue with Extension Mobility. When you are using CM3.3, Entension Mobility is native to the CM, there's no need to configure the CRA(S) scripts such as hotel.aef.
Also thanks Partha for jumping in on this one.
Cisco integrated the extension mobility serivce in CCM3.3. So you shut not install the cra with hotel.aef and hotelout.aef scipts. Look into the online help under extension mobility. There you can find how to configure this service in CCM3.3.
The official policy I've been told is that a product has to be available for 6 months before it's considered for inclusion in the lab. The clock probably hasn't even started on CallManager 4.0 yet, because it's still on new product hold.
Ben Ng, of course, would be able to give you the final word on this. Hi Ben! Loved the Packet interview. :)
A product must be released for at least 6 months before it becomes eligible to be included into Voice lab content. Therefore, the earliest time when candidates might see CM 4.0 in the Voice lab is sometime in the third calendar quarter of 2004.
Also, when there's any major software change in the Voice lab equipment, for example, CM 3.3 to CM 4.0 or IOS 12.2 to IOS 12.3, announcement will be posted on the CCIE page at least 2-3 months before the actual change take place.
Just want to check that if i sit the lab in Europe, i will only be tested on E1 material and not T1?
Just want to make sure.
No, you will be tested on _both_ T1 and E1 in the Brussells Voice lab, just like the current SJ Voice candidates are being tested on both T1 and E1.
Although Voice professionals in European countries are less likely to encounter Voice T1 configurations while performing their day-to-day responsiblities, they are expected to demostrate their understandings of T1 as a voice signalling protocol, just in case there's a need for them to configure such.
In summary, our goal is to recognize Voice CCIEs, not European Voice CCIEs or American Voice CCIEs. :-)
I think it's a good idea to test the european and the american systems.
I did the CCIP course years ago and we did not study very much about american standards (T1, North American Numbering Plan...).
At that time I thougt it's a good idea because I'm european and we don't have this stuff.
And guess what? Last year I was in New York and installed a VoIP system for one of our company locations. Pretty much the only trouble I had was with T1 and with NANP.
Do Plans exist for RTP to carry the voice lab ? Also are there any Cisco Approved Training Partners that are offering approved preparation classes. E.g CCIE BootCamp.
As of today, there's no plan to open the Voice lab in RTP. However, we do evaluate our expansion plan based on demand and other logistic concerns. If there's any changes to the current plan, we will for sure to announce it on our announcement page at:
Secondly, I am not aware of any CCIE Voice Bootcamp being offered by Cisco Approved Training Partners, although I have to admit that I am not very current on such class offerings. However, I know a few outside test preparation organizations already have Voice CCIE Bootcamps available.
I also invite inputs from others monitoring this thread to post on the subject if they have any additional information.
Hi Ben, I am about two months old in supporting a VoIP environment, which our two biggest headaches are ICD interfacing with IVR and fax machines. Could recommend some locations where I can find specific setups for the various fax machines in a VoIP environment? Do you have any recommendations on trouble shooting these critters? Thanks for the insight. Rob
Here is useful link which provides some very good information on faxing in an VoIP environment and related troubleshooting tips.
Also you might find the following NetPro discuss on fax and modem useful:
Is there a possibility that we have to modify registry settings on any server using the "regedt32" registry editor? (In the lab exam)
I like the idea of doing the Voice IE, but am overwhelmed by the number of different products it encompasses. Unless you work for a Cisco Partner or very large enterprise, getting a lab together is nigh impossible. Please can you pass along a request for more feature limited, downloadable (and installable on non-MCS platform) applications. Unity is perfect.
Then we just need an NM-HDV / 3550 / 7960 etc.etc. - still plenty of money for Cisco.
Sorry for the delay in getting back, I just realized my answer did not show up when I first submitted it. So let me try again.
First let me assure you that I will pass your request along. Next let me share my point of view on the subject of frequent candidates' concerns on the difficulty, financial or logistical, of pulling together a pratice lab.
The CCIE Voice lab aims at testing expert level knowledge in the IP Telephony field, these knowledge could be acquired via a number of ways, such as related work experience, study groups, home or rented pratice lab, bootcamps, and so on.
Although a personal practice lab is an important part of a candidate's preparation efforts, sometimes I see candidates spend a lots of efforts trying to collect the exact replica of the equipment in the Voice lab. IMHO, studying/preparation can start (and continue) with just a few devices, and the goal is to acquire working understanding of not only the configuration, but also how the proctocols work with each other, and how to troubleshoot and recover if something goes wrong. For the HW devices which are difficult to get hands-on play times, candidates could possibly rent equipment times from compnaies which provides such services.
VoiceRack.com will be offering online equipment rental
and self-study training specific to the Voice CCIE track
later this Spring.
You may find this to be an inexpensive alternative if you
think that you need to evaluate a full mock lab.
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I'm new with Unity and my question should be simple for You. Is there a possibility to set handlers and prompts to default values on cisco unity 4.0(3)?
Thank You very much for an answer
You can go to c:\commserver and run configmgr.exe, then choose "reset all default config settings", although I don't know why you want to do that....
Perhaps you could be more specific in your question on what you want to do?
Also, since the topic of this discussion thread is on CCIE Voice Track preparation, I would appreciate if participant could stay within the scope of this topic and direct all other technical queries to the various forums under NetPro's Voice and Video discussions at:
Currently one of my client network topology is a VOIP legency via Nortel Opt-11 PBX from California to HK, there curently using 4-digit dialing plan on the 3640 between these two offices, I want to deploy a better solution AVVID. What would you recommend?
With the limited information you provided, I think you're looking to move from the current VoIP toll-by-pass model to an end-to-end AVVID solution.
Please take a look at the following Cisco IP Telephony Solutions Reference Network Design where you could find differnt recommendations on AVVID deployment models and best practices based on differnt customer needs.
In an AVAYA ISDN phone network that utilizes digital phones (and some analog) is it possible to stop ANI information at the CISCO router? Currently if AVAYA systems have implemented ANI on the digital phone network, there is no way to stop sending that information on a per phone basis. If I were to institute an IP digital phone on the AVAYA system...would I then be able (maybe via access list on the router) to stop this particular piece of information from being sent along with the call so that the reciever's phone would NOT pick up ANI (Caller-ID) information?
May I ask where is the Cisco router in this picture? On voice enabled Cisco routers, it is possible to use translation-rule to change or mask the calling party information (ANI) before the call is send out the local voice or IP interface. If you're using Call Manager, you can use translation patterns to accomplish the same thing.
Let me take the liberty here to request participants to keep the content of their posts within the scope of this discussion, which is on CCIE Voice Track. You could direct your other technical questions to the appropriate threads in the Voice and Video discussions listed there:
I too am surprised at how few questions you're getting about the exam. Does this give you some assumption that the CCIE-V exam isn't popular or hard to achieve?
You see.. I have a feeling that the written is going to have a large portion of it about CCM and IOS GW questions, with added CRA/ICD. So with that, is there any way you can provide us a ball-park figure as to the percentage of CCM vs. IOS GW vs. ICD questions?
Also, what is your feeling about the lab itself? Do you think it needs improvement so to ensure the "student" is getting the concept of the actual exam in case they have to take it a second time?
Thanks in advance
Thanks for your questions. Looks like the participation is picking up and we have some good questions waiting in line. :-) I can't really use this evaluate the popularity of the CCIE Voice exam, although from a booking standpoint I see the demand is still strong.
About the written exam, unfornately I could not disclose the percentage of the technology areas which you requested, however, the written exam focus more on telephony and VoIP theories and the exam does aim at evenly distributing weight across all areas outlined in the written exam blueprint outlined here:
My feeling about the lab is that it is a challenging exam in terms of contents, covering some of the advanced technical configurations and integration of different components to offer a unique solution. As far as improvements, there are lots of room to improve and we constantly solicit feedback from candidates and act upon them. For example, when the lab is intially offered, there are concerns the length of the test being too long, so modifications are made to reduce the redundant contents.
In short, candidates are very encouraged submit their feedbacks after they received the results, or they can simply write to email@example.com and the email will be forwarded to us. And as I just mentioned we do value feedbacks and we do act upone them.
First, I apologize regarding this question being "outside the forum." This is my first post in any CISCO forum and I saw your expertise and thought you might be the best person to ask this question. Other colleagues of mine have worked on this particular problem and we really haven't found a good solution yet. I thought that the CISCO router path might be the best because of the ability of the CISCO routers to filter traffic at the packet level, which, I believe in this case would be the only way to do it.
Now..onto the scenario. There isn't at this point a CISCO router (or any router) inserted where this phone is. That's not really an issue either. One can be inserted at any time, it's just how to configure it once inserted that is my question. We don't currently use Call Manager and I don't believe we will either, only because we have a 'Call Manager' type software in place from AVAYA that works well for us. You mentioned translation patterns...could you expound on this please? Thanks