Re: [OSL | CCIE_Voice] CCIE Collab study group on facebook
Hi Wayne, I used ipexpert Racks during my preparation, does that mean I am a member. If the answer is yes, then how to join ? Thanks Sent from my iPhone On May 28, 2014, at 11:42 PM, Wayne Lawson waynelawson-...@ipexpert.com wrote: Yes - you will be a member. Regards, Wayne A. Lawson II Founder CEO - iPexpert CCIE #5244 / Emeritus :: World-Class Cisco Certification Training Mobile: +1.810.334.1564 :: Free Videos :: Free Training / Product Offerings :: CCIE Blog :: Twitter On May 28, 2014, at 4:05 PM, Shinobi CCIE shinobicr.c...@gmail.com wrote: Hello Wayne, I purchased the OWLE for the previous CCIE version unfortunately I did not pass. Does that make me a member? How do we do to get access to the Next Generation? I am not actively preparing for the lab but I hope to keep reading and helping the group if possible. Regards, Roger Carpio. On Sun, May 25, 2014 at 4:47 PM, Attila Rumy rumy.att...@gmail.com wrote: Hi Wayne, I'm looking forward to the Next Generation! In the meantime, could you please tell me when will the Collaboration workbooks be released? There was a post on the 14th of May about Andy writing a blog about the ETA's the next day, but since then there're no news. Thanks and regards, Attila 2014.05.25. 22:41 ezt írta (Wayne Lawson waynelawson-...@ipexpert.com): If you're a customer - if you've EVER bought a product - you're a member!! Regards, Wayne A. Lawson II Founder CEO - iPexpert CCIE #5244 / Emeritus :: World-Class Cisco Certification Training Mobile: +1.810.334.1564 :: Free Videos :: Free Training / Product Offerings :: CCIE Blog :: Twitter On May 25, 2014, at 4:16 PM, Daniel Gómez danie...@gmail.com wrote: Hello Wayne, Sounds great! Unfortunately I wasn't able to pass my CCIE Voice. I hope to be able to join the new community as I'n still preparing for CCIE Collaboration. Regards, Daniel Gømez. Enviado desde mi iPhone El 25/05/2014, a las 14:57, Wayne Lawson waynelawson-...@ipexpert.com escribió: Guys - I want you all to know that as part of our Next Generation relaunch - we will be shutting down the OSL lists, and launching a Member's Only Support Community. It will accomplish 2 things; - First, get you much quicker response from my instructors, support team, training advisors, student coordinators and me. - Secondly, spam and trolls will be eliminated. You guys are going to absolutely LOVE what we have planned and what we'll be launching when Next Gen goes live - it's less than 2 weeks away! Regards, Wayne A. Lawson II Founder CEO - iPexpert CCIE #5244 / Emeritus :: World-Class Cisco Certification Training Mobile: +1.810.334.1564 :: Free Videos :: Free Training / Product Offerings :: CCIE Blog :: Twitter On May 25, 2014, at 3:33 PM, Chris Avants cava...@gmail.com wrote: I thought that sounded so interesting, just must provide company details and have an email, lol! Has to be the same people trying different stuff to pedal their crap. On May 25, 2014 3:30 PM, Wayne Lawson waynelawson-...@ipexpert.com wrote: You're going to be banned from this group for life. This isn't a personal FB marketing avenue for you. Regards, Wayne A. Lawson II Founder CEO - iPexpert CCIE #5244 / Emeritus :: World-Class Cisco Certification Training Mobile: +1.810.334.1564 :: Free Videos :: Free Training / Product Offerings :: CCIE Blog :: Twitter On May 25, 2014, at 3:12 PM, Bashar Aziz bashar1a...@gmail.com wrote: Today I would like to announce publishing a new study group on Facebook, this group is a closed one, which means that you have to ask a permission to join and see posts. It is meant for professionals and experts in CCIE Collaboration field who preparing for the LAB solution, to join this this group you should meet the following requirements: 1. Linkedin profile 2. Business email address and website Regards, ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless
Re: [OSL | CCIE_Voice] Pause in speed dial
This feature can be found in cucm 9.x, I am not sure if it is there in CME 9 Sent from my iPhone On May 9, 2014, at 4:22 PM, Minh Dang dangquangm...@vnpro.org wrote: Group, We have a corporate directory that include extensions. Just wonder if i could insert pause when dialing out with speed dial. Our system include CME 9.x and IP Phones 7965s, 7975s, 7942s… Thank you in advance, Minh ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] UCCE Script: Transfer Call to IP Phone UCCE Script
Hi Dears, Thanks for the contribution , I was able to transfer the call to any extension within the call manager, by adding a dynamic label node on my ICM script, then add the needed static route in the CVP pointing back to CUCM. But the strange thing is, I am not able to transfer that call to any PSTN number, which should be the same way. the call routing to the PSTN from the CUCM is working fine Any Idea My friends ? Thanks Again On Mon, Apr 7, 2014 at 3:25 PM, John V. Casale john.v.cas...@gmail.comwrote: Andy is correct. Use a label node in the ICM script and send the call wherever you want (PSTN, IP Phone, etc.) Unless they added UCCE to the CCIE Collab (they didn't), not sure how anything related to UCCE is relevant on this list. Sincerely, *John V. Casale* Cell: (919) 371-8541 On Sun, Apr 6, 2014 at 11:07 PM, Andy Thanh tanthanh2...@gmail.comwrote: Hi, you can create a Label for a IP phone extension but you need to make sure to create the correct routing client label and you need to track where the label returned to. Let say, if there are a CVP system need to get the label, you need to crate dial plan on CVP and point back to CUCM. Andy On Mon, Apr 7, 2014 at 9:35 AM, Pavan K pav.c...@gmail.com wrote: As for the original question, it should be possible to return a label to whatever target on UCM in the ICM script. Am I missing something non trivial? On Apr 6, 2014 1:19 PM, Chrysostomos Christofi ch.christ...@logicom.net wrote: Hi If you have CUE you can achieve this task in the same way as into UCCX #Chrysostomos *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *CCIEing *Sent:* Sunday, April 6, 2014 9:28 AM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] UCCE Script: Transfer Call to IP Phone UCCE Script Dear Group Members, I would to ask about this feature ..After a time check (if the time after mid night) I have a UCCE system and I need to create a script that allow the call to be transferred to an IP Phone (Not Agent) just a number or PSTN number , the most important is This is not agent. As you may all know this is easy from the UCCX , but is that doable from UCCE ? Thanks ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
[OSL | CCIE_Voice] UCCE Script: Transfer Call to IP Phone UCCE Script
Dear Group Members, I would to ask about this feature ..After a time check (if the time after mid night) I have a UCCE system and I need to create a script that allow the call to be transferred to an IP Phone (Not Agent) just a number or PSTN number , the most important is This is not agent. As you may all know this is easy from the UCCX , but is that doable from UCCE ? Thanks ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] AXL servers and CUC
Why the guys keep getting not 100% in voicemail integration, what common mistakes happened Sent from my iPhone On Oct 25, 2013, at 12:42 AM, William Bell b...@ucguerrilla.com wrote: In this instance, best practice is a relative concept. Many applications leverage AXL to retrieve information and you can leverage that API to retrieve information on any node in the cluster. Applications that push configurations should be leveraging the publisher to do so. That said, I would think that you could send a SQL update, addphone, updateuser, etc. AXL command to a subscriber node. Of course, it would only succeed if the publisher node is on line. Anyway, my policy is to enable a secondary AXL server in the cluster if I have applications that are leveraging AXL to pull information. Like CUxAC, CUC, UCM IM/P, etc. CCX actually writes using the AXL API. Not that having a redundant AXL server would hurt but if I just have CCX and UCM, I typically go with a single AXL instance. IMO, you will not harm your cluster if you enable AXL on subsequent nodes in the cluster. You must have it on the first node (pub) if you are going to have it at all. In the IE lab, at least the current blue print, you will want to enable the AXL service on both Pub and Sub. If for no other reason than to avoid goofy issues during initialization of CUE in a scenario where CUE registers directly to UCM. HTH. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Oct 24, 2013, at 2:38 PM, probert...@gmail.com wrote: Hi, I have few questions related to AXL servers and CUC. Should CUC be configured to use both sub and pub as AXL servers? According to doc: Cisco AXL Web Service Activate on the first node only. Failing to activate this service causes the inability to update Cisco Unified Communications Manager from client-based applications that use AXL. But as far as I know CUC will not be updating anything on CUCM using AXL it is just used to read data during the user import. So since AXL can be activated on SUB can we use it with CUC? I know it works I just want best practice, if we don't than we have no redundancy. I guess this also applies to UCCX. Should we activate AXL on sub in the lab and should we configure CUC to use both AXL servers in the lab? Thanks! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME/B-ACD Documents in revamped Documentation Support page
But the question is that.. Does the search supported in the website in real lab? On Thu, Oct 24, 2013 at 3:03 PM, Todd Carswell tcar0...@gmail.com wrote: I noticed the same thing. From the main documentation page, just type Communications Manager Express in the Find section and it'll come right up. --Todd On Oct 24, 2013, at 7:51 AM, StefanoS stefan...@gmail.com wrote: Hello everyone. This is a silly question, maybe I'm too tired but I'll ask anyway. A couple of days before Cisco did a rearrangement in Documentation Support page. So for example the section for UCM documents went under Products Unified Communications Call Control, or phones under Products Collaboration Endpoints Phones etc. I've found some but I can't find the path for the CME category and B-ACD docs path anywhere. It's not under CUCM (Call Manager) in Call Control section as I was expecting. So where is it? Thank's in advance. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] MVA the right way to configure it
Can you past your config here to see what you did? Sent from my iPhone On Sep 28, 2013, at 11:40 AM, Bashar Aziz bashar1a...@gmail.com wrote: Why I am getting 0% in Voice Gateway and Signalling for the 6th time, 100% tested and worked, what is the trick ? Regards, On Fri, Sep 27, 2013 at 4:29 PM, sanity insanity networksanitytoinsan...@gmail.com wrote: Hi Guys, Thanks once again for your replies. @Lakshmish using your method of creating a seperate partition for RDP ( on the left side) and not having the SB PH1 have access to it . I noticed that when a call is made from PSTN ( with calling number 525) to 3300 and if we enter the pin and dial a number say 2001 ( internal) . The 2001 phone rings and the call can be answered. However the SBPH1 ( physical phone) is unable show that the 3001 line is active by showing a red light and therefore this does not appear to the requirement for MVA is achieved . What do you think? -MJ On Fri, Sep 27, 2013 at 2:11 AM, Lakshmish NS lakshmish...@gmail.com wrote: Hi MJ, Martin is right, I had issues with SNR after configuring the RD to 7 digits and setting the service parameter to complete match, MVA and SNR wouldn't go together. Martin however has proposed a new fix, you could try it. The workaround I used for this was to create an Application Dial Rule, which would certainly solve the issue. Cheers, Laksh On Tue, Sep 24, 2013 at 8:39 PM, Martin Sloan martinsloa...@gmail.com wrote: Hi MJ, 1) If you set the partial match to 7 digits and then configure your remote destination as a 10 digit number, you'll get a match if the ANI is either 7 or 10 digits since the match rule takes 'X' partial-match digits from the RD starting with the last number (2 in this case) and compares it to the ANI of the calling number, but the calling party number must be equal to or shorter in length than the configured remote destination, which is why it's good to just set your RD at 10+ digits if you're using partial match. Here are some scenarios and the outcome for partial match: Partial Match = True Number of Digits For Match = 7 digits Remote Destination = 972525 Calling Party Number = 525 Result = Match Partial Match = True Number of Digits For Match = 7 digits Remote Destination = 972525 Calling Party Number = 972525 Result = Match Partial Match = True Number of Digits For Match = 7 digits Remote Destination = 525 Calling Party Number = 525 Result = Match Partial Match = True Number of Digits For Match = 7 digits Remote Destination = 525 Calling Party Number = 972525 Result = No Match (ANI is longer than RD) When using Complete match, the ANI and RD have to be exactly the same. I like to make a call into SB from the PSTN phone prior to configuring SNR and I can quickly see what the ANI is, which is what I then make my RD. I had mentioned some buggy behavior with SNR though I never spent time working with partial match since when I heard about that issue I just stuck with complete match but I wanted to test my info above to make sure I wasn't sending incorrect info. It wasn't too hard to run into this buggy behavior. I found a workaround as well so I thought I'd share. When changing the Complete Match service parameter to Partial Match you get a screen pop that says to remember and set the Number of Digits for Caller ID Partial Match service parameter. The default for that parameter is 10 and the bug that I found is that on the initial change from default 10 to 7, the new setting does not take effect. After changing from 10-7 I started to make test calls and my CLID to SB PH1 was showing as the 7 digit ANI of the PSTN phone and not SB PHONE 2 3002 like it should. I dug around for a bit and tweaked a couple parameters and re-tested. The deal is that you have change Complete Match to Partial Match - Save then change Partial Match digits from 10 to 7 and Save again. 2) For this one if your service parameter is set to Complete Match and your ANI is 7 digits, just set your RD to the 7 digit number then use route patterns/xlations to manipulate as needed. 3) Not sure about that one. I've definitely seen conflicting information on certain things but I've realized that some of the training material is years in the making and when things are discovered or updated, maybe the old information is not or it's just floating out there. I can confirm that based on some recent experience with trusted trainers it was reiterated not to use partial match, maybe in part because of the issue that I hit today. Marty On Tue, Sep 24, 2013 at 8:19 AM, sanity insanity networksanitytoinsan...@gmail.com wrote: Hi Guys , Thanks a lot for taking time out to reply to my question. It was really helpful. I was trying to understand the difference between full match
Re: [OSL | CCIE_Voice] proctorlabs replication
Did you try to disable csa? On Thu, May 23, 2013 at 8:21 PM, Karen Johnson karen.johnson...@yahoo.cawrote: hi all, I used auto phone register in proctorlabs. When UCM group start with SUB, phone never registered. But when I move PUB to 1st Server in Group, phone registered fine. And I also checked in Cisco CallManager Reporting for DB summary: replication look good 2 and CLI also look good any idea what wrong and what is command to check in this case? tks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] cups best practice
I do not set to the exam yet.. but here are below the steps I use to configure my CUPS: CUPC Configuration (Softphone) Form CUPS Side : - Application -- Settings - Application -- CUCP -- add Voice Mail Server - Application -- CUCP -- add Mail Store (default 143) - Application -- CUPC -- add Voicemail Profile , then add users to this profile - Application -- CUCP -- add CTI Gateway (should be created automatic when start services ) - Application -- CUCP -- add CTI Gateway Profile Form CUCM Side : - Add CUPC soft phone: Device -- add Cisco Unified Personal Communicator with the name UPCUSERNAME Desk phone Configuration - CUCM -- Associate device to end user (End user configuration page-- add the device in the Device association list ) - CUCM -- Specify Primary extension for the end user (the same Ext on his device) - CUCM -- Add the user to the groups (Standard CTI enables CUCM end user) - CUPS -- Assign user to the CTI Gateway profile (the one that relative to the user's phone DP) - Login/logout from the CUPC and test. On Thu, May 23, 2013 at 4:54 AM, Karen Johnson karen.johnson...@yahoo.cawrote: hi experts, can anyone share how to config desk mode and soft mode best practice for CUPS in exam? I can't figure out why did not got point for cups even it is working fine. tks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] LAN QOS
HI All, I am really struggling with QOS I can understand what this command means : mls qos srr-queue output cos-map queue 1 threshold 3 5 How to use it ??? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Can't get to proctorlabs
It is working just now Sent from my iPhone On May 14, 2013, at 12:50 PM, Josh Petro josh.pe...@gmail.com wrote: Hi Randall, I also can get to their site and I'm able to login. I don't have any rack time now, so I obviously cannot check that far, but I was able to get to their site. The site came up very slowly, just FYI. Josh On Mon, May 13, 2013 at 10:36 PM, Randall Saborio ill2...@gmail.com wrote: Hi, Anyone can just give it a quick check and let me know? On Mon, May 13, 2013 at 7:15 PM, Randall Saborio ill2...@gmail.com wrote: Hi guys, Just checking if is it my provider or proctorlabs problem. Can't open up proctorlabs.com and I am missing my session already :( -- Randall da ill Saborio CCIE Voice Wannabe #10054675811 -- Randall da ill Saborio CCIE Voice Wannabe #10054675811 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Calling Name for MVA
Hi all, I had the following problem , when I make a call from remote Destination to the MVA IVR number, then to internal extension, the calling name for the configured in the line for the remote destination profile does not appear on the internal phone when it answered the call, this happen only when I use the enable inbound faststart with my h323 gateway (where the MVA configured) . If I unchecked the inbound faststart the calling name appear. any idea ? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] ssh client
So how to increase the buffer :) Sent from my iPhone On May 15, 2013, at 1:47 AM, Barrera, Hugo hugo.barr...@nexusis.com wrote: This dang client cost me about 20 min in the Lab because I didn't know how to increase the buffer. Regards, Hugo On May 14, 2013, at 3:32 PM, Bill whl...@gmail.com wrote: I think it is an old version of secure CRT and not one easily found on the web. I think something like version 3 or 4 but I really did not worry about that, I use the current version and it works similar but don't expect much more that very basic interface Sent from my iPad On May 14, 2013, at 5:17 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: I think it should be v2 however I am not quite sure On 14 May 2013 15:07, Barrera, Hugo hugo.barr...@nexusis.com wrote: Anybody know what version of ssh client that is in the real lab on the CUPC Test PC? - Hugo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Unity Connection Unity Express Ports Region Interregion Relationship
You need transcoder on the hq side Sent from my iPhone On May 2, 2013, at 5:32 AM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Dear Experts, I'd like to ask when I configure the Regions between HQ , SB and SC Usually for Interregion relationship is G729 Codec is used while for Intraregion we use G711 Codec. So , In case of the Unity Connection and Unity Express. I wonder if i should apply the same rule on them? On Unity Connection it has Device Pool and usually you apply HQ for it. So When SB communicates with unity then is it should be G729 What is your recommendation of how to make it in the test? Thanks, Hesham ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Control IP Phones while in SRST
Hi all, when IP Phones goes to SRST mode.. (normal call-manager-fallback) how I can see that phone, how to control it, I need to make calls from that phone.. While the phone is register with call manager I can control it via phoneview.. but when it went to SRST mode , how to control the phone ?? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] G729 Only for UCCX
Step 1: i think call manager will generate all type of codec when you upload the file, so you can upload the g711 version copy. Step 4: you need to select both codec as the g711 will be used in case the call happened between 2 end points in same region Sent from my iPhone On May 9, 2013, at 12:09 AM, Barrera, Hugo hugo.barr...@nexusis.com wrote: Hi, I have put together this list from labbing and I wanted to get feedback if you think my steps are in the right direction or I entirely missing something hear? Step 1. On uccx grab ringback.wav file (shouldn't we be grabbing the file from the g729 folder and not the g711 folder?) Step 2. Upload this wav file to both cucm servers Step 3. Apply this file as the network on hold audio source thru uccx for CTI Ports and CTI-RP's Step 4. Go into the IPVMSA service and ensure both g711 and g729 codecs are selected (Or should I be selecting g729 only?) Step 5. Restart IPVMSA service on both cucm servers Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] VM Message button gives No services configured
Did you configure the vicemail profile?? How about your voicemail services , double check if it is exists under Device-- device settings -- phone services If it is not found you have to create it Sent from my iPhone On May 8, 2013, at 2:37 PM, Vikky Kumar vikkyne...@gmail.com wrote: Hi Experts, I am getting no services configured when i press VM Message button Please advise. Regards, Vikky ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] IP Expert Phone view stop working
Hi All, does any one has any idea about the phone view used by ip experts.. I was working normal, when it stop respond to my transactions.. I am able to see the phones from call manager.. but a I am no longer able to make any transaction on the phone, it always give me failure!!! any idea. any other software to control my ip phones in the racks ??? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Debug ISDn q931
Hi all, Some times when I debug isdn q931 , I am not able to see the calling number on the debug.. only the called number appear.. any idea ??? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Sending Calling Name to PSTN
For mgcp, from rote pattern configuration , you have to allow sending the calling name. For h323 under interface serial configuration ( for the used pri) You have to use the commands Isdn outgoing ie display Isdn outgoing ie redirecting-number Sent from my iPhone On May 7, 2013, at 1:44 PM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: Hi Experts, How can i send Calling name to PSTN over MGCP and H323 Gateway? thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUE signaling QOS on switch
Dears, So the above access list is correct ?? CUE signaling traffic only used this port ? 2748 Is there any document showed the used port for the CUE traffic ?? On Thu, Apr 18, 2013 at 12:46 AM, Jack Kamina kamina.j...@yahoo.com wrote: on one of the practice lab the need is to police the signaling packets to and from CUE inbound into the HQ switch to 32 kbps and then remark the DSCP to 0. I built up the config below but dont see any packets matched on the show policy-map interface command. CUE IP is 10.1.6.253 . CUCM IP is 10.10.210.10 (pub) and 10.10.210.11 (sub) .is the access list built correctly? access-list 110 permit tcp host *10.1.6.253* any eq 2748 ! class-map match-all CUE-SIGNAL match access-group 110 ! policy-map CUEMAP class cti-qbe set dscp af31 bandwidth 20 ! interface Fa0/1/0 description HQ-ROUTER-INTERFACE service-policy input CUEMAP mls qos mls qos map cos 0 8 16 24 32 46 48 56 mls qos map policed-dscp 24 26 to 8 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Translate DSCP values to Numbers
Hi all, Is there any document tell how to translate the DSCP values for hexadecimal to number i.e DSCP EF == 46 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CEM as SRST, SiteC
Thanks So much, your input is really helpful On Mon, Apr 22, 2013 at 7:22 AM, Suresh Bhandari bring...@gmail.com wrote: srst mode auto-provision all displays all learned ephones and DNs in your running configuration in SRST mode. You can tweak the configuration as per requirement. It also preserves those learned ephones and DNs even after the gateway re-registers to the CUCM. srst-mode auto-pro none will not display/store the learned DNs, so no chances of tweaking of ephones or DNs to meet some requirement. IPExpert has a very good set of articles on the topic of your interest: https://www.ipexpert.com/Cisco/CCIE/Voice/Free-Resources Check the High Availability section. You can find the different types of HA, and their differences, as well. HTH On Mon, Apr 22, 2013 at 2:19 AM, CCIEing aboaz...@gmail.com wrote: Dear All, In case the HA requirements was to use SRST, then which command to use Srst mode auto-provision all or Srst mode auto-provision none ? and what is the difference between them , Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CEM as SRST, SiteC
Dear All, In case the HA requirements was to use SRST, then which command to use Srst mode auto-provision all or Srst mode auto-provision none ? and what is the difference between them , Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Full Lab, More than one question
Thank man for your input. Appreciate it On Thu, Apr 18, 2013 at 9:04 AM, Chrysostomos Christofi ch.christ...@logicom.net wrote: Hi ** ** Answers below with red ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *CCIEing *Sent:* 18 April 2013 03:17 *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] Full Lab, More than one question ** ** Hi All, ** ** Today I tried to make a full lab , it was my 1st try to do all tasks in 8 hours session! ** ** the following questions appeared : ** ** 1-Do i need to enable the auto registration on both CUCM servers, including the starting/End DN information ? No its ok with only one cucm server ** ** 2- When you use the auto registration method to register your phone, what exactly define which device pool the new registered phone will belong to The default device pool. Then edit to the correct. Note that sometimes you get an error with the auto registration so you have to disable all the security services at both cucms and ADD the phones manually ** ** 3- What can be checked if SiteB Routed was not synchronized wirt HQ-Router, my router stay un-synchronize Which interface in HQ router is the ntp master? If is the loopback then try to add ip ospf network point-to-point into the HQ Loopback interface** ** ** ** 4- When it comes to any update of the Ringlists, do I need to restart TFTP service in both servers, and does it mandatory to upload the Files to both TFTP? Yes for both questions And Finally, I really shocked how much time the full lab needs, I did not finish 50 % of the lab in the 8 hours, I really need your advise regarding the time challenge !! The best choice is the device base approach. Search into YouTube to understand the process Best Regards ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] IPexpert SiteB DID's
Hi all, For the seek of solving the MVA task for siteB, a DID number needed. How to know the range of DID's for the T1 PRI used with Site B? Appreciate your input ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] HQ incoming calls Filed until I reload the router
Hi all, Very strange thing happen 2 times during my practice to lab, the calls to hq failed until i reload the router before reload, ISDN status was ok mgcp gateways was registered and everything seems to be fine, the call arrive the gateway but the phone does not ring, then call dropped .. Only when I reload the router of HQ things come fine!!! DO I am hitting a bug or what ? What do you think Guys ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] HQ incoming calls Filed until I reload the router
Yes On Fri, Apr 19, 2013 at 2:57 AM, Bill Lake whl...@gmail.com wrote: Was this an MGCP gateway? Sent from my iPhone On Apr 18, 2013, at 6:24 PM, CCIEing aboaz...@gmail.com wrote: Hi all, Very strange thing happen 2 times during my practice to lab, the calls to hq failed until i reload the router before reload, ISDN status was ok mgcp gateways was registered and everything seems to be fine, the call arrive the gateway but the phone does not ring, then call dropped .. Only when I reload the router of HQ things come fine!!! DO I am hitting a bug or what ? What do you think Guys ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SiteB Phones are not talking IP Address
Hi Josh, You mean the voice sub interface ? And the native vlan will be the data vlan?? Sent from my iPhone On Apr 17, 2013, at 3:33 AM, Josh Petro josh.pe...@gmail.com wrote: Also, to echo Bills point number 2, make sure the encapsulation dotq1 xx native command is configured on the router. That burned me twice early on. Josh On Monday, April 15, 2013, William Bell wrote: I'd check: 1. DHCP snooping on the switch (sh ip dhcp snoop) 2. Dot1q trunk on HQ-RTR interface (on the switch). Make sure that all appropriate vlans are allowed on the trunk link and that native VLAN lines up (1 is default). 3. Ensure VLANs are provisioned correctly, assigned to the right interfaces, and active (sh vlan b) 4. Double check scope config on CUCM Pub. Check each parameter. If the above check out then I'd restart the DHCP service on the Pub. If that didn't work, I would do the following on the phone: 1. Settings key 2. **# to unlock 3. Press more softkey when it pops up 4. Press Erase softkey -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Apr 15, 2013, at 8:16 PM, CCIEing wrote: Dear all, I am using proctorlabs racks, My siteB phone are not talking IP addresses from CUCM-PUB DHCP. I have configured the switch port connected to the IP phone with the correct access/voice vlan informaation. I also apply the IP hdcp helper-address command on the voice-vlan interface on the router, and pointed to the IP address of CUCM-PUB. When debuging IP dhcp server events/packets the router show the following messages : Apr 16 00:09:53.946: DHCPD: Finding a relay for client 0100.12d9.78ef.01 on interface Vlan240. Apr 16 00:09:53.946: DHCPD: Seeing if there is an internally specified pool class: Apr 16 00:09:53.946: DHCPD: htype 1 chaddr 0012.d978.ef01 Apr 16 00:09:53.946: DHCPD: remote id 020a0a0ac90110f0 Apr 16 00:09:53.950: DHCPD: circuit id Apr 16 00:09:53.950: DHCPD: there is no pool for 10.10.201.1. Apr 16 00:09:53.950: DHCPD: setting giaddr to 10.10.201.1. Apr 16 00:09:53.950: DHCPD: BOOTREQUEST from 0100.12d9.78ef.01 forwarded to 10.10.210.10. Any Idea Please ! Thanks in advance ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Full Lab, More than one question
Hi All, Today I tried to make a full lab , it was my 1st try to do all tasks in 8 hours session! the following questions appeared : 1-Do i need to enable the auto registration on both CUCM servers, including the starting/End DN information ? 2- When you use the auto registration method to register your phone, what exactly define which device pool the new registered phone will belong to 3- What can be checked if SiteB Routed was not synchronized wirt HQ-Router, my router stay un-synchronize 4- When it comes to any update of the Ringlists, do I need to restart TFTP service in both servers, and does it mandatory to upload the Files to both TFTP? And Finally, I really shocked how much time the full lab needs, I did not finish 50 % of the lab in the 8 hours, I really need your advise regarding the time challenge !! Best Regards ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Lab Exam Speed Strategy
Dear All, In the devise base strategy which is better to start with the CUCM and the applications or to start with the Router devices ?? I believe to start with CUCM is better that allow the phones to register earlier, as all your testing then will depend on it. Appreciate your feedback and advice On Wed, Mar 27, 2013 at 2:40 AM, Dane Warner dwar...@epochuniversal.comwrote: To All, ** ** I took my second attempt on Monday, March 25 and did not pass. I was hoping for some insight on concrete suggestions to get faster. I didn’t get hung up on any one task, I seemed to keep moving forward and tried to type as fast as I could, using CLI shortcuts, etc. I used the device-based methodology and I feel pretty confident of my technical knowledge. Yet I didn’t even get to many tasks at all, I would have needed another 2-3 hours to complete all tasks. I hear of candidates completing all tasks in 6-7 hours, which means I would need to become twice as fast as my last attempt. It almost sounds insurmountable. Do I need to take typing classes? ** ** Any recommendations that don’t break the NDA would be greatly appreciated. ** ** Regards, ** ** *Dane Warner, CCVP* *Sr. Network Engineer* *Epoch Universal, Inc.* *(909)226-0755* *dwar...@epochuniversal.com * *[image: Epoch_Logo_Smaller_Transparent]* ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com image001.png___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] SiteB Phones are not talking IP Address
Dear all, I am using proctorlabs racks, My siteB phone are not talking IP addresses from CUCM-PUB DHCP. I have configured the switch port connected to the IP phone with the correct access/voice vlan informaation. I also apply the IP hdcp helper-address command on the voice-vlan interface on the router, and pointed to the IP address of CUCM-PUB. When debuging IP dhcp server events/packets the router show the following messages : Apr 16 00:09:53.946: DHCPD: Finding a relay for client 0100.12d9.78ef.01 on interface Vlan240. Apr 16 00:09:53.946: DHCPD: Seeing if there is an internally specified pool class: Apr 16 00:09:53.946: DHCPD: htype 1 chaddr 0012.d978.ef01 Apr 16 00:09:53.946: DHCPD: remote id 020a0a0ac90110f0 Apr 16 00:09:53.950: DHCPD: circuit id Apr 16 00:09:53.950: DHCPD: there is no pool for 10.10.201.1. Apr 16 00:09:53.950: DHCPD: setting giaddr to 10.10.201.1. Apr 16 00:09:53.950: DHCPD: BOOTREQUEST from 0100.12d9.78ef.01 forwarded to 10.10.210.10. Any Idea Please ! Thanks in advance ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Back to back session IPexperts racks
Dears,, When I reserve a back to back racks sessions , does the pod number changed? does my configuration removed one I go from one time session to the next one Appreciated ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] 7941 won't take the siteC router as SRST
Hi all, I have 2 phones connected to my SiteC switch, both of them configured to be in Site device pool. when I drop the WAN connection between SiteC and HQ, only my 7960 phone goes in SRST . but my 7941 phone does not .. when i try to check the phone url , it does not show that information of SRST ip !! any idea .. Appreciated ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Forwarded Routing Rule in CUC
Dear All, I have an issue with CUC -- call routing -- forward routing rules, when add new rule with condition Forwarding Station = 0998 where 0998 is DN on CTRP that has the CFW_ALL to voice mail it does not work at ALL any help? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] IP DHCP failed assigning IP's
Hi all, I configured a DCHP pool , ip phones failed to get IP's when debug the IP hdcp server event and packets I got the following error *DHCPD: Allocate an address without class information (177.3.11.0)* *DHCPD: subnet [177.3.11.1,177.3.11.254] in address pool sitec is empty.* *DHCPD: Sending notification of ASSIGNMENT FAILURE:* * DHCPD: htype 1 chaddr 001d.45e8.a5e6* * DHCPD: remote id 020ab1030b01000b* * DHCPD: circuit id * *DHCPD: Sending notification of ASSIGNMENT_FAILURE:* * DHCPD: due to: POOL EXHAUSTED* * DHCPD: htype 1 chaddr 001d.45e8.a5e6* * DHCPD: remote id 020ab1030b01000b* * DHCPD: circuit id * Here is my configuration : *ip dhcp excluded-address 177.3.11.1 177.3.11.25* *ip dhcp excluded-address 177.3.11.26 177.3.11.254* *!* *ip dhcp pool sitec* * network 177.3.11.0 255.255.255.0* * default-router 177.3.11.1 * * option 150 ip 177.1.10.20 177.1.10.10* * * Any Idea Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] IP DHCP failed assigning IP's
OH what a mistake !!! I updated my config to the correct one *ip dhcp excluded-address 177.3.11.1 177.3.11.25* *ip dhcp excluded-address 177.3.11.28 177.3.11.254* * * * * *:)* On Sat, Mar 30, 2013 at 2:22 PM, CCIEing aboaz...@gmail.com wrote: Hi all, I configured a DCHP pool , ip phones failed to get IP's when debug the IP hdcp server event and packets I got the following error *DHCPD: Allocate an address without class information (177.3.11.0)* *DHCPD: subnet [177.3.11.1,177.3.11.254] in address pool sitec is empty.* *DHCPD: Sending notification of ASSIGNMENT FAILURE:* * DHCPD: htype 1 chaddr 001d.45e8.a5e6* * DHCPD: remote id 020ab1030b01000b* * DHCPD: circuit id * *DHCPD: Sending notification of ASSIGNMENT_FAILURE:* * DHCPD: due to: POOL EXHAUSTED* * DHCPD: htype 1 chaddr 001d.45e8.a5e6* * DHCPD: remote id 020ab1030b01000b* * DHCPD: circuit id * Here is my configuration : *ip dhcp excluded-address 177.3.11.1 177.3.11.25* *ip dhcp excluded-address 177.3.11.26 177.3.11.254* *!* *ip dhcp pool sitec* * network 177.3.11.0 255.255.255.0* * default-router 177.3.11.1 * * option 150 ip 177.1.10.20 177.1.10.10* * * Any Idea Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CUCM Call Routing
Hi again for all, I have question regarding the exam, in the CUCM call routing section : If the question does not clearly mention that the Called Party Type and plan is required in some parts of the call routing points.. Is it better to configure the them (call type and plan) or to leave them on the default configuration ? Your input is appreciated ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUCM Call Routing
Thanks Justin, I will adopt your strategy On Sat, Mar 30, 2013 at 9:15 PM, Justin Carney justin.s.car...@gmail.comwrote: I don't know how the lab is graded, but I first answer the requirements of the question, which sometimes states to set proper plan/type and sometimes unknown, then for all other call flows that don't specify I set the proper plan/type for those. I do this for both ani and dnis. There are two reasons why I do this - first, I type all the CLI in notepad and configure all the routers at the beginning of the lab (after taking basic notes on gw type, # of channels, etc) and then I go through the gw and call routing sections and modify as needed. Second, it can make reading debugs a little easier as I am used to verifying plan/type for all calls and I only need to make a note of which calls require unknown as the exceptions. Hope this helps... -Justin On Mar 30, 2013 2:04 PM, CCIEing aboaz...@gmail.com wrote: Hi again for all, I have question regarding the exam, in the CUCM call routing section : If the question does not clearly mention that the Called Party Type and plan is required in some parts of the call routing points.. Is it better to configure the them (call type and plan) or to leave them on the default configuration ? Your input is appreciated ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Lab Exam Speed Strategy
Hi Jamie, Would you please explain this more : ***You have to setup most devices with little or no prior configuration, there are things that cannot change. Know these things and practice them over and over so you do not have to think about them* Thanks in advance On Wed, Mar 27, 2013 at 9:00 PM, Jamie Parr (jamparr) jamp...@cisco.comwrote: First attempt I was very slow – did not use the device based approach, did not finish all tasks. Second I was much faster – using the device based approach, finished with 3 hours to test. Third attempt I finished with more than 3 hours to test and pick up the issues – Passed ** ** My advice: **· **I found the more I practiced the faster I got, practice practice practice **· **Use notepad to write all your device configs first, you can copy and paste large sections of config saving a lot of time **· **Do not be so strict to the device based approach, use it as a base and create your own hybrid **· **You have to setup most devices with little or no prior configuration, there are things that cannot change. Know these things and practice them over and over so you do not have to think about them **· **Persevere, it’s not easy and it sucks most of the time but you will get there ** ** Hope this helps ** ** *Jamie Parr* CCIE #38633 (voice) Engineer - IT jamp...@cisco.com Phone: *+44 20 8824 2641* Mobile: *+44 7590622049* ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Dane Warner *Sent:* 26 March 2013 23:41 *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] Lab Exam Speed Strategy ** ** To All, ** ** I took my second attempt on Monday, March 25 and did not pass. I was hoping for some insight on concrete suggestions to get faster. I didn’t get hung up on any one task, I seemed to keep moving forward and tried to type as fast as I could, using CLI shortcuts, etc. I used the device-based methodology and I feel pretty confident of my technical knowledge. Yet I didn’t even get to many tasks at all, I would have needed another 2-3 hours to complete all tasks. I hear of candidates completing all tasks in 6-7 hours, which means I would need to become twice as fast as my last attempt. It almost sounds insurmountable. Do I need to take typing classes? ** ** Any recommendations that don’t break the NDA would be greatly appreciated. ** ** Regards, ** ** *Dane Warner, CCVP* *Sr. Network Engineer* *Epoch Universal, Inc.* *(909)226-0755* *dwar...@epochuniversal.com * *[image: Epoch_Logo_Smaller_Transparent]* ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com image001.png___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] MVA functionality
Hello Friends... I have the following setup, I am not sure if the will be suitable to enable the MVA feature ! I have CUCM cluster, but his CUCM cluster has no voice GW or DID .. but this CUCM cluster has Inter-cluster trunk to another CUCM cluster which has the DID numbers ? Can I configure the MVA for this setup.. Appreciate your input. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Outgoing Calls via T1 failed
Hi, Changing the selection to top down solve the issue :) thanks On Sat, Mar 23, 2013 at 2:27 AM, ikizoo4 kwon ikiz...@hotmail.com wrote: use Topdwon for channel selection order in GW -- Date: Sat, 23 Mar 2013 00:52:10 +0300 From: aboaz...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Outgoing Calls via T1 failed Hi all, After configuring my HQ GW as MGCP, then configure my T1 to register with cucm , I was govern by the lack of the DSP resources, which force me to define only 16 channel out of the 23 on my pri-group under the T1 controller configuration !! here is the config : controller T1 0/0/0 pri-group timeslots 1-16 service mgcp Then I faced a problem with my outgoing calls , the calls was dropping due to the cause *Requested circuit/channel not available* My Question here, as there is 16 channel in my Pri-group are already configured, why all calls get dropped with cause of non availability of the resources, Why not to use one of the available channels (1-16) Appreciate your help here is below the output of debug q931 for one of my outgoing calls: *SDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref = 0x000F * *Bearer Capability i = 0x8090A2 * *Standard = CCITT * *Transfer Capability = Speech * *Transfer Mode = Circuit * *Transfer Rate = 64 kbit/s * *Channel ID i = 0xA98390 * *Exclusive, Channel 16 * *Display i = 'HQ PH1' * *Calling Party Number i = 0x2181, '7772022001' * *Plan:ISDN, Type:National * *Called Party Number i = 0x81, '911' * *Plan:ISDN, Type:Unknown* *ISDN Se0/0/0:23 Q931: RX - RELEASE_COMP pd = 8 callref = 0x800F * *Cause i = 0x82AC1810 - Requested circuit/channel not available* ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Outgoing Calls via T1 failed
Dears, Thanks for your input , I'll try the suggested and let you know.. but do you think changing the value of the Service Parameters -- CCM-- change B-channale status to = s0/su0/...@ VGname =. affect my configuration ? Regards On Sat, Mar 23, 2013 at 3:59 AM, Justin Carney justin.s.car...@gmail.comwrote: Your debug output has a few clues...but I can't recall offhand if channel 16 in that debug starts at 1 (meaning this is the 16th channel) or 0 (meaning this is the 17th channel). Do inbound calls from pstn work? If yes, its more likely the second option. In the first case, it would appear your issue is on the pstn side. Run show ISDN status and layer 2 should show multiple frame established and layer 3 should show ccm-manager (or similar). If however layer 2 shows tei assigned try the following: Mgcp bind media source lo0 Mgcp bind control source lo0 (Paste those commands twice) Int s0/0/0 No ISDN bind-l3 ccm ISDN bind-l3 ccm No mgcp Mgcp Show ISDN status (Ensure you see multi frame established) Also type show ccm and ensure the gw is registered to your cucm. If not, make sure that your hostname on the router matches what you have in cucm. If you have IP domain-name ipexpert.com in your config then you need to use the fqdn in cucm, such as r3.ipexpert.com. however if you don't have a domain name on the router then you should just have the routers hostname w/o domain such as r3. Now, for the other situation if channel 16 in the debug is really channel 17, that could be caused by using the ccm config command. With this, every time in cucm you reset the mgcp gw it will apply a no mgcp then mgcp and download the config from cucm to the router (and configure a FULL PRI). Ccm config command doesn't work with a fractional PRI, but you could use it to download all the commands, then no ccm config and change the controller commands to use timeslots 1-16 rather than 1-24. (Need to shut voice port, shut int s0/0/0 and no ISDN bind-l3 ccm, shut controller and remove timeslots command, then apply commands in reverse order using fractional timeslots). Not a solution, but just for reference, the default channel order for mgcp PRI is bottom up. If your issue is the latter (ccm config downloaded a full PRI config) and you were set to use ascending channels you would not have seen this issue until the 17th call came from pstn...in real lab you would lose points for having a full PRI I stead of fractional, even if calls did work. The point here is make sure you remove ccm config if you have a fractional PRI. Hope this helps... Justin On Mar 22, 2013 7:36 PM, Bill whl...@gmail.com wrote: Is your gateway registered in CUCM? Are you getting the proper output of your show commands? Show isdn status, show ccm Do you have int seri x/x/x Isdn bind-l3 ccm Did you try no MGCP MGCP? Can you post more of your config? Sent from my iPad On Mar 22, 2013, at 4:52 PM, CCIEing aboaz...@gmail.com wrote: Hi all, After configuring my HQ GW as MGCP, then configure my T1 to register with cucm , I was govern by the lack of the DSP resources, which force me to define only 16 channel out of the 23 on my pri-group under the T1 controller configuration !! here is the config : controller T1 0/0/0 pri-group timeslots 1-16 service mgcp Then I faced a problem with my outgoing calls , the calls was dropping due to the cause *Requested circuit/channel not available* My Question here, as there is 16 channel in my Pri-group are already configured, why all calls get dropped with cause of non availability of the resources, Why not to use one of the available channels (1-16) Appreciate your help here is below the output of debug q931 for one of my outgoing calls: *SDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref = 0x000F * *Bearer Capability i = 0x8090A2 * *Standard = CCITT * *Transfer Capability = Speech * *Transfer Mode = Circuit * *Transfer Rate = 64 kbit/s * *Channel ID i = 0xA98390 * *Exclusive, Channel 16 * *Display i = 'HQ PH1' * *Calling Party Number i = 0x2181, '7772022001' * *Plan:ISDN, Type:National * *Called Party Number i = 0x81, '911' * *Plan:ISDN, Type:Unknown* *ISDN Se0/0/0:23 Q931: RX - RELEASE_COMP pd = 8 callref = 0x800F * *Cause i = 0x82AC1810 - Requested circuit/channel not available* ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out
Re: [OSL | CCIE_Voice] Outgoing Calls via T1 failed
Well, I configured the pri-timeslots 1-16, because I have no dsp to configure all the 23 b-channels . On Sun, Mar 24, 2013 at 12:44 AM, Edgar Feliz ejzi...@gmail.com wrote: Are you sure the PSTN router has 1-16 configured for that GW. I had a similar issue because I had forgotten what I had set on PSTN router with only 4 slots and configure my GW with 8. Edgar On Fri, Mar 22, 2013 at 5:52 PM, CCIEing aboaz...@gmail.com wrote: Hi all, After configuring my HQ GW as MGCP, then configure my T1 to register with cucm , I was govern by the lack of the DSP resources, which force me to define only 16 channel out of the 23 on my pri-group under the T1 controller configuration !! here is the config : controller T1 0/0/0 pri-group timeslots 1-16 service mgcp Then I faced a problem with my outgoing calls , the calls was dropping due to the cause *Requested circuit/channel not available* My Question here, as there is 16 channel in my Pri-group are already configured, why all calls get dropped with cause of non availability of the resources, Why not to use one of the available channels (1-16) Appreciate your help here is below the output of debug q931 for one of my outgoing calls: *SDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref = 0x000F * *Bearer Capability i = 0x8090A2 * *Standard = CCITT * *Transfer Capability = Speech * *Transfer Mode = Circuit * *Transfer Rate = 64 kbit/s * *Channel ID i = 0xA98390 * *Exclusive, Channel 16 * *Display i = 'HQ PH1' * *Calling Party Number i = 0x2181, '7772022001' * *Plan:ISDN, Type:National * *Called Party Number i = 0x81, '911' * *Plan:ISDN, Type:Unknown* *ISDN Se0/0/0:23 Q931: RX - RELEASE_COMP pd = 8 callref = 0x800F * *Cause i = 0x82AC1810 - Requested circuit/channel not available* ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Network side vs User side clocking !
Hi geeks :) What is the difference between using Network side clocking and User Side clocking. Regarding the exam, do they ask us to use any one of the both in particular ? I saw practice question informing that the PRI circuit layer 2 should be user side where as it will be a network side clocking for layer 1 as for the last sentence (network site), I would assume that we will use *network-clock- participate wic X* * * *Waiting your valuable input * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] ISDN Interface won't to be bind with ccm-manager
Hi experts, I am trying to configure Branch2-R3 as MGCP VG, and I configured an interface E1. I have problem when try to bind-l3 in the serial interface s0/0/0:15 with ccm-manager , the only option appear is q931??? the gateway won't to register with ccm.. any idea ??? Appreciate your help! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] ISDN Interface won't to be bind with ccm-manager
Dears, The problem was with this command : controller e1 0/0/0 pri-group timeslots 1-12 where it should be controller e1 0/0/0 pri-group timeslots 1-12 service mgcp Thanks On Fri, Mar 22, 2013 at 9:57 PM, CCIEing aboaz...@gmail.com wrote: Hi experts, I am trying to configure Branch2-R3 as MGCP VG, and I configured an interface E1. I have problem when try to bind-l3 in the serial interface s0/0/0:15 with ccm-manager , the only option appear is q931??? the gateway won't to register with ccm.. any idea ??? Appreciate your help! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Outgoing Calls via T1 failed
Hi all, After configuring my HQ GW as MGCP, then configure my T1 to register with cucm , I was govern by the lack of the DSP resources, which force me to define only 16 channel out of the 23 on my pri-group under the T1 controller configuration !! here is the config : controller T1 0/0/0 pri-group timeslots 1-16 service mgcp Then I faced a problem with my outgoing calls , the calls was dropping due to the cause *Requested circuit/channel not available* My Question here, as there is 16 channel in my Pri-group are already configured, why all calls get dropped with cause of non availability of the resources, Why not to use one of the available channels (1-16) Appreciate your help here is below the output of debug q931 for one of my outgoing calls: *SDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref = 0x000F * *Bearer Capability i = 0x8090A2 * *Standard = CCITT * *Transfer Capability = Speech * *Transfer Mode = Circuit * *Transfer Rate = 64 kbit/s * *Channel ID i = 0xA98390 * *Exclusive, Channel 16 * *Display i = 'HQ PH1' * *Calling Party Number i = 0x2181, '7772022001' * *Plan:ISDN, Type:National * *Called Party Number i = 0x81, '911' * *Plan:ISDN, Type:Unknown* *ISDN Se0/0/0:23 Q931: RX - RELEASE_COMP pd = 8 callref = 0x800F * *Cause i = 0x82AC1810 - Requested circuit/channel not available* ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Calling Number not appear when call PSTN
Hello Experts,, Does any one of you encountered the following problem : I have MGCP gateway registered with CUCM 8.6, the GW have a FXO card installed and analog line connected to that FXO.. when I try to call any PSTN number the calling ID for my analog line that is connected to the FXO does not appear to the PSTN user (e.x: mobile user), there is not special service on the analog line to prevent the calling number.. Any Idea!!! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Globalizing on mgcp vg
Hi Jamie, Ya you are right regarding the globalize , But I may express in wrong words.. What I need is to add 9 for the calling numbers that appear on the Missed calls ..I am putting the 9 under the Incoming Calling Party Settings . But this digit is not appear on the missed calls.. Appreciate your help all On Fri, Mar 15, 2013 at 10:32 AM, Jamie Parr (jamparr) jamp...@cisco.comwrote: Adding 9 to the incoming number is not globalizing, you need to convert the number to the full E.164 number to globalize Jamie Parr Engineer - IT jamp...@cisco.com Phone: +44 20 8824 2641 Mobile: +44 7590622049 -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ahmad Gmail Sent: 14 March 2013 19:14 To: CCIE Study Subject: [OSL | CCIE_Voice] Globalizing on mgcp vg Dears, I have a 2 Gw's each one with E1 PRI, they are grouped by RL. I am trying to globalizing the incoming numbers (which is send by pstn an type: national ) by adding the digit (9) as prefix. But the 9 is not showing at all in the directory (missed/placed/received) Any idea !! Appreciated Sent from my iPhone ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Globalizing on mgcp vg
Man I did it, under GW -- Incoming Calling Party Settings.. * * *But it did not work :(* *!!! * On Fri, Mar 15, 2013 at 11:52 AM, Chrysostomos Christofi ch.christ...@logicom.net wrote: Why you don’t apply the 9 for calling numbers in the gw/? ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *CCIEing *Sent:* 15 March 2013 10:49 *To:* Jamie Parr (jamparr) *Cc:* CCIE Study *Subject:* Re: [OSL | CCIE_Voice] Globalizing on mgcp vg ** ** Hi Jamie, Ya you are right regarding the globalize , But I may express in wrong words.. ** ** What I need is to add 9 for the calling numbers that appear on the Missed calls ..I am putting the 9 under the I*ncoming Calling Party Settings* . But this digit is not appear on the missed calls.. ** ** Appreciate your help all ** ** On Fri, Mar 15, 2013 at 10:32 AM, Jamie Parr (jamparr) jamp...@cisco.com wrote: Adding 9 to the incoming number is not globalizing, you need to convert the number to the full E.164 number to globalize Jamie Parr Engineer - IT jamp...@cisco.com Phone: +44 20 8824 2641 Mobile: +44 7590622049 -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ahmad Gmail Sent: 14 March 2013 19:14 To: CCIE Study Subject: [OSL | CCIE_Voice] Globalizing on mgcp vg Dears, I have a 2 Gw's each one with E1 PRI, they are grouped by RL. I am trying to globalizing the incoming numbers (which is send by pstn an type: national ) by adding the digit (9) as prefix. But the 9 is not showing at all in the directory (missed/placed/received) Any idea !! Appreciated Sent from my iPhone ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Globalizing on mgcp vg
Ya actually I am not on site now.. I will try it and let you all know .. Thanks so much for your input .. On Fri, Mar 15, 2013 at 12:16 PM, Chrysostomos Christofi ch.christ...@logicom.net wrote: Hi ** ** ok I see ** ** **1) **Go to VG configuration. *Incoming calling Party National Number prefix* and add 9 there **2) **Issue no mgcp , mgcp .Then dial again and you should see in the directory 9 in front of national calls ** ** ** ** Regards ** ** ** ** *From:* CCIEing [mailto:aboaz...@gmail.com] *Sent:* 15 March 2013 11:04 *To:* Chrysostomos Christofi *Cc:* Jamie Parr (jamparr); CCIE Study *Subject:* Re: [OSL | CCIE_Voice] Globalizing on mgcp vg ** ** The GW is MGCP , and should work from the CUCM side.. On Fri, Mar 15, 2013 at 12:03 PM, Chrysostomos Christofi ch.christ...@logicom.net wrote: I mean into VG with translation rule and not in CUCM Is any reason that you need it in cucm? *From:* CCIEing [mailto:aboaz...@gmail.com] *Sent:* 15 March 2013 11:00 *To:* Chrysostomos Christofi *Cc:* Jamie Parr (jamparr); CCIE Study *Subject:* Re: [OSL | CCIE_Voice] Globalizing on mgcp vg Man I did it, under GW -- *Incoming Calling Party Settings..* *But it did not work :(* *!!!* On Fri, Mar 15, 2013 at 11:52 AM, Chrysostomos Christofi ch.christ...@logicom.net wrote: Why you don’t apply the 9 for calling numbers in the gw/? *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *CCIEing *Sent:* 15 March 2013 10:49 *To:* Jamie Parr (jamparr) *Cc:* CCIE Study *Subject:* Re: [OSL | CCIE_Voice] Globalizing on mgcp vg Hi Jamie, Ya you are right regarding the globalize , But I may express in wrong words.. What I need is to add 9 for the calling numbers that appear on the Missed calls ..I am putting the 9 under the I*ncoming Calling Party Settings* . But this digit is not appear on the missed calls.. Appreciate your help all On Fri, Mar 15, 2013 at 10:32 AM, Jamie Parr (jamparr) jamp...@cisco.com wrote: Adding 9 to the incoming number is not globalizing, you need to convert the number to the full E.164 number to globalize Jamie Parr Engineer - IT jamp...@cisco.com Phone: +44 20 8824 2641 Mobile: +44 7590622049 -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ahmad Gmail Sent: 14 March 2013 19:14 To: CCIE Study Subject: [OSL | CCIE_Voice] Globalizing on mgcp vg Dears, I have a 2 Gw's each one with E1 PRI, they are grouped by RL. I am trying to globalizing the incoming numbers (which is send by pstn an type: national ) by adding the digit (9) as prefix. But the 9 is not showing at all in the directory (missed/placed/received) Any idea !! Appreciated Sent from my iPhone ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Globalizing on mgcp vg
Hi Bill, The call is in type of national, I already applied the prefix , but it seems I did not issue the no mgcp/mgcp. Sent from my iPhone On Mar 15, 2013, at 1:11 PM, Bill whl...@gmail.com wrote: So the number comes in a MGCP GW and you want to change it from 555-1234 to 9555-1234 is that correct? If so, then use debug isdn q931 to verify the incoming numbers ANI and isdn type. If it is say local/isdn then under the GW page drill down near the bottom and prefix a 9 to the local incoming numbers. If not please post the isdn debug and the desired goal and remember this is a global change and will happen to all calls arriving with isdn and local in bound. Sent from my iPad On Mar 15, 2013, at 2:16 AM, Ahmad Gmail aboaz...@gmail.com wrote: Hi, All what i need is to show the calling number on the missed call list, but it is not working !!! Sent from my iPhone On Mar 15, 2013, at 1:20 AM, sethuvign...@yahoo.co.in wrote: Hi Ahmad, You are trying to globalise but to reflect on the phones you have to do localizations as well. Thanks, Vignesh Sent from Yahoo! Mail for iPhone From: Ahmad Gmail aboaz...@gmail.com; To: CCIE Study ccie_voice@onlinestudylist.com; Subject: [OSL | CCIE_Voice] Globalizing on mgcp vg Sent: Thu, Mar 14, 2013 7:14:19 PM Dears, I have a 2 Gw's each one with E1 PRI, they are grouped by RL. I am trying to globalizing the incoming numbers (which is send by pstn an type: national ) by adding the digit (9) as prefix. But the 9 is not showing at all in the directory (missed/placed/received) Any idea !! Appreciated Sent from my iPhone ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Enable Multicast routing on Routers
Hi Friends.. I have question on the MOH multicast, In case we have HQ and SiteB are connected to the same CUCM cluster, and we need to enable the the multicasting to be used with MOH. Which interfaces on both router should we enable the Multicast traffic , and based on which criteria ?? Cheers for all ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Enable Multicast routing on Routers
Great Input Bell, appreciated On Mon, Mar 11, 2013 at 11:34 PM, William Bell b...@ucguerrilla.com wrote: If you are asked to do multicast over the WAN then you need to: a. Consider CODEC. Likely, you will need to support G729 across the WAN and you will want to update the IPVMS, Regions/DP, etc. to facilitate that b. Enable ip multicast-routing on HQ and SiteB routers. c. Enable pim dense-mode on all layer 3 interfaces/hops between the CUCM, the Site B phones, and PSTN callers d. Provision mcast audio source, mcast on MOH server(s), mcast on MRGs. On MOH Servers, ensure you have the proper hop count. In the IE lab v3.0 topology, there should be 3 hops from CUCM servers and SiteB phones. e. Make sure you use ccm-manager music-on-hold on SiteB gateway to service the PSTN callers -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 11, 2013, at 3:52 PM, CCIEing wrote: Hi Friends.. I have question on the MOH multicast, In case we have HQ and SiteB are connected to the same CUCM cluster, and we need to enable the the multicasting to be used with MOH. Which interfaces on both router should we enable the Multicast traffic , and based on which criteria ?? Cheers for all ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Voice-card
Hi all, I always has leak of understanding regarding the dspfarm command..some times they use it with the no command under the voice-card x mode.. Can any one explain more about this command.. Why we use it and how to use it in the correct way.. appreciate your input in advance no dspfarm dsp service dspfarm ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] B Channel Busy Out
Hi Jason, I have question about step # 9 clear counters is it the normal clear counter command? e.x: E1 card clear counter interface s x/y/z:15 Thanks On Wed, Feb 27, 2013 at 3:02 PM, Jason Lee jas7...@gmail.com wrote: I use this as a strategy for checking my gateway configuration Ensure that your are meeting requirements on the following 1. display-ie 2. BCHAN order selection (Ascending, Descending) 3. BCHAN number 1. How many BCHANs? 1. If not specified create a full PRI. 2. If fractional 1. Set BCHAN Maintenance in Advanced Service Parameters 2. Check the *Check Status *checkbox in GW config 4. Clocking 1. Network clock participate 2. network clock select 1 t10/0/0 5. ISDN Switch-Type 6. Source-Address 7. 911 1. Done in gateway section. 1. Make sure to have routed correctly, SLRG?, 8. Direct Inward Dial 9. clear counters On Wed, Feb 27, 2013 at 3:38 AM, Jamie Parr (jamparr) jamp...@cisco.comwrote: I am also curious as to the grading on the gateways, I received very low marks on this section. Can anyone help? ** ** Thanks ** ** *Jamie Parr* Engineer - IT jamp...@cisco.com Phone: *+44 20 8824 2641* Mobile: *+44 7590622049* ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Pixar Perfect *Sent:* 26 February 2013 19:35 *To:* Steve Keller; GARY CLARK *Cc:* CCIE Voice OSL *Subject:* [OSL | CCIE_Voice] B Channel Busy Out ** ** Gary ..you mentioned B channel busyout on service parameter. in my understanding this was only needed when you would download the GW config from CCM i.e., ccm-manager config. it doesn't make any sense to use this service parameter as most of the solution guides (INE, IPXEPERT, 360) do not encourage the use of ccm-manager config except initial stage of your config and then disable it. I have heard ppl who passed just using standard configs but not sure if they did the B channel busy out on service parameter. ** ** ** ** mgcp mgcp call-agent 10.10.210.11 --sub mgcp dtmf mgcp bind ... (2x2) ** ** ccm-mana fall ccm-mana music ccm-mana mgcp ccm-mana red 10.10.210.10 -- pub ** ** ** ** if B channel status is *really graded *on the exam then it is one of those things that doesn't make sense to have it there but is needed to score points [image: Emoji] ** ** experts, any comments or advise from the recent Experts ? ** ** ** ** PIXAR ** ** ** ** -- Date: Mon, 25 Feb 2013 14:31:12 -0500 From: skeller...@gmail.com To: garyclark...@gmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] lab7 failed for 1% I recieved 29% in RTP on GW Signalling section and Call Routing as well. I am very discouraged i could score very low marks on these sections as i took my time and felt like i had nailed them. I scored really well in all other areas but failed because of these 2 sections. It is a mystery to me what the proctor is doing to arrive at that score, when all my calls worked, the debugs matched the requirements, i was binding to the correct interfaces, setting up the correct protocol and channels,etc. I would love to hear what insight folks have as to why the scores could be so low when everything looked to be working beautifully, without breaking NDA of course. thanks steve On Mon, Feb 25, 2013 at 1:54 PM, GARY CLARK garyclark...@gmail.com wrote: Hi friends, I got lab 7 in SJC and i completed that in 6 hrs and tested for 2 hrs time. I thought i have passed 1000% but when i saw my result i was surprised.** ** I almost got everywhere 100% except VG / 29% which was 17 marks section.* *** Same story with my friends do anyone got 100% in VG for lab 7 If anyone interested to share the hidden secrets then welcome as people are getting lab 7 repeating now very eager to understand what could be wrong. Please email me for further discussion. We 3 friends attempted out of which i also did busy out channel but that also did not helped its 29% only why so ** ** Regards ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are
[OSL | CCIE_Voice] H323 VG dial-peers
Hi All, For site B let us say that the lab ask us to configure it as H323 gateway, and the question mentioned that call have to go to SUB CUCM then to PUB, in that case we have to create the 2 voip dial peers pointing to sub and PUB as below : My question here , in order to make sure the preference of the 1st dial peer , do we have to hard coded the preference command in the dial-peers or the tag index of the dial peer will grantee that the sup dial peer will chosen 1st as it is tag is less than the pub dial-peer Thanks dial-peer voice 2 voip destination-pattern 12341$ session target ipv4:Sub_IP_Address codec g711ulaw dtmf-relay h245-alphanumaric h245-siganal cisco no vad dial-peer voice 2 voip destination-pattern 12341$ session target ipv4:Sub_IP_Address codec g711ulaw dtmf-relay h245-alphanumaric h245-siganal cisco no vad ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] H323 VG dial-peers
I am sorry the below dial-peers should be : dial-peer voice 2 voip destination-pattern 12341$ session target ipv4:Sub_IP_Address codec g711ulaw dtmf-relay h245-alphanumaric h245-siganal cisco no vad dial-peer voice 3 voip destination-pattern 12341$ session target ipv4:Pub_IP_Address codec g711ulaw dtmf-relay h245-alphanumaric h245-siganal cisco no vad On Sat, Mar 9, 2013 at 11:14 PM, CCIEing aboaz...@gmail.com wrote: Hi All, For site B let us say that the lab ask us to configure it as H323 gateway, and the question mentioned that call have to go to SUB CUCM then to PUB, in that case we have to create the 2 voip dial peers pointing to sub and PUB as below : My question here , in order to make sure the preference of the 1st dial peer , do we have to hard coded the preference command in the dial-peers or the tag index of the dial peer will grantee that the sup dial peer will chosen 1st as it is tag is less than the pub dial-peer Thanks dial-peer voice 2 voip destination-pattern 12341$ session target ipv4:Sub_IP_Address codec g711ulaw dtmf-relay h245-alphanumaric h245-siganal cisco no vad dial-peer voice 2 voip destination-pattern 12341$ session target ipv4:Sub_IP_Address codec g711ulaw dtmf-relay h245-alphanumaric h245-siganal cisco no vad ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] H323 VG dial-peers
Hi, Thanks all, Got it On Sun, Mar 10, 2013 at 12:01 AM, Amp amccar...@cciequest.com wrote: Hey, I would hard code the preferences. Being that the default preference in canonical terms is 0, you could set the preference on the pub dial peer to preference 1, or you could set the sub dial-peer to preference 1 and the pub to preference 2. Either way, I would hard code it. Amp Quoting CCIEing aboaz...@gmail.com: I am sorry the below dial-peers should be : dial-peer voice 2 voip destination-pattern 12341$ session target ipv4:Sub_IP_Address codec g711ulaw dtmf-relay h245-alphanumaric h245-siganal cisco no vad dial-peer voice 3 voip destination-pattern 12341$ session target ipv4:Pub_IP_Address codec g711ulaw dtmf-relay h245-alphanumaric h245-siganal cisco no vad On Sat, Mar 9, 2013 at 11:14 PM, CCIEing aboaz...@gmail.com wrote: Hi All, For site B let us say that the lab ask us to configure it as H323 gateway, and the question mentioned that call have to go to SUB CUCM then to PUB, in that case we have to create the 2 voip dial peers pointing to sub and PUB as below : My question here , in order to make sure the preference of the 1st dial peer , do we have to hard coded the preference command in the dial-peers or the tag index of the dial peer will grantee that the sup dial peer will chosen 1st as it is tag is less than the pub dial-peer Thanks dial-peer voice 2 voip destination-pattern 12341$ session target ipv4:Sub_IP_Address codec g711ulaw dtmf-relay h245-alphanumaric h245-siganal cisco no vad dial-peer voice 2 voip destination-pattern 12341$ session target ipv4:Sub_IP_Address codec g711ulaw dtmf-relay h245-alphanumaric h245-siganal cisco no vad ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] How to know ISDN Switch Type?
Hi all, When it comes to the configuration of the T1/E1, how we can know the isdn switch type that must be used ? Thanks in advance ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] How to know ISDN Switch Type?
Got it, Thanks guys.. On Fri, Mar 8, 2013 at 5:44 PM, Chrysostomos Christofi ch.christ...@logicom.net wrote: Hi ** ** It should say the question: Set this site with T1 controller , then the isdn switch type is* isdn switch-type primary-ni* If the question say this site has E1 controller then the isdn switch type is *isdn switch-type primary-net5* ** ** ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *CCIEing *Sent:* Παρασκευή, 8 Μαρτίου 2013 4:31 μμ *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] How to know ISDN Switch Type? ** ** Hi all, ** ** When it comes to the configuration of the T1/E1, how we can know the isdn switch type that must be used ? ** ** ** ** Thanks in advance ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Phone NTP Reference Vs NTP Reference
Hello All, The following question cross my mind while doing the NTP configuration stuff.. What is the difference between the Phone NTP reference configuration in the CCM Web administration page and The NTP reference on the OS Administration page?? does the 1st one for the endpoints where the 2nd one is for the CUCM itself? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Phone NTP Reference Vs NTP Reference
Oh thanks a lot for your input. Appreciated .. On Tue, Mar 5, 2013 at 12:48 AM, Cory Gray corygray22...@hotmail.comwrote: Phone ntp reference is for SIP phones only Sent from my iPhone On Mar 4, 2013, at 4:42 PM, CCIEing aboaz...@gmail.com wrote: Hello All, The following question cross my mind while doing the NTP configuration stuff.. What is the difference between the Phone NTP reference configuration in the CCM Web administration page and The NTP reference on the OS Administration page?? does the 1st one for the endpoints where the 2nd one is for the CUCM itself? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CCIE Voice Rack Rental
Hi All, I need to know if anyone has information about the IPexpert rack rental, I bought a sessions from IpExpert site, but now i am not able to find where to schedule my sessions? appreciate your help guys. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] IP Phone Agent
I used the same URL, with the same error appear ! On Sun, Jan 27, 2013 at 3:06 AM, Bill Lake whl...@gmail.com wrote: go here, get correct IP agent http at the number 5 down the page. It is bold even so easy to find http://www.cisco.com/en/US/partner/products/sw/custcosw/ps1846/products_tech_note09186a008029e6d5.shtml Here is the link you want to use to setup IPPA *http://xxx.xxx.xxx.xx:/ipphone/jsp/sciphonexml/IPAgentLogin.jsp * On Sat, Jan 26, 2013 at 3:28 PM, CCIEing aboaz...@gmail.com wrote: Hi All, I am suffering the *Cannot connect to the IP Phone Agent service* error, when try to browse the url *http://uccx_IP_address:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp*; I am not able to continue the configuration of the IP Phone agent. I have changed the Ip address of the UCCX, does this affect ? The full output of when try to the url above is : ?xml version=1.0? -CiscoIPPhoneText TitleError/Title TextCannot connect to the IP Phone Agent service./Text Prompt/- SoftKeyItem NameOK/Name URLKey:Services/URL Position1/Position /SoftKeyItem /CiscoIPPhoneText any idea?? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] IP Phone Agent
Yes Man, I restarted it, the same error still occurs !!! On Sat, Jan 26, 2013 at 11:59 PM, Gurpreet Singh Kukreja tycoononway1...@gmail.com wrote: Hey Man, Tried restarting the BIPPA service from CCX yet? - G On Sat, Jan 26, 2013 at 4:28 PM, CCIEing aboaz...@gmail.com wrote: Hi All, I am suffering the *Cannot connect to the IP Phone Agent service* error, when try to browse the url *http://uccx_IP_address:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp*; I am not able to continue the configuration of the IP Phone agent. I have changed the Ip address of the UCCX, does this affect ? The full output of when try to the url above is : ?xml version=1.0? -CiscoIPPhoneText TitleError/Title TextCannot connect to the IP Phone Agent service./Text Prompt/- SoftKeyItem NameOK/Name URLKey:Services/URL Position1/Position /SoftKeyItem /CiscoIPPhoneText any idea?? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] IP Phone Agent
I want to add the following information: These 2 services are not able to start: Cisco Desktop LDAP Monitor Service Cisco Desktop Sync Service On Sun, Jan 27, 2013 at 10:57 PM, CCIEing aboaz...@gmail.com wrote: Yes Man, I restarted it, the same error still occurs !!! On Sat, Jan 26, 2013 at 11:59 PM, Gurpreet Singh Kukreja tycoononway1...@gmail.com wrote: Hey Man, Tried restarting the BIPPA service from CCX yet? - G On Sat, Jan 26, 2013 at 4:28 PM, CCIEing aboaz...@gmail.com wrote: Hi All, I am suffering the *Cannot connect to the IP Phone Agent service* error, when try to browse the url *http://uccx_IP_address:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp * I am not able to continue the configuration of the IP Phone agent. I have changed the Ip address of the UCCX, does this affect ? The full output of when try to the url above is : ?xml version=1.0? -CiscoIPPhoneText TitleError/Title TextCannot connect to the IP Phone Agent service./Text Prompt/- SoftKeyItem NameOK/Name URLKey:Services/URL Position1/Position /SoftKeyItem /CiscoIPPhoneText any idea?? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] IP Phone Agent
Not forget to mention that I have changed the IP address of the server using ! Is that related to my problem.. Appreciate your help, as this issue take a lot of my time :( On Sun, Jan 27, 2013 at 11:12 PM, CCIEing aboaz...@gmail.com wrote: I want to add the following information: These 2 services are not able to start: Cisco Desktop LDAP Monitor Service Cisco Desktop Sync Service On Sun, Jan 27, 2013 at 10:57 PM, CCIEing aboaz...@gmail.com wrote: Yes Man, I restarted it, the same error still occurs !!! On Sat, Jan 26, 2013 at 11:59 PM, Gurpreet Singh Kukreja tycoononway1...@gmail.com wrote: Hey Man, Tried restarting the BIPPA service from CCX yet? - G On Sat, Jan 26, 2013 at 4:28 PM, CCIEing aboaz...@gmail.com wrote: Hi All, I am suffering the *Cannot connect to the IP Phone Agent service* error, when try to browse the url * http://uccx_IP_address:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp*; I am not able to continue the configuration of the IP Phone agent. I have changed the Ip address of the UCCX, does this affect ? The full output of when try to the url above is : ?xml version=1.0? -CiscoIPPhoneText TitleError/Title TextCannot connect to the IP Phone Agent service./Text Prompt/- SoftKeyItem NameOK/Name URLKey:Services/URL Position1/Position /SoftKeyItem /CiscoIPPhoneText any idea?? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] IP Phone Agent
Finally it worked :) woow.. It seems changing the IP address of the server made all these problems, I do some regedit update to reflect the new IP address change, then restart the server, and things went in the correct direction . Thanks all for your input. On Sun, Jan 27, 2013 at 11:53 PM, CCIEing aboaz...@gmail.com wrote: Not forget to mention that I have changed the IP address of the server using ! Is that related to my problem.. Appreciate your help, as this issue take a lot of my time :( On Sun, Jan 27, 2013 at 11:12 PM, CCIEing aboaz...@gmail.com wrote: I want to add the following information: These 2 services are not able to start: Cisco Desktop LDAP Monitor Service Cisco Desktop Sync Service On Sun, Jan 27, 2013 at 10:57 PM, CCIEing aboaz...@gmail.com wrote: Yes Man, I restarted it, the same error still occurs !!! On Sat, Jan 26, 2013 at 11:59 PM, Gurpreet Singh Kukreja tycoononway1...@gmail.com wrote: Hey Man, Tried restarting the BIPPA service from CCX yet? - G On Sat, Jan 26, 2013 at 4:28 PM, CCIEing aboaz...@gmail.com wrote: Hi All, I am suffering the *Cannot connect to the IP Phone Agent service* error, when try to browse the url * http://uccx_IP_address:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp * I am not able to continue the configuration of the IP Phone agent. I have changed the Ip address of the UCCX, does this affect ? The full output of when try to the url above is : ?xml version=1.0? -CiscoIPPhoneText TitleError/Title TextCannot connect to the IP Phone Agent service./Text Prompt/- SoftKeyItem NameOK/Name URLKey:Services/URL Position1/Position /SoftKeyItem /CiscoIPPhoneText any idea?? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] IP Phone Agent
Hi All, I am suffering the *Cannot connect to the IP Phone Agent service* error, when try to browse the url *http://uccx_IP_address:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp*; I am not able to continue the configuration of the IP Phone agent. I have changed the Ip address of the UCCX, does this affect ? The full output of when try to the url above is : ?xml version=1.0? -CiscoIPPhoneText TitleError/Title TextCannot connect to the IP Phone Agent service./Text Prompt/- SoftKeyItem NameOK/Name URLKey:Services/URL Position1/Position /SoftKeyItem /CiscoIPPhoneText any idea?? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Add CUC as Application Server
Hi all, How much (important and mandatory) is adding the CUC or CUPS to the CUCM as Application Server? thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] cisco DocCD
Hi All, I was practicing the cisco Documentation CD that will be available during the lab http://www.cisco.com/cisco/web/psa/default.html to get used on it, I was searching for the topic in unity connection How to set up the Phone View ,this topic is mentioned in the System Administration for Cisco Unity Connection Release 7.X ( http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/administration/guide/7xcucsag245.html ) I was not able to find the path in the Doc CD, I used Google to fine this guide from cisco site. Do you guys practice the Doc CD - If not, then you have to, as this is the only documentation resource during the exam..-, do you know the Path to the system administration guide for the products. Awaiting your response, Appreciate that. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] cisco DocCD
Here we go, I found what I need under this path : Product Voice and UC IP Telephony Unified Messaging CUC Maintain and Operate (left panel ) Maintain and Operate Guides System Administration Guide for Cisco Unity Connection Release 7.x and finally *Setting Up Phone View* * * Thanks everyone :) * * On Mon, Jan 14, 2013 at 6:42 PM, Suresh Bhandari bring...@gmail.com wrote: And the SysAdmin guide can be found following ProductsVoice and UCIP TelephonyUnified MessagingCUCConfigure (left pane links)Configuration Guides CAN'T FIND WHAT YOU WANT? Check the Documentation Guide DG for CUC 7.x And you are there with most, if not all, of the links for the version selected. Cheers! On Mon, Jan 14, 2013 at 10:08 PM, Cory Gray corygray22...@hotmail.comwrote: The doc cd was retired from the lab many years ago. It is now the support pages with the three windows for you to navigate through. Sent from my iPhone On Jan 14, 2013, at 11:18 AM, CCIEing aboaz...@gmail.com wrote: Hi All, I was practicing the cisco Documentation CD that will be available during the lab http://www.cisco.com/cisco/web/psa/default.html to get used on it, I was searching for the topic in unity connection How to set up the Phone View ,this topic is mentioned in the System Administration for Cisco Unity Connection Release 7.X ( http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/administration/guide/7xcucsag245.html ) I was not able to find the path in the Doc CD, I used Google to fine this guide from cisco site. Do you guys practice the Doc CD - If not, then you have to, as this is the only documentation resource during the exam..-, do you know the Path to the system administration guide for the products. Awaiting your response, Appreciate that. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Configuriong Conferance that used with LiveRecord in CUE
Hello Guys, The following question passed my mind while I am practicing the LiveRecord feature under the CUE topic. What type of conference should we used for the purpose of configuring the Liverecord under CUE task, is it the - Ad-hoc conference , just like below configuration example : Ephone-dn 1 dual-line Number A101 Conference Ad-hoc ! Ephone-dn 2 dual-line Number A101 Conference Ad-hoc ! Ephone-dn 3 dual-line Number A101 Conference Ad-hoc ! Ephone-dn 4 dual-line Number A101 Conference Ad-hoc OR Use the - Hardware conference bridge and based on which conditions shall we chose between the above two options . Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Tab on the LAB exam
Hello Friends, I have a small question about the exam, does the Tab (to complete commands ) in the routers CLI is enabled ? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Tab on the LAB exam
Oh thanks a lot guys .. On Thu, Jan 10, 2013 at 3:31 AM, Gurpreet Singh Kukreja tycoononway1...@gmail.com wrote: Yup On Wed, Jan 9, 2013 at 6:26 PM, CCIEing aboaz...@gmail.com wrote: Hello Friends, I have a small question about the exam, does the Tab (to complete commands ) in the routers CLI is enabled ? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Tab on the LAB exam
So, What type of MWI is better to use ?? outcalling , unsolicited , or sub-notify ?? On Thu, Jan 10, 2013 at 3:58 AM, CCIEing aboaz...@gmail.com wrote: Oh thanks a lot guys .. On Thu, Jan 10, 2013 at 3:31 AM, Gurpreet Singh Kukreja tycoononway1...@gmail.com wrote: Yup On Wed, Jan 9, 2013 at 6:26 PM, CCIEing aboaz...@gmail.com wrote: Hello Friends, I have a small question about the exam, does the Tab (to complete commands ) in the routers CLI is enabled ? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] MWI off for broadcast message is not working.
Hi All, I configured a user in the CUE to send broadcast message, he is able to send the message, the MWI ON appear on the phone, but when recipient hear the broadcast message and the message deleted, the red alarm and the envelop does not disappear , the MWI OFF is not working, until I perform the mwi refresh comman from CUE CLI. Any help please. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] MWI off for broadcast message is not working.
Guys Seems that I am asking, then I can find an answer to my questions, :) I find the following cisco document : * http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/design/design21/cuevmdes.html#wp1008620 * Which says clearly that WI operation for broadcast messaging cannot be controlled per individual mailbox, it is an attribute of the system on which the message is delivered. If MWI for broadcast messaging is configured to be on, the MWI light is lit for all mailboxes on the system when a broadcast message becomes active. MWI is *refreshed when a broadcast message expires* so that any mailboxes that have not listened to the message will never receive it and the MWI state is turned off (unless there are other new messages in the mailbox). On Wed, Jan 9, 2013 at 1:02 AM, CCIEing aboaz...@gmail.com wrote: Hi All, I configured a user in the CUE to send broadcast message, he is able to send the message, the MWI ON appear on the phone, but when recipient hear the broadcast message and the message deleted, the red alarm and the envelop does not disappear , the MWI OFF is not working, until I perform the mwi refresh comman from CUE CLI. Any help please. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUE CLI configuration
Hello Again guys, Return to CUE configuration using CLI . So practicing CLI Imposes on us to know all the parameters for all applications, but there are a lot of parameters like the autoattendant app, right? is there any way to remember them from inside the cli cue help, does any one have a difficulty to learn them . Appreciate your cooperation On Fri, Jan 4, 2013 at 2:12 PM, Chrysostomos Christofi ch.christ...@logicom.net wrote: Sorry ** ** Apologize wrong type ** ** Correction: I heard from a lot guys that they had troubles to access the cue through gui J ** ** ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Chrysostomos Christofi *Sent:* Παρασκευή, 4 Ιανουαρίου 2013 1:09 μμ *To:* Ahmad Taamneh; William Bell *Cc:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] CUE CLI configuration ** ** I heard from a lot guys that they had troubles to access the cue through cli So we have to learn both ways ** ** Also its more fast with cli either with cme or cucm integration ** ** Regards Chrysostomos ** ** ** ** ** ** *From:* ccie_voice-boun...@onlinestudylist.com [ mailto:ccie_voice-boun...@onlinestudylist.comccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ahmad Taamneh *Sent:* Παρασκευή, 4 Ιανουαρίου 2013 12:54 μμ *To:* William Bell *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] CUE CLI configuration ** ** Thanks Bell, Appreciate your valuable opinion , ya you are right regarding CUE-CME, i will be much faster to use CLI, Sent from my iPhone On Jan 4, 2013, at 3:51 AM, William Bell b...@ucguerrilla.com wrote:** ** I suppose they could but I'd actually expect they wouldn't force you to use one or the other. The advice I see others give in this forum is to make sure you know how to do both. Usually that advice is followed with a ...besides, the CLI is much faster. Using the CLI is faster, at least for the CUE-CME integration. I haven't attempted it for CUE-CUCM. ** ** But that isn't your question. The generic answer is they could require you to use the CLI. My opinion is that would be quite lame if they did. There are plenty of ways to screw with you and restricting you from using the GUI on CUE seems petty. Just my opinion. That said, I am practicing both and will use the one that is fastest and the one I am most comfortable with under a time pressure. ** ** ** ** -Bill ** ** William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla ** ** ** ** ** ** On Jan 3, 2013, at 6:26 PM, CCIEing wrote: ** ** Hi All, ** ** I have a question regarding the required method to configure CUE, during the lab exam, can the ask us to configure the CUE using only command Line? and not using the GUI at all. ** ** Thanks ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CUE CLI configuration
Hi All, I have a question regarding the required method to configure CUE, during the lab exam, can the ask us to configure the CUE using only command Line? and not using the GUI at all. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUE with CME integration
Hello All, At first Happy new your to all of you, wish you a pass attempt . Regarding this topic I opened a PDI case, they answered me with the following : *it is mandatory to house CME CUE in the same router.* * * *You might want to first try installing the CME module along with CME in same ISR let me know in case of any further issues*. Thanks On Mon, Dec 31, 2012 at 12:05 AM, CCIEing aboaz...@gmail.com wrote: Dear All, I am integrating CUE with CME, When I run the initialization Wizard , the system keep giving me the following message Login to Call Manager Express as Administrator Failed. Check your Call Manager Express Configuration I am using a Call Manager Express not on box with the CUE module, it is on other router, the Telephony-Service Configuration that is related to the username and password is here below : *telephony-service* * max-ephones 10* * max-dn 10* * ip source-address 192.168.35.1 port 2000* * system message CME as SRST* * time-zone 31* * keepalive 10* * max-conferences 8 gain -6* * web admin system name admin password cisco * * * My Questions Here: 1- Is it mandatory for CUE to be in the same router with CME? 2- Why this this message continue appear, I Put the correct information about the Web user name and Passwor. Appreciate your help. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CUE with CME integration
Dear All, I am integrating CUE with CME, When I run the initialization Wizard , the system keep giving me the following message Login to Call Manager Express as Administrator Failed. Check your Call Manager Express Configuration I am using a Call Manager Express not on box with the CUE module, it is on other router, the Telephony-Service Configuration that is related to the username and password is here below : *telephony-service* * max-ephones 10* * max-dn 10* * ip source-address 192.168.35.1 port 2000* * system message CME as SRST* * time-zone 31* * keepalive 10* * max-conferences 8 gain -6* * web admin system name admin password cisco * * * My Questions Here: 1- Is it mandatory for CUE to be in the same router with CME? 2- Why this this message continue appear, I Put the correct information about the Web user name and Passwor. Appreciate your help. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUE with CUCM
Thanks All, Do you have a good document -you already used-about this integration On Sat, Dec 29, 2012 at 2:35 AM, Bill Lake whl...@gmail.com wrote: You should be prepared to integrate cue with either CME or CUCM and you should be prepared to do any normal task and some extra ordinary tasks with it. On Fri, Dec 28, 2012 at 4:15 PM, CCIEing aboaz...@gmail.com wrote: Hi All, I want to make sure that the CUE integration with CUCM is included of the LAB exam , right ? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CUE integration With CUCM
Dear All, I did the configuration of the CUE - CUCM integration, I was able to import the users from cucm, also the CTI Ports are registered with the CUCM, more over I went through the initialization wizard for the CUE, the voice mail pilot, profile already configured and assigned to phones , the CUCM can ping botht the VG and CUE module and visa v. but when I try to call the voice mail pilot, or push the messages button I get this message on the phone Can not reach the number. Any idea. Thanks in advance for your time . Ahmad ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUE integration With CUCM
Dear All, It seems it is a compatibility issue, I am using CUCM 8, and based on the CUE 7.X compatibility matrix * http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/compatibility/cuecomp.htm#CUCM * it is not compatible :(. Thanks All again for your time On Sat, Dec 29, 2012 at 8:03 PM, CCIEing aboaz...@gmail.com wrote: Dear All, I did the configuration of the CUE - CUCM integration, I was able to import the users from cucm, also the CTI Ports are registered with the CUCM, more over I went through the initialization wizard for the CUE, the voice mail pilot, profile already configured and assigned to phones , the CUCM can ping botht the VG and CUE module and visa v. but when I try to call the voice mail pilot, or push the messages button I get this message on the phone Can not reach the number. Any idea. Thanks in advance for your time . Ahmad ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Terminal used for lab is putty or CRT or any other terminal
Dears, I hear both could be user Putty, SecureCRT. On Sat, Dec 29, 2012 at 8:24 PM, singh singh8...@in.com wrote: hi Guys, I am interesting in knowning the following ... 1) From the CCIE voice lab which is the terminal connnection used ( putty or crt)? 2)Is it ssh or telnet? -singh Get Yourself a cool, short *@in.com* Email ID now!http://www3.in.com/sso/commonregister.php?ref=INutm_source=inviteutm_medium=outgoing ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Cisco Unity Connection - VM pilot and VM profile made default but not applied automatically on phones
Hi virajith, You may update all your phone settings using bulk edit. On Sat, Dec 29, 2012 at 8:36 PM, virajith vir...@rediffmail.com wrote: hi All, I am noticing that for my cisco unity connection integrated with CUCM - the VM pilot and profile that I have made as default VM pilot and profile is not getting automatically applied on my phones and therefore any call forward busy or no answer gets a busy tone. I have to manually update the VM profile on the DN level of the phone to indicate that the default is the unity connection. How do I correct this behaviour so that the phones automatically can get the default VM settings without me having to specify it? -Vir http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? Catch India as it happens with the *Rediff News App*. To download it for FREE, click herehttp://track.rediff.com/click?url=___http://www.rediff.com/newsapp___lnk=signaturenewservice=newsapp ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Unity Connection
Hi Abdullin, I tried to delete the user then create it again in both methods Manually and also imported, then changed the Display name to be 2001, and it worked with me this time. I think at -but not sure why it did not work-1st when I tested the feature the user has some greeting setting, that why it did not work. Any way thanks guys a lot for your response . Ahamd On Wed, Dec 26, 2012 at 9:25 AM, Abdullin Kamil kabdulli...@gmail.comwrote: Hi, For this purpose it is necessary to bring manually the user to CUC, instead of to import users. 2012/12/26, CCIEing aboaz...@gmail.com: Hi all, is there any possibility to achieve the following : *When someone calls from PSTN to phone number 2001 (Br2 Phone1) it should not say HQ PHONE 1 not available it should say* *2001 is not available please leave your message after the tone*. modifying the personal greetings is not allowed I tried to change the display name for this user to 2001, but it did not work with me. Thanks in Advance. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Unity Connection Call handler
Hi All, a small question about the system call handlers, when you create a new system call handler, do we have to stuck with time zone for HQ site? or just leave it to Use System Default Time Zone. Thanks in advance ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Unity Connection
Hi all, is there any possibility to achieve the following : *When someone calls from PSTN to phone number 2001 (Br2 Phone1) it should not say HQ PHONE 1 not available it should say* *2001 is not available please leave your message after the tone*. modifying the personal greetings is not allowed I tried to change the display name for this user to 2001, but it did not work with me. Thanks in Advance. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
