[OSL | CCIE_Voice] Accessing files in downloads are of IPX web site - support is stumped.
anyone having issues downloading the PDFs from the IPX ebook / downloads area? worked with support and didn't get anywhere, wasn't sure if anyone else was having an issue. using both chrome and IE, same issue. using Adobe Acrobat Pro and Adobe Reader 9.X, same issue. when I click on a file to view (i.e. a workbook), it dumps me right back ot the ebooks / downloads page but never tries to open the file. when I right mouse click on a link, it's an .htm file and not a .pdf. support is stumped, just seeing if anyone else has ran into this issue. thanks for whatever help you can provide. ___ 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] VMware ESXi for CCM servers
Make sure you're installing on an intel CPU. Had major issues installing to an AMD. I needed to install on an Intel pc (running server) the. 'export' to the AMD box. Once migrated it runs fine. On Saturday, March 21, 2009, Arun Kumar arunv...@gmail.com wrote: Hi I'm running CUCM 7 and Unity Connection 7 on ESXi with 6GB of RAM and 500GB of HDD and it's working fine. Not tested on Linux. Thanks On Sat, Mar 21, 2009 at 6:44 PM, WorkerBee cisco...@gmail.com wrote: Anyone has tried using ESXi with Quad core/8G ram instead of using VMware server on a Linux? Does ESXi gives a better performance? 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] First Attempt...Failed miserably
Adam, word of experience. Don't give yourself too long. A couple weeks, any longer than that and you will lose focus. That happened to me and it's like starting over as far as the study habits go. On Saturday, March 12, 2011, Roger Källberg roger.kallb...@cygate.se wrote: Hi Adam, Don't be to bummed out by this, only a very small percentage pass on the first attempt. Use this experience as part of your learning curve. Take a short brake and recuperate to get your motivation back, then start to analyze what you need to do better next time around. Focus on your identifiedO weak areas and make a strong comeback on your next attempt. Sincerely Roger Källberg CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Från: adam compton [com...@gmail.com] Skickat: den 10 mars 2011 14:58 Till: ccie_voice@onlinestudylist.com Ämne: [OSL | CCIE_Voice] First Attempt...Failed miserably Just giving everybody a status report. I failed the Voice lab yesterday. I'm really bummed out. It's not that I failed that bums me out. It's that a lot of areas I though I nailed, I got 0 percent. It's going to be hard to get back on the horse and do it again, but I will probably try again in 30 days. Adam Compton ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Unity in a Windows 2008 R2 domain
A meeting with a customer led to a discussion about their AD structure. They have Unity 4.0.5 still in place and are going to move to Unity 7 or UC in the near future. The question on the AD structure brought out that they were in the midst of upgrading now. Even Unity 7 is not supported on a 2008 R2 domain (only 2008 first release), though I can't get Cisco to tell me what specifically isn't supported. They've already upgraded their AD and the global catalog and most of the other servers in the cluster are already at 2008 R2. does anyone have any experience with what is 'broken' with 2008 R2, or what issues I can expect moving forward? any help is appreciated Mike ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] New VOD shippping today?
from what I understand...for the low, low cost of $50 (plus shipping), you will get this DVD shipped to you. or, you can wait until about april 16th when the HardDrive BLS upgrade will be available. I assume you send in your HD and they reimage it. On Wed, Mar 10, 2010 at 10:26 AM, Steve Sarrick ssarr...@drsllc.net wrote: Just curious if there are any rumors to the new VOD shipping today based on the website date of no later than March 10th. Has anyone seen/heard anything? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] I passed the CCIE Voice Lab
dude, I had all kinds of things that i was queueing up to say to you (and none of them were nice with how flippant you were being about passing the exam), but that was f$#ing funny!!! On Mon, Mar 1, 2010 at 9:46 PM, Jason Granat j...@slash128.com wrote: It was actually pretty easy to pass the lab on the first try. If you have been to the San Jose campus you know what I mean. If you are traveling West on Tasman you have to pass the lab and flip a u-turn at Champion to get to the Bldg C driveway where the lab is :-) Unfortunately, I failed my first attempt at the exam... http://slash128.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE - new software release
I would think that the end result would be anything in the 7.X.X train is testable. I don't forsee them jumping major revisions like that as the additional content available is to be pretty substantial. On Tue, Feb 9, 2010 at 4:54 PM, Saeed IDris saee...@gmail.com wrote: Hi : First of all I would like to Thanks community support from IP Expert , can someone confirm the below line: - CCIE Voice - Cisco .com: “ *CCIE* *Software Versions:* *Any major software release which has been generally* *available for six months is eligible for testing in the* *CCIE Voice Lab Exam*.” What I understand from the above lines that Cisco know release CUCM 8.0 this month which mean it will be testable topic after 6 month. “JULY 2010”, because im plan my lab on July. Regards, SID ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Thread Hijack: VoD's released this Friday?
yes, it would suck...but who's going to pay for their OC48 Internet Connection to addres all of us trying to download the updates at once? On Mon, Jul 27, 2009 at 4:46 PM, Tanner Ezell tanner.ez...@gmail.comwrote: Hopefully we'll be able to download them, having to send back the hard drive I just got would kinda suck :) On Mon, Jul 27, 2009 at 1:42 PM, Jason Granat j...@slash128.com wrote: Hi Mark, Sorry to hijack the thread but wanted to know if the new VoD's you mention will be available for download to current customers this Friday or will they be shipped only? Thanks, Jason -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Snow Sent: Monday, July 27, 2009 1:22 PM To: Aamir Panjwani Cc: ccie_voice@onlinestudylist.com; Michael Ciarfello Subject: Re: [OSL | CCIE_Voice] Any primer on the international escape character? Aamir, As you mentioned, I am hard at work on the new VoDs. ('cept when I take a few moments to answer Q's via this medium :P ). They will be released by Friday - end of this week, and shipping out the door first thing Monday (week from today). I do cover Call Routing in exhaustive detail (in fact it is a 4.5 hour section alone) - and I do go great detail into +dialing, Globalization, Localization, Calling Party XFormation CSS, etc. Also as Kevin mentioned - if you went or know someone who went to Networkers from your company - those Breakout Session PDF's are a great resource! Back to recording. Cheers, -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: ms...@ipexpert.com -- Join our free online support and peer group communities: http://www.IPexpert.com/communities http://www.ipexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On- Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Jul 24, 2009, at 10:41 PM, Aamir Panjwani wrote: I would also like to know how people are preparing for topics like +dialing, calling party normalization (localizing the calling number, globalizing the calling party number, mapping the global party number to its local variant). I have read above mentioned topics in SRND few times and it is extremely confusing and poorly explained to say the least! Mark Snow wrote calling party normalization part 1 back in March on ipexpert blog which was explained in simple english, I know he is extremely busy making VOD for us but I hope he can write part 2 soon for us :) By the way does anyone know whether these topics are covered in the 2 new volume 2 labs released recently? Thanks -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Michael Ciarfello Sent: Saturday, 25 July 2009 12:09 PM To: Jonathan Charles; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Any primer on the international escape character? Can you re-phrase that? Like SRND 7.x, page 10-65 escape character? From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jonathan Charles [jonv...@gmail.com] Sent: Friday, July 24, 2009 4:01 PM To: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net Subject: [OSL | CCIE_Voice] Any primer on the international escape character? Anyone have a guide to how to configure it? To make sure sure telco sends it, and it is received, etc.? Jonathan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit
Re: [OSL | CCIE_Voice] Thread Hijack: VoD's released this Friday?
I am pretty sure that Wayne isn't going to put this material out on a BitTorrent for the world to scavange. On Mon, Jul 27, 2009 at 7:29 PM, Tanner Ezell tanner.ez...@gmail.comwrote: Indeed, but on the other hand.. that is what things like BitTorrent are best for :) On Mon, Jul 27, 2009 at 4:27 PM, Michael Thompson mthompson...@gmail.comwrote: yes, it would suck...but who's going to pay for their OC48 Internet Connection to addres all of us trying to download the updates at once? On Mon, Jul 27, 2009 at 4:46 PM, Tanner Ezell tanner.ez...@gmail.comwrote: Hopefully we'll be able to download them, having to send back the hard drive I just got would kinda suck :) On Mon, Jul 27, 2009 at 1:42 PM, Jason Granat j...@slash128.com wrote: Hi Mark, Sorry to hijack the thread but wanted to know if the new VoD's you mention will be available for download to current customers this Friday or will they be shipped only? Thanks, Jason -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Snow Sent: Monday, July 27, 2009 1:22 PM To: Aamir Panjwani Cc: ccie_voice@onlinestudylist.com; Michael Ciarfello Subject: Re: [OSL | CCIE_Voice] Any primer on the international escape character? Aamir, As you mentioned, I am hard at work on the new VoDs. ('cept when I take a few moments to answer Q's via this medium :P ). They will be released by Friday - end of this week, and shipping out the door first thing Monday (week from today). I do cover Call Routing in exhaustive detail (in fact it is a 4.5 hour section alone) - and I do go great detail into +dialing, Globalization, Localization, Calling Party XFormation CSS, etc. Also as Kevin mentioned - if you went or know someone who went to Networkers from your company - those Breakout Session PDF's are a great resource! Back to recording. Cheers, -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: ms...@ipexpert.com -- Join our free online support and peer group communities: http://www.IPexpert.com/communitieshttp://www.ipexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On- Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Jul 24, 2009, at 10:41 PM, Aamir Panjwani wrote: I would also like to know how people are preparing for topics like +dialing, calling party normalization (localizing the calling number, globalizing the calling party number, mapping the global party number to its local variant). I have read above mentioned topics in SRND few times and it is extremely confusing and poorly explained to say the least! Mark Snow wrote calling party normalization part 1 back in March on ipexpert blog which was explained in simple english, I know he is extremely busy making VOD for us but I hope he can write part 2 soon for us :) By the way does anyone know whether these topics are covered in the 2 new volume 2 labs released recently? Thanks -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Michael Ciarfello Sent: Saturday, 25 July 2009 12:09 PM To: Jonathan Charles; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Any primer on the international escape character? Can you re-phrase that? Like SRND 7.x, page 10-65 escape character? From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jonathan Charles [jonv...@gmail.com] Sent: Friday, July 24, 2009 4:01 PM To: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net Subject: [OSL | CCIE_Voice] Any primer on the international escape character? Anyone have a guide to how to configure it? To make sure sure telco sends it, and it is received, etc.? Jonathan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com
Re: [OSL | CCIE_Voice] Thread Hijack: VoD's released this Friday?
good thing for me there's no bittorrent setup on the lab, eh?? :) On Mon, Jul 27, 2009 at 7:56 PM, Tanner Ezell tanner.ez...@gmail.comwrote: You can control who is able to download, and what. On Mon, Jul 27, 2009 at 4:43 PM, Michael Thompson mthompson...@gmail.comwrote: I am pretty sure that Wayne isn't going to put this material out on a BitTorrent for the world to scavange. On Mon, Jul 27, 2009 at 7:29 PM, Tanner Ezell tanner.ez...@gmail.comwrote: Indeed, but on the other hand.. that is what things like BitTorrent are best for :) On Mon, Jul 27, 2009 at 4:27 PM, Michael Thompson mthompson...@gmail.com wrote: yes, it would suck...but who's going to pay for their OC48 Internet Connection to addres all of us trying to download the updates at once? On Mon, Jul 27, 2009 at 4:46 PM, Tanner Ezell tanner.ez...@gmail.comwrote: Hopefully we'll be able to download them, having to send back the hard drive I just got would kinda suck :) On Mon, Jul 27, 2009 at 1:42 PM, Jason Granat j...@slash128.com wrote: Hi Mark, Sorry to hijack the thread but wanted to know if the new VoD's you mention will be available for download to current customers this Friday or will they be shipped only? Thanks, Jason -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Snow Sent: Monday, July 27, 2009 1:22 PM To: Aamir Panjwani Cc: ccie_voice@onlinestudylist.com; Michael Ciarfello Subject: Re: [OSL | CCIE_Voice] Any primer on the international escape character? Aamir, As you mentioned, I am hard at work on the new VoDs. ('cept when I take a few moments to answer Q's via this medium :P ). They will be released by Friday - end of this week, and shipping out the door first thing Monday (week from today). I do cover Call Routing in exhaustive detail (in fact it is a 4.5 hour section alone) - and I do go great detail into +dialing, Globalization, Localization, Calling Party XFormation CSS, etc. Also as Kevin mentioned - if you went or know someone who went to Networkers from your company - those Breakout Session PDF's are a great resource! Back to recording. Cheers, -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: ms...@ipexpert.com -- Join our free online support and peer group communities: http://www.IPexpert.com/communitieshttp://www.ipexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On- Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Jul 24, 2009, at 10:41 PM, Aamir Panjwani wrote: I would also like to know how people are preparing for topics like +dialing, calling party normalization (localizing the calling number, globalizing the calling party number, mapping the global party number to its local variant). I have read above mentioned topics in SRND few times and it is extremely confusing and poorly explained to say the least! Mark Snow wrote calling party normalization part 1 back in March on ipexpert blog which was explained in simple english, I know he is extremely busy making VOD for us but I hope he can write part 2 soon for us :) By the way does anyone know whether these topics are covered in the 2 new volume 2 labs released recently? Thanks -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Michael Ciarfello Sent: Saturday, 25 July 2009 12:09 PM To: Jonathan Charles; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Any primer on the international escape character? Can you re-phrase that? Like SRND 7.x, page 10-65 escape character? From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jonathan Charles [jonv...@gmail.com] Sent: Friday, July 24, 2009 4:01 PM To: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net Subject: [OSL | CCIE_Voice] Any primer on the international escape character? Anyone have a guide to how to configure it? To make sure sure telco sends it, and it is received, etc.? Jonathan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http
[OSL | CCIE_Voice] Lab 4A style versus function
looking for opinions on what I could be missing on 2 facets of the tasks. There are 2 things that I'm looking at that are style points that I want to get opinions on. for the calls out via the E1-PRI from the HQ/BR1 sites. creating the dial peer fo the outbound call, is there an advantage / necessity to creating it with a 901134T destination pattern versus simply making it 901134? since we're obviously dealing with the T.302 timer on the UCM side, we won't be adding more digits once the call is submitted to the h.323 gateway (BR2 router). adding a T on the end of the destination patter adds ambiguity and seems sloppy. the other is regarding inbound calls to the CME router. I know that creating the translation rule and forwarding the call as it comes in is sexier, but the same can be accomplished by creating an ephone-dn w/ number 3000 and CFwdAll to 1002. that is very likely over simplifying it, but I can't find anything in this question that would predicate us from doing so. That being said, I know that the exercise of this is to gain comfort with more complex configuration and structure. BUT, that complexity would cost you a little valuable time. what caveats am I missing that would make the CFwdAll solution fail? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Lab2A Q2.4
I’m going back over some of the earlier labs again and I think I’m just overanalyzing something. The question states to use Codec G711u for all calls. It’s under the SIP Endpoint section and I’m weak on SIP. Is there something in SIP that needs to be done or is this simply a statement to manipulate the Regions so that the inter region default isn’t used? Thanks for your input, MT ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] silly doubt regarding lab2A
Joel, a few key points: 1)Only 1 UCM Group can have auto registration enabled. if you have 12 groups in the UCM cluster, still only 1 can have auto reg. NOTE that this means that no matter WHAT device pool you WANT a phone to end up in, it's going to take the auto-register information from the UCM config and apply it to that phone. 2) you can have all 3 servers in your UCM group (if you had as many) with auto reg turned on. style points whether each of those servers have different DN ranges or they pull from the same 'pool' of numbers...I personally try to break them up (helps you tell which server assigned the DN). which leads me to my next point... 3) I don't know if this is a technical edict or just experience speaking. The first UCM server in the UCM group will exhaust its entire pool of numbers before it goes to the second server to look for a DN (keeping in mind those pesky 'unassigned numbers in the Route Plan Report). The PFM as it relates to getting a DN of 1000, probably a snapshot sequencing issue. the only downside I've found with how quickly these guys busted their asses to get these products out as fast as they did. All in all, I'm good with a few typos :) On Sat, Jul 4, 2009 at 11:05 AM, Joel Jose joeljose...@gmail.com wrote: inorder to have HQ phones and B1phones autoregister.. how do we go about it?? i mean they have diff DN ranges right...so got a little confusing... in the PG the screenshots solving this are causing me to be confused... both pub and sub will have the same autoregister DN range?? and can more than 1 server have autoregistration enabled in the same cluster? how does in the subsequent screenshot the BR1 phone gets DN of 1000 range automagically?? thanks for your time, joel. -- it's not the years in your life that count. It's the life in your years. Abraham Lincoln ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] New School switchport config with QOS
Ryan, I think that as long as you have the mls qos dscp mapping in place (cos to dscp), it shouldn't matter which you trust. The EF / CoS 5 data should be treated the same. Can anyone confirm / deny? _ From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ryan Trauernicht Sent: Monday, January 12, 2009 9:27 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] New School switchport config with QOS For Campus QOS. If you configure a port on the 3550: int fa0/1 switchport mode access switchport access vlan 100 switchport voice vlan 200 spanning-tree portfast Does the mls qos trust cos still apply since it isnt a dot1q port. COS is a 802.1q header. Or would you need to change it to mls qos trust dscp? Also what is the proper configuration for an ATA. Since there is no PC port do we want to still configure it as a trunk or dedicate it to only the voice vlan? Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] calling number passed to pstn although itset to restricted
Is it actually showing up on the PSTN phone or do you just see it in the Q.931 debug. Technically, when you set CLID to restricted, the number still presents on the line. If the receiving side does not respect the restricted switch setting, then the call can end up being displayed on the receiving phone. If you do the debug, do you see the calling party name as restricted? Im assuming that this isnt on a 6608 port. _ From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of jeremy co Sent: Saturday, January 10, 2009 5:31 PM To: Sergio Polizer Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] calling number passed to pstn although itset to restricted Hi, Thanx, but in my scenario I want to restrict one particular RP, so I should allow on GW. I use MGCP GW. any idea How can I do this? Cheers Jeremy On Sun, Jan 11, 2009 at 2:07 AM, Sergio Polizer spoli...@hotmail.com wrote: I think that in this case it will block the CLID for all calls that are going out through this gw. If you keep the GW with CLID=default (not allowed) is just fine and your config at the RP to restrict the CLID will works. Sergio. _ From: narinder.ku...@uxcg.com.au To: jeremy.coo...@gmail.com; ccie_voice@onlinestudylist.com Date: Sat, 10 Jan 2009 22:26:23 +1100 Subject: Re: [OSL | CCIE_Voice] calling number passed to pstn although it set to restricted What kind of GW are you using MGCP or H323. Check the GW configuration on CCM the GW should also have the CLID and CNAME restricted. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of jeremy co Sent: Saturday, 10 January 2009 7:56 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] calling number passed to pstn although it set to restricted Hi, I set calling line ID presentation Restricted calling name presentation Restricted on Route Pattern calling name is not show up but I can still see my DID phone number (2001) on pstn phone. it should not pass it to pstn, any idea why this happened? Jeremy _ CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not Getronics Australia. While we endeavour to protect our network from computer viruses, Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained in this email before loading onto any computer system. _ Notícias direto do New York Times, gols do Lance, videocassetadas e muitos outros vídeos no MSN Videos! Confira já! http://video.msn.com/?mkt=pt-br
Re: [OSL | CCIE_Voice] calling number passed to pstn although itset to restricted
Good test mechanism Jeremy, do you show 2001 while its alerting, after connection, or both? _ From: anil batra [mailto:anil...@yahoo.com] Sent: Saturday, January 10, 2009 10:00 PM To: Michael Thompson; jeremy co Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] calling number passed to pstn although itset to restricted Did you try using TP insated of RP. I mean create a TP 9[2-9]XX and restrict name and number there and then check... --- On Sun, 1/11/09, jeremy co jeremy.coo...@gmail.com wrote: From: jeremy co jeremy.coo...@gmail.com Subject: Re: [OSL | CCIE_Voice] calling number passed to pstn although itset to restricted To: Michael Thompson mthompson...@gmail.com Cc: ccie_voice@onlinestudylist.com Date: Sunday, January 11, 2009, 8:17 AM Thanx Michael, scenario is : ipphone---ccm-- GW(MGCP) ---PSTN --PSTN_Phone Well it should show up in pstn phone. Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Calling Party Number i = 0x0081, '2001' Plan:Unknown, Type:Unknown Called Party Number i = 0x80, '0111234' Plan:Unknown, Type:Unknown Mar 11 16:26:42.989: ISDN Se0/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x8003 Channel ID i = 0xA98383 Exclusive, Channel 3 I cannot even see calling name as restricted , just it would not show p on pstn phone. but I can see 2001 on pstn phone. I want to restrict calling name and number for particular RP, so putting restrict on GW is not an option. Any idea? Regards, Jeremy On Sun, Jan 11, 2009 at 1:02 PM, Michael Thompson mthompson...@gmail.com wrote: Is it actually showing up on the PSTN phone or do you just see it in the Q.931 debug. Technically, when you set CLID to restricted, the number still presents on the line. If the receiving side does not respect the restricted 'switch' setting, then the call can end up being displayed on the receiving phone. If you do the debug, do you see the calling party name as restricted? I'm assuming that this isn't on a 6608 port. _ From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of jeremy co Sent: Saturday, January 10, 2009 5:31 PM To: Sergio Polizer Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] calling number passed to pstn although itset to restricted Hi, Thanx, but in my scenario I want to restrict one particular RP, so I should allow on GW. I use MGCP GW. any idea How can I do this? Cheers Jeremy On Sun, Jan 11, 2009 at 2:07 AM, Sergio Polizer spoli...@hotmail.com wrote: I think that in this case it will block the CLID for all calls that are going out through this gw. If you keep the GW with CLID=default (not allowed) is just fine and your config at the RP to restrict the CLID will works. Sergio. _ From: narinder.ku...@uxcg.com.au To: jeremy.coo...@gmail.com; ccie_voice@onlinestudylist.com Date: Sat, 10 Jan 2009 22:26:23 +1100 Subject: Re: [OSL | CCIE_Voice] calling number passed to pstn although it set to restricted What kind of GW are you using MGCP or H323. Check the GW configuration on CCM the GW should also have the CLID and CNAME restricted. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of jeremy co Sent: Saturday, 10 January 2009 7:56 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] calling number passed to pstn although it set to restricted Hi, I set calling line ID presentation Restricted calling name presentation Restricted on Route Pattern calling name is not show up but I can still see my DID phone number (2001) on pstn phone. it should not pass it to pstn, any idea why this happened? Jeremy _ CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not Getronics Australia. While we endeavour to protect our network from computer viruses, Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained in this email before loading onto any computer system. _ Notícias direto do New York Times, gols
Re: [OSL | CCIE_Voice] Low Volume On Forwarded Calls
Amp, I would have to assume that the Call fwd destination is the same type of trunk, if not the same port (i.e. in and out PRI, or in and out the same FXO ports). I've seen forwarded calls from FXO to FXO show similar issues because a high input attenuation combined with a negative input gain combines for low volumes. Can you give a more 'physical' topology? -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of amccar...@cciequest.com Sent: Tuesday, December 30, 2008 1:59 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Low Volume On Forwarded Calls Hey gang, I have a question for all of you gurus and experts. Have any of you ever run into a problem where one of your callers dials an extension (in another country) that has been forwarded to a cell phone, the calls audio volume is low, but when they call the cell phone directly the audio volume is fine? Thanks, Amp
Re: [OSL | CCIE_Voice] Calling line id restriction on a per call basisfor Call Manager
Simple version is to use a translation pattern. On that translation pattern, set the CLID Name and number to restricted. You can do the same on the route pattern if needed. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jonny vegas Sent: Saturday, November 01, 2008 2:59 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Calling line id restriction on a per call basisfor Call Manager Just browsing through one of the well know books and it mentions of this feature and ccm 4.1 Does anyone know how to configure it for 4.1? Thanks
Re: [OSL | CCIE_Voice] Current version in LAB - CCM SR?
Technically you can expect any release of 4.1.3 that's out there. Depending on when you take the lab, I would doubt that they would all be updated up to sr8, but you just don't know. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shadab Abbasi (moabbasi) Sent: Thursday, October 30, 2008 2:13 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Current version in LAB - CCM SR? People, What would be the current CCM SR in the lab? I see SR8 is released, can we expect this one? Regards, Shadab Abbasi TSN SE - Unified Communications Technology Solutions Network (TSN) [EMAIL PROTECTED] Ph: +91.80.4103.6436 (off)+91.974.009.0334(Mob) TSN-WiKi: Home http://gsops-wiki.cisco.com/confluence/display/TSNKB/Home Page
Re: [OSL | CCIE_Voice] Alias then Redirect IE 12.4.3j - Not Healthy
Check the RDNIS of the Q.931 debug. Check for the value 0xFF right before the RDNIS number. Is it present? _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jonny vegas Sent: Wednesday, October 01, 2008 11:48 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Alias then Redirect IE 12.4.3j - Not Healthy Following on from previous message. Test 12.4.3j on 2821. Scenario SRST mode. alias from XXX2 to XXX1 cfw HQ-VM This scenario still has rubbish Redirect IE. Test show best to leave the isdn outgoing ie redirect off the ISDN since it also seems to break VM-Integration.
Re: [OSL | CCIE_Voice] IPExpert workbook last updated
Only change in the last 2 months that I'm aware of is the removal of the documents CD and the use of the Web based documentation. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rachell Thornton Sent: Wednesday, October 01, 2008 1:15 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] IPExpert workbook last updated When was the IPExpert voice lab workbook last updated? I have been told that the lab exam has been modified during the past two months.
Re: [OSL | CCIE_Voice] ISDN Redirecting number 12.4.3j - looking morehealthy
Jonny, I'm confused. You're getting the correct RDNIS now, correct? What's to test? There's a bug in early 12.4 codes (and I'm not sure if it's late 12.3) that sets the IE element value to FF and essentially strips the RDNIS value. That isn't fixed until versions later in 12.4. versions that we'll not likely see in the lab. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jonny vegas Sent: Wednesday, October 01, 2008 11:31 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] ISDN Redirecting number 12.4.3j - looking morehealthy Test run on 12.4.3j on 2821 Debug ISDN Q931 on SRST rtr shows some positive info Redirecting number IE is no longer 0xFF rubbish. It is being interpreted by CCM and forwarded to unity. Does anyone have 37xx rtrs to test this on? Perhaps worth checking before doing anything too fancy. Anyone got anything to add?
Re: [OSL | CCIE_Voice] CUE: Enable Debugs
In CUE they're referred to as traces... First thing is to do a no trace all From there, trace ? and that will give you an overview of what's available. I think that once you change your search syntax to trace versus debug you'll get quite a bit more info. MT -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Schuknecht Sent: Thursday, October 02, 2008 5:00 PM To: OSL CCIE Voice Lab Exam Subject: [OSL | CCIE_Voice] CUE: Enable Debugs Hi List, were can i find some information for enabling debugs, for example an debug command reference, for the CUE Module. I would like to be able to do some debugs of VPIM, SIP Signalling, Voicemail access and so on On CCO i could not find anything. /Robert
Re: [OSL | CCIE_Voice] PSTN config - International
and I doubt this is an issue, but if you have any Locations CAC set up in the config, make sure that AAR isn't prepending anything to the call. On Tue, Aug 19, 2008 at 9:19 PM, Kevin Porter [EMAIL PROTECTED]wrote: Check each route group member in the route list and make sure there is not a 9 in the Prefix Digits (outgoing calls) box for Called Party Transformations… *Kevin Porter** Systems Engineer L4* Netelligent Corporation 400 South Woods Mill Drive, Suite 105 St. Louis, MO 63017 Office: (314) 392-6921 Cell: (314) 852-1252 Fax: (314) 392-9760 [EMAIL PROTECTED] www.netelligent.com Bridging The Gap Between Good and GREAT IP Communications! -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *cisco.voip *Sent:* Tuesday, August 19, 2008 7:51 PM *To:* Jonathan Charles *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] PSTN config - International OK, I put the translstion in to get it working, but I am still wondering why the CCM Router List is not stripping the PreDot, as the router pattern in CCM is 9.011!, I would expect only 0119876543 to be sent to the PSTN router? - Original Message - *From:* Jonathan Charles [EMAIL PROTECTED] *To:* cisco.voip [EMAIL PROTECTED] *Cc:* ccie_voice@onlinestudylist.com *Sent:* Tuesday, August 19, 2008 4:41 AM *Subject:* Re: [OSL | CCIE_Voice] PSTN config - International You need to translate 90119876543 to remove the 9. Jonathan On Mon, Aug 18, 2008 at 9:25 PM, cisco.voip [EMAIL PROTECTED] wrote: Can someone help me with my PSTN config using the ipexpert supplied PSTN configuration. The PSTN router is CME on a 3825 with a 7960 phone. The problem I have is getting the PSTN international line to ring. All other lines are working, I see the call coming into the PSTN router, however I receive the unassigned number error. ephone-dn 1 number 911 no-reg primary ! ! ephone-dn 2 number 212225 no-reg primary ! ! ephone-dn 3 number 617525 no-reg primary ! ! ephone-dn 4 number 331325 no-reg primary ! ! ephone-dn 5 number 0119876543 no-reg primary ephone 1 device-security-mode none mac-address 000D.BD38.82D8 type 7960 button 1:1 2:2 3:3 4:4 button 5:5 PSTN# .Aug 19 02:24:05.901: ISDN Se0/0/1:23 Q931: RX - SETUP pd = 8 callref = 0x0002 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Display i = 'HQ PHONE 2' Calling Party Number i = 0x0081, '1002' Plan:Unknown, Type:Unknown Called Party Number i = 0x80, '90119876543' Plan:Unknown, Type:Unknown .Aug 19 02:24:05.909: ISDN Se0/0/0:23 Q931: pak_private_number: Invalid type/plan 0x0 0x0 may be overriden; sw-type 13 .Aug 19 02:24:05.909: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is 0x2 0x1, Called num 0119876543 .Aug 19 02:24:05.909: ISDN Se0/0/1:23 Q931: TX - CALL_PROC pd = 8 callref = 0x8002 Channel ID i = 0xA98383 Exclusive, Channel 3 .Aug 19 02:24:05.909: ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref = 0x0081 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Display i = 'HQ PHONE 2' Calling Party Number i = 0x0081, '1002' Plan:Unknown, Type:Unknown Called Party Number i = 0xA1, '0119876543' Plan:ISDN, Type:National .Aug 19 02:24:05.933: ISDN Se0/0/0:23 Q931: RX - RELEASE_COMP pd = 8 callref = 0x8081 Cause i = 0x8081 - Unallocated/unassigned number .Aug 19 02:24:05.937: ISDN Se0/0/1:23 Q931: TX - DISCONNECT pd = 8 callref = 0x8002 Cause i = 0x8281 - Unallocated/unassigned number .Aug 19 02:24:05.985: ISDN Se0/0/1:23 Q931: RX - RELEASE pd = 8 callref = 0x0002 .Aug 19 02:24:05.985: ISDN Se0/0/1:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x8002
Re: [OSL | CCIE_Voice] [cisco-voip] FYI: VMware ESXi is now free
Has anyone played with installing it on generic hardware though? My problem with ESX is that I tried to install it on my commercial NVidia chipset based MoBo and it bombed. MT _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carter, Bill Sent: Wednesday, July 30, 2008 10:42 AM To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Subject: [cisco-voip] FYI: VMware ESXi is now free http://www.vmware.com/download/esxi/
Re: [OSL | CCIE_Voice] [cisco-voip] FYI: VMware ESXi is now free
Bummer, I was hoping that they relaxed that a little so I could get rid of the Microsoft overhead of my server. Don't know enough about Linux to install on that. May have to learn Ubuntu. -Original Message- From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 30, 2008 12:08 PM To: Michael Thompson Cc: Carter, Bill; [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Subject: Re: [cisco-voip] FYI: VMware ESXi is now free you need a server from the hardware compatibility list... Jonathan On Wed, Jul 30, 2008 at 10:18 AM, Michael Thompson [EMAIL PROTECTED] wrote: Has anyone played with installing it on generic hardware though? My problem with ESX is that I tried to install it on my commercial NVidia chipset based MoBo and it bombed. MT From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carter, Bill Sent: Wednesday, July 30, 2008 10:42 AM To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Subject: [cisco-voip] FYI: VMware ESXi is now free http://www.vmware.com/download/esxi/ ___ cisco-voip mailing list [EMAIL PROTECTED] https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [OSL | CCIE_Voice] [cisco-voip] FYI: VMware ESXi is now free
I just meant as a platform to install the VMWare on. I have VMServer (free version) running on Server 2008. I'm running 08 because I have 4Gb or RAM and XP wasn't acknowledging all my memory (which kinda pissed me off a LOT). My thought is to install Linux as a base OS (less overhead resources meaning more available for my VMWare). Not to mention, Vista and 2008 don't like NVidia chipset MoBo hardware. They're intel bigots. -Original Message- From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 30, 2008 12:31 PM To: Michael Thompson Cc: Carter, Bill; [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Subject: Re: [cisco-voip] FYI: VMware ESXi is now free You don't need to know Linux... The installation is quite easy... just next a lot (add IPs and stuff...) and then after it completes it will tell you to open a web browser and then install the infrastructure 3 client.then you can do VMWare stuff... Personally, I think Workstation is more suitable for our needs (I am running Workstation 6) and I have CCM 4.1/IPCC a Unity 4 box, a CUCM 6.1 box and Unity 7 box, plus an XP VM for stuff... Jonathan On Wed, Jul 30, 2008 at 11:17 AM, Michael Thompson [EMAIL PROTECTED] wrote: Bummer, I was hoping that they relaxed that a little so I could get rid of the Microsoft overhead of my server. Don't know enough about Linux to install on that. May have to learn Ubuntu. -Original Message- From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 30, 2008 12:08 PM To: Michael Thompson Cc: Carter, Bill; [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Subject: Re: [cisco-voip] FYI: VMware ESXi is now free you need a server from the hardware compatibility list... Jonathan On Wed, Jul 30, 2008 at 10:18 AM, Michael Thompson [EMAIL PROTECTED] wrote: Has anyone played with installing it on generic hardware though? My problem with ESX is that I tried to install it on my commercial NVidia chipset based MoBo and it bombed. MT From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carter, Bill Sent: Wednesday, July 30, 2008 10:42 AM To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Subject: [cisco-voip] FYI: VMware ESXi is now free http://www.vmware.com/download/esxi/ ___ cisco-voip mailing list [EMAIL PROTECTED] https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [OSL | CCIE_Voice] [cisco-voip] FYI: VMware ESXi is now free
I thought the big draw of ESX is light weight and manageability for server clusters. -Original Message- From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 30, 2008 12:45 PM To: Carter, Bill Cc: Michael Thompson; [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Subject: Re: [cisco-voip] FYI: VMware ESXi is now free You can get a DL 380 G3 for like $300 on ebay Just a word to the wise, ESX is not a lightweight application... it needs an insane amount of memory (32 or 64GB is a good start) and LOTS of processing space... I tried it on a DL380 G3 with 8GB of RAM and dual 2.4Ghz Xeons and it could barely run a single CCM 4.1 VM (and it was crawling... phones would take 30-45 seconds to get dial tone when they went offhook). Jonathan On Wed, Jul 30, 2008 at 11:42 AM, Carter, Bill [EMAIL PROTECTED] wrote: ESXi installation just bombed on a MCS-7815i-3.0. The IBM X306 server is not in the hardware compatibility list. -Original Message- From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 30, 2008 11:39 AM To: Michael Thompson Cc: Carter, Bill; [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Subject: Re: [cisco-voip] FYI: VMware ESXi is now free Yeah, that won't work, ESX is very hardware specific Just a side note, because I see this misunderstanding a lot. On a 32-bit OS, you have a 32-bit memory address space, which equates to roughly 4 billion address locations (about 4GB of RAM supported), however, Windows allocates a nice sized chunk to I/O and other stuff... If you want to recognize more than about 3.5GB, you need to run a 64-bit OS (I am running 64-bit Vista with no problems (except the silly CIsco VPN client doesn't work)... Jonathan On Wed, Jul 30, 2008 at 11:33 AM, Michael Thompson [EMAIL PROTECTED] wrote: I just meant as a platform to install the VMWare on. I have VMServer (free version) running on Server 2008. I'm running 08 because I have 4Gb or RAM and XP wasn't acknowledging all my memory (which kinda pissed me off a LOT). My thought is to install Linux as a base OS (less overhead resources meaning more available for my VMWare). Not to mention, Vista and 2008 don't like NVidia chipset MoBo hardware. They're intel bigots. -Original Message- From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 30, 2008 12:31 PM To: Michael Thompson Cc: Carter, Bill; [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Subject: Re: [cisco-voip] FYI: VMware ESXi is now free You don't need to know Linux... The installation is quite easy... just next a lot (add IPs and stuff...) and then after it completes it will tell you to open a web browser and then install the infrastructure 3 client.then you can do VMWare stuff... Personally, I think Workstation is more suitable for our needs (I am running Workstation 6) and I have CCM 4.1/IPCC a Unity 4 box, a CUCM 6.1 box and Unity 7 box, plus an XP VM for stuff... Jonathan On Wed, Jul 30, 2008 at 11:17 AM, Michael Thompson [EMAIL PROTECTED] wrote: Bummer, I was hoping that they relaxed that a little so I could get rid of the Microsoft overhead of my server. Don't know enough about Linux to install on that. May have to learn Ubuntu. -Original Message- From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 30, 2008 12:08 PM To: Michael Thompson Cc: Carter, Bill; [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Subject: Re: [cisco-voip] FYI: VMware ESXi is now free you need a server from the hardware compatibility list... Jonathan On Wed, Jul 30, 2008 at 10:18 AM, Michael Thompson [EMAIL PROTECTED] wrote: Has anyone played with installing it on generic hardware though? My problem with ESX is that I tried to install it on my commercial NVidia chipset based MoBo and it bombed. MT From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carter, Bill Sent: Wednesday, July 30, 2008 10:42 AM To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Subject: [cisco-voip] FYI: VMware ESXi is now free http://www.vmware.com/download/esxi/ ___ cisco-voip mailing list [EMAIL PROTECTED] https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [OSL | CCIE_Voice] [cisco-voip] FYI: VMware ESXi is now free
Exactly, and that's how I ended up on Server 2008. tried vista x64 and ran into a bunch of PITA problems. Once everything came out in the wash as far as Windows overhead, I ended up with somewhere between 3.2 and 3.4Gb or RAM. Long story short, after my lab attempt next week I'm probably going to migrate this box to dual boot XP (for gaming and cheesy stuff like that) and Ubuntu (or some other simple open source Linux) for the VM pieces. -Original Message- From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 30, 2008 12:39 PM To: Michael Thompson Cc: Carter, Bill; [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Subject: Re: [cisco-voip] FYI: VMware ESXi is now free Yeah, that won't work, ESX is very hardware specific Just a side note, because I see this misunderstanding a lot. On a 32-bit OS, you have a 32-bit memory address space, which equates to roughly 4 billion address locations (about 4GB of RAM supported), however, Windows allocates a nice sized chunk to I/O and other stuff... If you want to recognize more than about 3.5GB, you need to run a 64-bit OS (I am running 64-bit Vista with no problems (except the silly CIsco VPN client doesn't work)... Jonathan On Wed, Jul 30, 2008 at 11:33 AM, Michael Thompson [EMAIL PROTECTED] wrote: I just meant as a platform to install the VMWare on. I have VMServer (free version) running on Server 2008. I'm running 08 because I have 4Gb or RAM and XP wasn't acknowledging all my memory (which kinda pissed me off a LOT). My thought is to install Linux as a base OS (less overhead resources meaning more available for my VMWare). Not to mention, Vista and 2008 don't like NVidia chipset MoBo hardware. They're intel bigots. -Original Message- From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 30, 2008 12:31 PM To: Michael Thompson Cc: Carter, Bill; [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Subject: Re: [cisco-voip] FYI: VMware ESXi is now free You don't need to know Linux... The installation is quite easy... just next a lot (add IPs and stuff...) and then after it completes it will tell you to open a web browser and then install the infrastructure 3 client.then you can do VMWare stuff... Personally, I think Workstation is more suitable for our needs (I am running Workstation 6) and I have CCM 4.1/IPCC a Unity 4 box, a CUCM 6.1 box and Unity 7 box, plus an XP VM for stuff... Jonathan On Wed, Jul 30, 2008 at 11:17 AM, Michael Thompson [EMAIL PROTECTED] wrote: Bummer, I was hoping that they relaxed that a little so I could get rid of the Microsoft overhead of my server. Don't know enough about Linux to install on that. May have to learn Ubuntu. -Original Message- From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 30, 2008 12:08 PM To: Michael Thompson Cc: Carter, Bill; [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Subject: Re: [cisco-voip] FYI: VMware ESXi is now free you need a server from the hardware compatibility list... Jonathan On Wed, Jul 30, 2008 at 10:18 AM, Michael Thompson [EMAIL PROTECTED] wrote: Has anyone played with installing it on generic hardware though? My problem with ESX is that I tried to install it on my commercial NVidia chipset based MoBo and it bombed. MT From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carter, Bill Sent: Wednesday, July 30, 2008 10:42 AM To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Subject: [cisco-voip] FYI: VMware ESXi is now free http://www.vmware.com/download/esxi/ ___ cisco-voip mailing list [EMAIL PROTECTED] https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [OSL | CCIE_Voice] This message is on behalf of Jane Ryer
Well technically we know which one to use, but not which to avoid. Don't see any reference to the version of code she was running. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Snow Sent: Tuesday, July 29, 2008 2:43 PM To: Mark Cardwell; Jane Ryer (jryer); Matthew Bynum Cc: CCIE Voice Maillist; Vik Malhi; Glenn Champine Subject: Re: [OSL | CCIE_Voice] This message is on behalf of Jane Ryer Importance: High All, After working extensively to try to figure out the root of the problem that was reported here a few days back regarding IOS Routers running EasyVPN connected to ProctorLabs for use with our Voice vRacks - Jane Ryer successfully deduced, debugged, and has decided that the issue was not related to anything on ProctorLabs network - but instead was due to a documented IOS bug on her local client router that somehow just happened to pop up in the past 30-45 days. The magic bullet? 12.4.21 Mainline IOS code. Up or Downgrade - depending on where you are. She and myself both tested this to be a definite fix to the issue which involved a malformed reassembly of NAT'd Skinny packets. Many, many thanks to Jane Ryer and her determination to find the problem!! That's the kind of dedicated troubleshooting it takes to study for, and pass the CCIE Lab! (even if this troubleshooting didn't help her for the Voice lab ;-). But hey - she and we all know what IOS to avoid for our consulting clients if we needed to use EasyVPN and Skinny! -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] -- Join our free online support and peer group communities: http://www.IPexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Jul 25, 2008, at 7:02 AM, Cardwell, Mark wrote: I have experienced the same issue. I use the easy vpn and it worked great until about 1-1.5 months ago and then no go, phones would never resister. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Lepla Sent: Thursday, July 24, 2008 6:21 PM To: ccie_voice@onlinestudylist.com Cc: 'Glenn Champine' Subject: [OSL | CCIE_Voice] This message is on behalf of Jane Ryer From: Jane Ryer (jryer) Sent: Thursday, July 24, 2008 3:44 PM To: 'ccie_voice@onlinestudylist.com' Subject: phones will not load Softkey Template through EasyVPN tunnel Is there anyone else out there using the EasyVPN solution and hardware phones to connect to Proctor Lab racks? Have you done so successfully in the last three weeks? My home setup consists of one router to bring up the EasyVPN tunnel, and four hardware phones (two 7960's, two 7961's). I have used this setup repeatedly over the last six months for more than 60 rack sessions, with no problems getting the phones to auto-register until the last month. I have booked four sessions in the last month and was only able to actually accomplish any study scenarios during one of the four sessions. For the other three sessions, I have had issues auto-registering some or all of my phones (to both Call Manager and CME) where they download the default config file, but when they then request the SoftKey template, they just hang with a blank display. I sent some debug output (from the BR2 router running CME) to Proctor Labs support Monday night, along with the complete configuration of my EasyVPN router and sho ip nat translations. Mark Snow worked with me a little on Tuesday evening, but we were not able to resolve the issue. The reason for this post is to just gather information about whether EasyVPN connections have been working fine for other people during the last few weeks. If the answer to that is yes, then I'll focus on what might have changed in my setup. (I did see a comcast domain name and some DNS server addresses in my phones at one point on Monday night, and don't remember ever seeing that before.) If anyone else has had similar issues, then I'd like to compare notes. Thanks, Jane If I can be of further assistance, please let me know. -- Drew LePla Comp TIA A+, CCNA Technical Support Engineer - IPexpert, Inc. Telephone: +1.810.326.1444 x204 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] -- Join our free online support and peer group communities: http://www.IPexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- Mark Cardwell | Systems Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive,
Re: [OSL | CCIE_Voice] Documentation CD for use during LAB
We Need Feedback From THE FROG. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Charles Sent: Monday, July 28, 2008 10:16 AM To: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] Documentation CD for use during LAB Anyone? On Sat, Jul 26, 2008 at 6:08 PM, Jonathan Charles [EMAIL PROTECTED] wrote: OK, so, Cisco has completely reconfigured their DocCD... so, what replaced it on the lab? I want to play with/familiarize myself with the locations of stuff, but I don't know what is currently being used on the lab (first attempt will be Sept 8) Any feedback would be much appreciated Jonathan
Re: [OSL | CCIE_Voice] Documentation CD for use during LAB
I doubt that a question of 'what documentation is on the desktop' falls into the NDA. I can see a question of did you have GK via IPIPGW, or a question of service configuration scenarios, but what .pdfs are on the desktop. That might be extending the spirit of the agreement a bit. That being said, if you don't feel comfortable, that's 100% your right to do so. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Arevalo Sent: Monday, July 28, 2008 7:01 PM To: Jonathan Charles Cc: OSL CCIE Voice Lab Exam; James Key Subject: Re: [OSL | CCIE_Voice] Documentation CD for use during LAB I was on the test, so i cannot say anymore... remember the NDA, But if i were you wouldnt hesitate to use it as i did! //r.a. On Mon, Jul 28, 2008 at 6:40 PM, Jonathan Charles [EMAIL PROTECTED] wrote: Is this what is actually on the lab? Or something that just happens to work? Jonathan On Mon, Jul 28, 2008 at 5:36 PM, Ricardo Arevalo [EMAIL PROTECTED] wrote: Jonathan Dont use any drop down menu... in cisco.com/univercd use this link instead: All Product Documentation rgds//r.a. On Mon, Jul 28, 2008 at 6:06 PM, Jonathan Charles [EMAIL PROTECTED] wrote: Yeah, following that path, I get a dead link as soon as I click on CallManager Express from the drop down... so much for learning how that worked Jonathan On Mon, Jul 28, 2008 at 4:33 PM, Ricardo Arevalo [EMAIL PROTECTED] wrote: The first time i went to RTP they had on desktop QoS SRND 3.3 and some services' pdfs like CSC Agent config, i think they have around 4 PDFs in the desktop There is a DocCD very needed in case there is no file on the flash about b-acd, you just have to remember the new path: http://www.cisco.com/univercd All Product Documentation Voice Products Cisco CallManager Express and Cisco IOS Telephony Cisco CallManager Express (CME) was formerly known as Cisco IOS Telephony Services (ITS). Cisco Unified CallManager Express 4.0 Cisco Unified CME B-ACD and Tcl Call-Handling Applications Cisco Unified CME Basic Automatic Call Distribution and Auto-Attendant Service (B-ACD) I tried this path and worked on the test. Rgds//r.a. On Mon, Jul 28, 2008 at 3:22 PM, James Key [EMAIL PROTECTED] wrote: I was going to ask the same question. Tried using the DocCD over the weekend for CME and the link redirection sends you to a page that is no longer available. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Charles Sent: Monday, July 28, 2008 9:16 AM To: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] Documentation CD for use during LAB Anyone? On Sat, Jul 26, 2008 at 6:08 PM, Jonathan Charles [EMAIL PROTECTED] wrote: OK, so, Cisco has completely reconfigured their DocCD... so, what replaced it on the lab? I want to play with/familiarize myself with the locations of stuff, but I don't know what is currently being used on the lab (first attempt will be Sept 8) Any feedback would be much appreciated Jonathan NOTICE: This electronic mail message and any files transmitted with it are intended exclusively for the individual or entity to which it is addressed. The message, together with any attachment, may contain confidential and/or privileged information. Any unauthorized review, use, printing, saving, copying, disclosure or distribution is strictly prohibited. If you have received this message in error, please immediately advise the sender by reply email and delete all copies.
Re: [OSL | CCIE_Voice] Hardware failures at San Diego resolved?
I'll tell you next Thursday. I just hope so, can't afford the lost clock cycles. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Charles Sent: Monday, July 28, 2008 8:13 PM To: OSL CCIE Voice Lab Exam Subject: [OSL | CCIE_Voice] Hardware failures at San Diego resolved? Lots of talk about bad 6608 blade and hyper-slow CCM response... wonder if this has been resolved... Jonathan
[OSL | CCIE_Voice] volume 1 section 4 of study guide. ICT / SIP trunk
I'm looking through the proctors guide on the explanation of the IPIPGW and there's one piece that doesn't make sense. Task 4.9 has you create both an H.323 (non-GK) ICT to the IPIPGW, that part make sense. what confuses me is why we're creating SIP trunk on CCM if traffic from BR2 should be coming via the IPIPGW. shouldn't that IPIPGW be translating the traffic from SIP to H.323? I could see if it was GK controlled ip calls, then the traffic comes across as native SIP, but this seems odd to me. anyone have any insight? Thanks, Mike
Re: [OSL | CCIE_Voice] disable VAD on ATA
but doesn't Fax turn off VAD by default? will that 'override' the audio mode if it's set to 0015? On Tue, Jul 15, 2008 at 1:21 PM, Mark Snow [EMAIL PROTECTED] wrote: You are exactly correct Juan - bit 0 set to 0! -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] -- Join our free online support and peer group communities: http://www.IPexpert.com/communities http://www.ipexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Jul 14, 2008, at 5:31 AM, Juan wrote: If you want to enable a ATA186 for fax passthrough, CCO recommends setting AudioMode to 0x00150015. But if I understand correctly, this enables VAD (LSB=1) - which is not supposed to be set for fax right? Therefore, should it be set to 0x00140014 instead? regards Juan
Re: [OSL | CCIE_Voice] Phone CoS Settings ?
folks, keep one thing in mind in the process of the PC packets getting to the switch infrastructure. the IP phone is a switch itself, so as packets come in port 0 (PC) and go out port 1 (to switch) those layer 2 headers are 'touched'. So, as it gets forwarded upstream (onto a 802.1q trunk, which is the only place that COS can exist) those packets can be marked (rewritten) with the appropriate COS value. On Wed, Jul 23, 2008 at 10:01 PM, Nick Marus [EMAIL PROTECTED] wrote: COS does exist in the 1q header in the 1p field, however the pc nic (most) has the ability to add this to the frame. (should see qos packet scheduler under you nic properties in windows) The 1q header can exist with out vlan tagging being set. unlike isl which re-encapsulates the ethernet frame inside an isl frame, adding the 1q header does not render the frame unreadable to non 1q devices. The 1q header is just inserted between the sa and the type/length field in the frame header. many devices that do not make use of 1q vlan tagging still use the 1p portion of the 1q header to mark their cos. A cisco switch that is setup for mode access will ignore the 1q vlan tagging and put the frame on the vlan that the switchport is configured to. But it will still look at the 1p portion of the 1q header if qos trust cos is configured on the port. Same thing goes for a computer attached to a phone. if the switch is configured to trust the cos marking of the device attached behind the phone, it will look at the 1p portion of the 1q header on the frames coming from the device attached to the phone and respect those values. otherwise it will have the phone remark those values before the frame gets to the switch and get prioritized. http://www.cisco.com/en/US/tech/tk389/tk689/technologies_tech_note09186a0080094665.shtml#topic1 On Wed, Jul 23, 2008 at 6:57 PM, Mike Brooks [EMAIL PROTECTED] wrote: Its my understanding that the CoS value is set in the 802.1p field within an 802.1q tag. Therefore, in order to set a CoS value you need have an 802.1q trunk. So a PC would not be able to set a CoS value, unless its uplink was an 802.1q trunk port, rather than an access port. So if the PC can't set the CoS value, why would you need to use the switchport priority extend cos 0 ? Please correct me if I am wrong. Regards, Mike Brooks CCIE#16027 (RS) On 7/23/08, Nick Marus [EMAIL PROTECTED] wrote: Unless I misunderstand you, COS is applied to a packet and does not require a 1q trunk. The connection between the pc and the phone is not 1q usually. Just the connection between the phone and the switch. Most PC nic's can be setup to mark it's packets with a cos value and effectively take priority on your switched network over you voice and other high priority packets if the switch is trusting and the phone is not remarking to 0. Nick On Wed, Jul 23, 2008 at 10:51 AM, Mike Brooks [EMAIL PROTECTED] wrote: Hi Everyone, I see that the standard practice on a switchport is to configure switchport priority extend cos 0 in order to allow the ip phone to reset the cos value received from the PC to 0. My question is how would a PC ever set a CoS value if the link between the ip phone and the PC is not an 802.1q trunk ? Can someone please help me understand this ? The only thing I can think of is that the PC would somehow have to support an 802.1q trunk to it, a trunk would have to be dynamically established between the phone and PC. And, then the user would have to manipulate the CoS value. Is this possible with a Cisco phone ? If this is the only case this would work then you would think that Cisco would document these pre-requisites. Perhaps I am confused. Please help ;-) Regards, Mike Brooks CCIE#16027 (RS) -- Nick Marus [EMAIL PROTECTED] -- Nick Marus [EMAIL PROTECTED]
[OSL | CCIE_Voice] IP Blue licensing
since we're using IP Blue in the lab, i wanted to download it and play with it a bit. problem is that the demo is 20 minutes and it resets. anyone know of a way around it short of shelling out the $600 for the 5 licenses I would need?? just trying to make things easier. MT
Re: [OSL | CCIE_Voice] [cisco-voip] Trusting CoS on router uplink from switch...
My assumption is that as long as you have your COS - DSCP mapping in place that you can do either. My preference is to do the DSCP values due to the granularity as well as similarity to what I normally do at customer sites (so it's more of a habit for me and one less thing to think about in the lab seems like a good thing). anyone in the know have any concrete thoughts to go with our assumptions? On Wed, Jul 23, 2008 at 12:32 PM, Jonathan Charles [EMAIL PROTECTED] wrote: Cos is recommended (per the QoS SRND) for any dot1q trunk... However, my concern is not the switch, but the voice gateway's Fast Ethernet to the switch... Do we need to trust outbound (switch to gateway), and inbound (on the gateway side...) Jonathan On Wed, Jul 23, 2008 at 11:28 AM, [EMAIL PROTECTED] wrote: Someone correct me if wrong and might actually depend on IOS and type of kit as well. As long as you don't have mls configured at all, you won't be rewriting the DSCP fields. If you have mls configured, you'll have to trust COS or DSCP whatever you prefer otherwise you'll overwrite with default value. I am not sure if it's better to trust on DSCP or COS in this case, If you are trunking you might as well stick to COS. Also you'll have wrr or srr queues setup which will map to COS. Haven't touched qos for a while and seems like I do have to freshen up on this topic ;) -Original Message- From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 23, 2008 5:18 PM To: Mayr, Thorsten: IT (LDN) Cc: cisco voip; OSL CCIE Voice Lab Exam Subject: Re: [cisco-voip] Trusting CoS on router uplink from switch... Well, I do have subinterfaces and it is a dot1q trunk to the gateway... just looking for a config to trust the CoS and DSCP... or wondering if the router will clear the DSCP and CoS Jonathan On Wed, Jul 23, 2008 at 9:38 AM, [EMAIL PROTECTED] wrote: Charles, You won't have COS on the link to the gateway as you won't have a dot1.q/isl tag, unless you use subinterfaces on the router. I'd trusting on DSCP in that case. Did I get this one right, now? ;) T -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Charles Sent: Wednesday, July 23, 2008 2:58 PM To: cisco voip; OSL CCIE Voice Lab Exam Subject: [cisco-voip] Trusting CoS on router uplink from switch... OK, so we set the switch ports to trust CoS on the dot1q trunk from phone to switch... are these then trusted from that point on, or do we need to trust them elsewhere as well? Do we need to add any config to the voice gateways to trust the CoS coming from the switches? Jonathan ___ cisco-voip mailing list [EMAIL PROTECTED] https://puck.nether.net/mailman/listinfo/cisco-voip ___ This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unless specifically indicated, this e-mail is not an offer to buy or sell or a solicitation to buy or sell any securities, investment products or other financial product or service, an official confirmation of any transaction, or an official statement of Barclays. Any views or opinions presented are solely those of the author and do not necessarily represent those of Barclays. This e-mail is subject to terms available at the following link: www.barcap.com/emaildisclaimer. By messaging with Barclays you consent to the foregoing. Barclays Capital is the investment banking division of Barclays Bank PLC, a company registered in England (number 1026167) with its registered office at 1 Churchill Place, London, E14 5HP. This email may relate to or be sent from other members of the Barclays Group. ___ ___ This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unless specifically indicated, this e-mail is not an offer to buy or sell or a solicitation to buy or sell any securities, investment products or other financial product or service, an official confirmation of any transaction, or an official statement of Barclays. Any views or opinions presented are solely those of the author and do not necessarily represent those of Barclays. This e-mail is subject to terms available at the
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 29, Issue 28
Vik has put together some good points on when to use which types of trunks. http://www.ipexpert.com/index.cfm/a/p/h323_SIPtrunks On Sun, Jul 13, 2008 at 7:10 PM, Kumar, Narinder [EMAIL PROTECTED] wrote: Question on trunk. The Sys Admin Guide for CME tells us to use ICT ( GK Controlled), when connecting the 2 clusters e.g UCM and UCME via GK Question in when do we have to use H225 GK controlled trunk ??
Re: [OSL | CCIE_Voice] 0 Conf max sessions
Onur, just curious where this ended up. On Mon, Jul 7, 2008 at 4:36 PM, Jonathan Charles [EMAIL PROTECTED] wrote: Post a show diag, I think your DSPs may be hosed... prob bad firmware but this is a WAG Jonathan On Mon, Jul 7, 2008 at 9:41 AM, Onur Tufekci [EMAIL PROTECTED] wrote: And this is if I do it other way around: router2(config-dspfarm-profile)#dspfarm profile 2 router2(config-dspfarm-profile)#shut Disabling profile will disconnect active TRANSCODING calls, do you want to continue ? [yes/no]y router2(config-dspfarm-profile)# router2(config-dspfarm-profile)# router2(config-dspfarm-profile)#dspfarm profile 1 router2(config-dspfarm-profile)#no shut router2(config-dspfarm-profile)#dspfarm profile 2 router2(config-dspfarm-profile)#no shut Enabling profile failed due to insufficient TRANSCODING resources, resources available to support 0 sessions; please modify the configuration and retry router2(config-dspfarm-profile)# On Mon, Jul 7, 2008 at 10:39 AM, Onur Tufekci [EMAIL PROTECTED] wrote: Here is the configuration and results: router2(config)#do show run Building configuration... Current configuration : 2938 bytes ! ! Last configuration change at 15:11:09 UTC Mon Jul 7 2008 ! NVRAM config last updated at 14:49:04 UTC Mon Jul 7 2008 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec service password-encryption ! hostname router2 ! boot-start-marker boot-end-marker ! card type t1 0 0 no logging console enable secret 5 $1$7cJ4$U9gWuPCv.H1DPdXYIyWJD0 ! no aaa new-model ! resource policy ! network-clock-participate wic 0 ! ! ip cef ! ! no ip domain lookup ! isdn switch-type primary-ni ! voice-card 0 dspfarm dsp services dspfarm ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! controller T1 0/0/0 framing esf linecode b8zs ! controller T1 0/0/1 framing esf linecode b8zs ! ! ! ! interface FastEthernet0/0 ip address 192.168.2.1 255.255.255.0 duplex full speed 100 h323-gateway voip bind srcaddr 192.168.2.1 ! interface FastEthernet0/1 description Management_IP ip address 192.168.10.2 255.255.255.0 duplex full speed 100 ! interface Service-Engine1/0 no ip address shutdown ! ip route 192.168.1.0 255.255.255.0 192.168.10.1 ip route 192.168.3.0 255.255.255.0 192.168.10.3 ! ip http server no ip http secure-server ! ! ! ! control-plane ! ! ! ! ! sccp local FastEthernet0/0 sccp ccm 192.168.2.1 identifier 1 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 2 register xcoder associate profile 1 register conference ! dspfarm profile 2 transcode codec g711ulaw codec g729r8 maximum sessions 4 associate application SCCP shutdown ! dspfarm profile 1 conference codec g711ulaw codec g729r8 maximum sessions 1 associate application SCCP shutdown ! ! dial-peer voice 1000 voip answer-address 1... destination-pattern 1... session target ipv4:192.168.1.1 codec g711ulaw ! dial-peer voice 9 pots destination-pattern 9T incoming called-number 14345552... direct-inward-dial ! ! ! telephony-service max-ephones 2 max-dn 2 ip source-address 192.168.2.1 port 2000 auto assign 1 to 2 system message Your current options sdspfarm units 5 sdspfarm tag 1 xcoder sdspfarm tag 2 conference dialplan-pattern 1 14345552... extension-length 4 max-conferences 8 gain -6 call-forward pattern .T web admin system name cisco password cisco dn-webedit time-webedit transfer-system full-consult transfer-pattern .T secondary-dialtone 9 ! ! ephone-dn 1 dual-line number 2001 description 14345552001 name Gil Grissom ! ! router2(config)#dspfarm router2(config)# router2(config)# router2(config)# router2(config)#dspfarm pro 2 router2(config-dspfarm-profile)#no shut router2(config-dspfarm-profile)#dspfarm pro 1 router2(config-dspfarm-profile)#no shut Enabling profile failed due to insufficient CONFERENCING resources, resources available to support 0 sessions; please modify the configuration and retry router2(config-dspfarm-profile)#do show inv NAME: 2811 chassis, DESCR: 2811 chassis PID: CISCO2811 , VID: V03 , SN: FHK103671CU NAME: VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1 on Slot 0 SubSlot 0, DESCR: VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1 PID: VWIC2-2MFT-T1/E1 , VID: V01 , SN: FOC102540TY NAME: PVDMII DSP SIMM with one DSP on Slot 0 SubSlot 4, DESCR: PVDMII DSP SIMM with one DSP PID: PVDM2-16 , VID: V01 , SN: FOC1032054R NAME: NM-SE on Slot 1, DESCR: NM-SE PID: NM-CUE, VID: V03, SN: FOC10120B2E NAME: 40GB IDE Disc Daughter Card on Slot 1 SubSlot 0, DESCR: 40GB IDE Disc
Re: [OSL | CCIE_Voice] unable to configure E1 controller
the router I'm using is for both data as well as PSTN. The channel groups are obviously a non issue, so ports 0/3/1, 0/3/0, and 0/2/1 are non issues. 0/2/0 is currently slated as my 6608 trunk (4 channels total) 0/1/1 is currently slated as my BR1 PRI (full PRI) 0/1/0 is currently slated as a 3640 that I'm using for HQ (don't have a 6608 blade yet, so I'm doing MGCP to this 3640 for a call path now...4 channels just like the 6608). 0/0/0 is my E1 to BR2. I've even tried wiping the config and configuring the E1 trunk first and it didn't matter. according to docs, the 2801 will allow wither a VIC or VWIC as long as the VWIC is used in voice mode... the router has a PVDM2-48 in it, so the 36 channels that I'm trying to use is WELL in range. On Thu, Jul 3, 2008 at 8:10 AM, Mark Snow [EMAIL PROTECTED] wrote: Not sure if you meant to, or even how many DSP resources you have in your router, but one of your T1 voice pri 0/1/1 has 24 (well really 23) timeslots assigned to it. Mark Snow Sr Technical Instructor IPexpert, Inc. Sent from my iPhone On Jul 3, 2008, at 12:01 AM, Michael Thompson [EMAIL PROTECTED] wrote: Well folks, I'm ashamed to even ask this, but it's late and I'm going to bed. I was hoping that in the wee overnight hours someone might have an epiphany. I'm configuring the BR2 E1 interface on the PSTN router. For some reason, I'm getting an error when I do: PSTN-WAN(config-controller)#pri-gr time 1-3,16 %Insufficient resources to create pri-group - it has been removed generally this is one of 2 issues, the lack of network-clock-participate wic X in the configuration or a lack of DSPs. I have 48 channels on this router and this was the first trunk I tried to configure, so it's not a DSP issue. I also confirmed the clock-participate configuration. I've tried 12.4.15 XK, 12.4.19b, as well as 12.4.15T5 (where I am now). Here is the pertinent config. any thoughts on what I overlooked? boot-start-marker boot system flash flash:c2801-ipvoice_ivs-mz.124-15.T5.bin boot-end-marker ! card type e1 0 0 logging buffered 16384 ! clock timezone EST -5 clock summer-time EDT recurring network-clock-participate wic 0 network-clock-participate wic 1 network-clock-participate wic 2 network-clock-participate wic 3 ip cef ! ! ! no ip domain lookup multilink bundle-name authenticated ! frame-relay switching isdn switch-type primary-ni isdn logging ! voice-card 0 dsp services dspfarm ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! controller E1 0/0/0 ! controller T1 0/1/0 description ** Connected to HQ 3640 Controller 0/0/0 for T1 Voice ** framing esf clock source internal linecode b8zs pri-group timeslots 1-3,24 ! controller T1 0/1/1 description ** Connected to BR1-Router Controller T1 0/0/0 for T1-PRI Voice ** framing esf clock source internal linecode b8zs pri-group timeslots 1-24 ! controller T1 0/2/0 description ** Connected to HQ-6500 (6608 port 1) for T1 Voice ** framing esf clock source internal linecode b8zs pri-group timeslots 1-3,24 ! controller T1 0/2/1 description ** Connected to HQ-Router Controller T1 0/3/X for T1 Data ** framing esf clock source internal linecode b8zs channel-group 0 timeslots 1-24 ! controller T1 0/3/0 description ** Connected to BR2-Router Controller T1 0/0/0 for T1 Data ** framing esf clock source internal linecode b8zs channel-group 0 timeslots 1-24 ! controller T1 0/3/1 description ** Connected to BR1-Router Controller T1 0/0 for T1 Data ** framing esf clock source internal linecode b8zs channel-group 0 timeslots 1-24 !
[OSL | CCIE_Voice] unable to configure E1 controller
Well folks, I'm ashamed to even ask this, but it's late and I'm going to bed. I was hoping that in the wee overnight hours someone might have an epiphany. I'm configuring the BR2 E1 interface on the PSTN router. For some reason, I'm getting an error when I do: PSTN-WAN(config-controller)#pri-gr time 1-3,16 %Insufficient resources to create pri-group - it has been removed generally this is one of 2 issues, the lack of network-clock-participate wic X in the configuration or a lack of DSPs. I have 48 channels on this router and this was the first trunk I tried to configure, so it's not a DSP issue. I also confirmed the clock-participate configuration. I've tried 12.4.15 XK, 12.4.19b, as well as 12.4.15T5 (where I am now). Here is the pertinent config. any thoughts on what I overlooked? boot-start-marker boot system flash flash:c2801-ipvoice_ivs-mz.124-15.T5.bin boot-end-marker ! card type e1 0 0 logging buffered 16384 ! clock timezone EST -5 clock summer-time EDT recurring network-clock-participate wic 0 network-clock-participate wic 1 network-clock-participate wic 2 network-clock-participate wic 3 ip cef ! ! ! no ip domain lookup multilink bundle-name authenticated ! frame-relay switching isdn switch-type primary-ni isdn logging ! voice-card 0 dsp services dspfarm ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! controller E1 0/0/0 ! controller T1 0/1/0 description ** Connected to HQ 3640 Controller 0/0/0 for T1 Voice ** framing esf clock source internal linecode b8zs pri-group timeslots 1-3,24 ! controller T1 0/1/1 description ** Connected to BR1-Router Controller T1 0/0/0 for T1-PRI Voice ** framing esf clock source internal linecode b8zs pri-group timeslots 1-24 ! controller T1 0/2/0 description ** Connected to HQ-6500 (6608 port 1) for T1 Voice ** framing esf clock source internal linecode b8zs pri-group timeslots 1-3,24 ! controller T1 0/2/1 description ** Connected to HQ-Router Controller T1 0/3/X for T1 Data ** framing esf clock source internal linecode b8zs channel-group 0 timeslots 1-24 ! controller T1 0/3/0 description ** Connected to BR2-Router Controller T1 0/0/0 for T1 Data ** framing esf clock source internal linecode b8zs channel-group 0 timeslots 1-24 ! controller T1 0/3/1 description ** Connected to BR1-Router Controller T1 0/0 for T1 Data ** framing esf clock source internal linecode b8zs channel-group 0 timeslots 1-24 !
Re: [OSL | CCIE_Voice] SRST voicemail
can you show all the dial peers? On Wed, Jun 25, 2008 at 6:32 PM, Gregory Jost (grjost) [EMAIL PROTECTED] wrote: I'm getting secondary dial-tone on forwards to voicemail in SRST mode from PSTN. I'm able to call internally and it routes to the appropriate mailbox in SRST mode, but calls from PSTN are routing back to the SRST router and giving secondary dial-tone. Any ideas? I'm sure it's something with my PSTN-WAN router. The weird thing is that that direct calls work, but if there's an RDNIS sent it loops back to the originating gateway. call-manager-fallback transfer-pattern .T voicemail 912122211600 call-forward pattern .T call-forward busy 91212221101. call-forward noan 91212221101. timeout 3 Greg Jost Network Consulting Engineer Unified Communications Practice Cisco Systems, Inc. 214-274-1922
Re: [OSL | CCIE_Voice] PLAR
From a design point, I'm NEVER a fan of leaving a partition empty. It seems the equivalent of letting a router use Dial Peer 0. I think you'll be better off assigning it a partition that is exclusively accessible by the calling phone. anyone have thoughts on this? On Thu, Jun 12, 2008 at 10:29 AM, Martin, William J (Bill) [EMAIL PROTECTED] wrote: Route pattern to 4 digit extension (5xxx) with partition of NONE seems to work. William J. Martin *veri**z**on**business* http://verizonbusiness.com/ IP Telephony Solution Architect CCVP, CCDP, CCNP, MCSE 518-426-2120 Office 518-588-2947 Cell 323-2120 VNET [EMAIL PROTECTED] -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *dschulz *Sent:* Thursday, June 12, 2008 10:09 AM *To:* 'OSL CCIE Voice Lab Exam' *Subject:* Re: [OSL | CCIE_Voice] PLAR You might think about using a translation pattern, leaving the pattern empty and have it's partition as the only partition in Phone A's CSS. HTH. Dave -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Martin, William J (Bill) *Sent:* Thursday, June 12, 2008 9:35 AM *To:* OSL CCIE Voice Lab Exam *Subject:* [OSL | CCIE_Voice] PLAR Is there a way to dial a phone that is configured to Auto Ring to another phone? For instance, Phone A auto rings phone B. What CSS/partition can be used to dial Phone A ( which is in the PLAR partition). William J. Martin *veri**z**on**business* http://verizonbusiness.com/ IP Telephony Solution Architect CCVP, CCDP, CCNP, MCSE 518-426-2120 Office 518-588-2947 Cell 323-2120 VNET [EMAIL PROTECTED]
Re: [OSL | CCIE_Voice] Happy Birthday Wayne
I like the way you guys think!!! On Mon, Jun 2, 2008 at 9:38 PM, WorkerBee [EMAIL PROTECTED] wrote: OOoooh..probably Wayne is giving out special discount to this community on IPexpert voice products. BIG 40 - Celebrate with 40% discount :D Dream on~ On Tue, Jun 3, 2008 at 3:05 AM, David L. Blair [EMAIL PROTECTED] wrote: I hear through the grapevine Wayne is getting a year older on 06-08-08. Could it be the BIG 40? David