Re: [OSL | CCIE_Voice] What do I 'really need' - home lab
Josh, I was in the same boat when I began my studies. Ultimately, my lab existed in about 3 phases: Phase 1: 3x 1720s 1x 2600xm 1x 2600 1x 3640 1x 3500xl 4x 7940 This allowed me to do the basic UCM configuration, but DSP configuration was based on the older config style, and CME/CUE support was limited to older versions. I quickly found myself frustrated with the limitations, particularly when trying to work through the IPExpert labs. Phase 2: I was able to borrow a couple of 2800s (one with CUE and switch HWIC), and re-arrange the lab setup. This allowed me to replicate more of the lab materials, although I still had to be cognizant of HW/SW limitations while testing and troubleshooting. I used this setup for the majority of my studies. One of the main limitations of this setup was the flexibility in practicing different lab scenarios. Any revision of the PSTN router configuration would end up being a mini-lab by itself, which, while valuable, took away from time dedicated to other topics. Phase 3: I went completely online with proctor labs. I went virtual for about the last 2 weeks of my test prep. The biggest advantage for me was not wasting time updating configurations and focusing on the tasks. I think that it's most important to have similar hardware in this order: PSTN BR2 HQ BR1 Having said that, my PSTN/WAN config was actually spread across 2x 1720s and the 3640, however, so if you're willing to pick and choose the labs you work on, you can drop the PSTN router to the tail end of that list. Hope that helps! mike On Tue, Nov 23, 2010 at 11:26 PM, Josh Kittle j...@ooglenetworks.comwrote: Ok guys so I'm pretty new to the CCIE Voice Lab Prep experience - starting to build the home lab. I've read tons of stuff along the line of 'you need 3x 2811's, etc and I've come to the group to ask - what do i REALLY need. Not so much cards, modules etc (although that feedback is much welcome and appreciated) - but from a router perspective. Does HQ, and the BR sites each need to really be 2811's to run the scenarios? Why not a 3725, etc? Is it hardware module support (lack thereof) - if so, for which modules give the grief as far as support goes? I know I can't skimp on everything - but certainly there have to be some compromises that could be made. If there aren't - tell me why - that answer is just as valuable in the learning process. I'm just starting to browse the lab guides, so forgive me if this is covered elsewhere (and if so, kindly reference the location). Thanks guys! Josh ___ 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] Thank you
Way to go, Mark! I'm not sure that I would have been able to handle testing on Friday and not getting results for four days! :) mike On Tue, Nov 2, 2010 at 6:17 PM, Mark Holloway m...@markholloway.com wrote: I want to say thank you to everyone on the OSL who has participated in any of my discussions or helped resolve issues that I encountered. I went to San Jose for my second attempt on Friday and received the news yesterday that I passed. CCIE #27384. Thanks, Mark ___ 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] SRST Display ID 4 digits ANI
I'd guess that your phones are configured with E.164 numbers as the external number mask. Remember that in SRST mode, the External Number Mask gets configured as the calling name for the SRST ephone, and you'll see this behavior. mike On Thu, Oct 21, 2010 at 9:18 AM, Afzal Bhutta azhar.bhu...@gmail.comwrote: Hello Folks, I have question in SRST mode.This is regarding Site-B which has H.323 gateway configured. In SRST Site-B should be able to call HQ Phones.The display ID should be 4 digits ANI as caller ID. Testing: When *I cal from 3001 to 2003 on phone-2003, I got below display, From +19723033001 (3001) ISDN Q931 shows below out put Display i = '+19723033001' Calling Party Number i = 0x0080, '3001' Plan:Unknown, Type:Unknown Called Party Number i = 0xA1, '14082022003' Plan:ISDN, Type:National .Oct 17 23:17:50.702: ISDN Se0/0/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x808D Channel ID i = 0xA98383 Exclusive, Channel 3 Question:How can i remove From field **From +19723033001* on HQ-phone-2003 display.The question requirement is 4 digits ANI as caller ID. Thanks * * ___ 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] Passed! And I have 50 extra lab vouchers for sale
Congrats Jim! On Wed, Oct 20, 2010 at 11:15 AM, Jim Engel jim.en...@ironbow.com wrote: So happy to have passed! J Anyone interested in purchasing any of my remaining vouchers please email me direct. jim.en...@ironbow.com Jim CCIE #6269 (Voice/RS) ___ 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] Order for arranging resources in MRGL
Sorry, let me try to explain with an example: You have an MRGL with SW CFB (on UCM), MOH, SW MTP (on Router), and XCODE resources available. If UCM needs to create a conference, it will select the SW CFB. Ordering does not matter, because there aren't any other CFB resources listed. However, if UCM needs to allocate an MTP, keep in mind that XCODE resources can be used as MTP, so the ordering of the MRG within the MRGL matters for these two resources. Typically, you'd want to choose the SW MTP over the XCODER (HW MTP), due to cost, so you would want to list the MRG with the SW MTP above the MRG with the XCODER. The SW CFB and MOH could be in either MRG, or in a completely separate MRG. UCM can use all of the resources within an MRG at the same time for various purposes, however similar resources within a single MRG will be round-robin load balanced. One or two of the IPExpert labs touches on this in reference to MOH resources. mike On Tue, Oct 5, 2010 at 9:32 PM, CCIE Voice GMAIL givemeccievoice2...@gmail.com wrote: Can you please elaborate on this statement “if you only have a single CFB, MOH, and MTP, you can list them in the same MRG because UCM won't can't use a CFB if you need to insert an MTP.” Are you saying that when in a MRG, having both a CFB and an MTP would result in only one resource being usable at a time? Or are you saying that having a CFB and MTP in the same MRGL will result in one resource being usable at a time? I’m just confused at the wording. Thanks for your help, Jeff *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *groganhockey *Sent:* Tuesday, October 05, 2010 7:14 PM *To:* Pithog Oil *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] Order for arranging resources in MRGL Keep in mind these guidelines when configuring MRGL: Resources are searched in the order listed in the MRGL, so if you insert three MRGs into the MRGL, UCM will search the first MRG for an appropriate resource, then search the second MRG, then search the third. Resources within an MRG are selected in a round-robin fashion. Ordering within the MRGL is only applicable to like-type of resources, so if you only have a single CFB, MOH, and MTP, you can list them in the same MRG because UCM won't can't use a CFB if you need to insert an MTP. mike On Tue, Oct 5, 2010 at 1:51 PM, Pithog Oil pithog...@yahoo.com wrote: Hi experts I know there is an Order for arranging arranging resources in the MRGL in a scenario where i have multiple resoucres in a site , but i need to figure out where Conference bridge fits in , in the order. Please correct me if wrong MOH first Transcoder second MTP third Also i while like to know if its possible to have a resources Glut. ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Order for arranging resources in MRGL
Keep in mind these guidelines when configuring MRGL: Resources are searched in the order listed in the MRGL, so if you insert three MRGs into the MRGL, UCM will search the first MRG for an appropriate resource, then search the second MRG, then search the third. Resources within an MRG are selected in a round-robin fashion. Ordering within the MRGL is only applicable to like-type of resources, so if you only have a single CFB, MOH, and MTP, you can list them in the same MRG because UCM won't can't use a CFB if you need to insert an MTP. mike On Tue, Oct 5, 2010 at 1:51 PM, Pithog Oil pithog...@yahoo.com wrote: Hi experts I know there is an Order for arranging arranging resources in the MRGL in a scenario where i have multiple resoucres in a site , but i need to figure out where Conference bridge fits in , in the order. Please correct me if wrong MOH first Transcoder second MTP third Also i while like to know if its possible to have a resources Glut. ___ 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] Single Number Reach
If I'm following your example correctly, Mark, then you aren't hitting on the translation pattern. The SNR call is matching the \+1408.6347694 RP, to go out, why would it be hitting the translation pattern? Perhaps you meant to configure this as a Calling Party Transformation? mike On Fri, Oct 1, 2010 at 2:38 AM, Mark Holloway m...@markholloway.com wrote: I'm having a hard time when an internal extension calls another internal extension that uses SNR, the From phone number on the PSTN phone is 4 digits instead of 7. For example, extension 2001 calls 2003, and 2003 simultaneously rings a PSTN phone number. The display on the PSTN phone says HqPh1 (2001) instead of the 7 digit or 10 digit number. I have created PT_SNR which is assigned to CSS_SNR. I have CSS_SNR assigned to the Remote Destination Profile for both CSS and Redirecting CSS. My SNR number is +14086347694 and I have a route pattern that contains \+1408.6347694 which egresses the RL_HQ_ONLY (this is not Standard Local Route Group). I also created a Translation Pattern with PT_SNR and I have checked Use External Phone Number Mask. I was expecting this to take the 4 digit Calling number and insert the External mask instead. I tried following the steps in the Mock Lab guide (I believe it is Lab 6) but I still cannot get it working. Any assistance would be appreciated. Perhaps someone has a blog post? Thanks, Mark ___ 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] Pass
Congrats Jeff! On Tue, Sep 21, 2010 at 4:50 PM, Jeff Cotter jcot...@voxns.com wrote: Finally….took more times than I care to admit! A big thanks to IPexpert (especially Vic) and everybody who has been a part of this list. Jeff Cotter CCIE #27033 ___ 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've got a new favorite number!
Thanks, everybody. I'd just like to re-iterate my thanks to the IPExpert folks and the study materials that they have put together. They really are top notch. Vik and Amy's video walkthroughs in conjuction with the written Proctor Guides are extremely thorough and clearly do a great job to prepare you for the lab. Just look at all the people who have passed in the last few months! Thanks! mike On Thu, Sep 16, 2010 at 2:50 PM, Ashar Siddiqui siddas...@gmail.com wrote: Well done! Congrats.. Ash CCIE#26244 (Voice) groganhockey wrote: 26966! I'm not sure how often it'll come up in everyday life, but there it is. I took the lab Monday in RTP and finally got my score report last night. IPExpert/Vik/Amy: Thank you for the excellent study guides, walkthroughs, audio and everything! Mike CCIE #26966 Voice -- ___ 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
[OSL | CCIE_Voice] I've got a new favorite number!
26966! I'm not sure how often it'll come up in everyday life, but there it is. I took the lab Monday in RTP and finally got my score report last night. IPExpert/Vik/Amy: Thank you for the excellent study guides, walkthroughs, audio and everything! Mike CCIE #26966 Voice ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] I've got a new favorite number!
In our case, they actually brought lunch in and we ate in an adjoining conference room rather than head to the cafeteria. At the time, I didn't think to ask if it was out of character for RTP or not? mike On Wed, Sep 15, 2010 at 11:57 AM, Amy Ryan ar...@ipexpert.com wrote: Bill, Lunch is usually between 30-45 minutes. During this time you will likely be escorted by the proctor to the cafeteria and will remain as a group. I am not sure taking a walk on your own will be an option for you. HTH, Amy --- Amy Ryan – CCIE #24677 (Voice) Technical Instructor - IPexpert, Inc. Mailto: *ar...@ipexpert.com *Telephone: +1.810.326.1444 Live Assistance, Please visit: www.ipexpert.com/chat * http://www.ipexpert.com/chat* eFax: +1.810.454.0130 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities *http://www.ipexpert.com/communities* and our public website at www.ipexpert.com *http://www.ipexpert.com/* -- *From: *Bill Lake whl...@gmail.com *Date: *Wed, 15 Sep 2010 10:45:36 -0500 *To: *ccie_voice@onlinestudylist.com *Subject: *[OSL | CCIE_Voice] I've got a new favorite number! Congratulations on your passing the exam. Not sure if it is a question you can answer (Don't break the Cisco NDA) but how long did you get for lunch? I am planning on my exam and since it is a long day, I am wondering if you had time to walk during lunch to help relieve the stress and boost your brain. -- ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SCCP to SIP conversion on CME
You need a LOAD statement under voice register global. mike On Tue, Sep 14, 2010 at 4:43 AM, linuxboss.9 linuxbos...@gmail.com wrote: I used below configuration to register 7961GE as SIP to CME but it is showing as SCCP registered. I have all the SIP firmware in root directory of flash. It should start downloading the SIP firmware but there is no action..there are no debug messages because the phone is already SCCP registered. Did switch port shut/no shut ..no change. Can anyone guide me where I am wrong. voice service voip allow-connections sip to sip fax protocol cisco sip bind control source-interface GigabitEthernet0/0.11 bind media source-interface GigabitEthernet0/0.11 registrar server expires max 1200 min 300 voice register global mode cme source-address 10.21.200.1 port 5060 max-dn 10 max-pool 5 load 7961GE SIP41.8-5-4S authenticate register tftp-path flash: create profile sync 0005355132715547 voice register dn 1 number name Br2Ph2 label Br2 voice register pool 1 id mac 0AAA.F999.D562 type 7961GE number 1 dn 1 dtmf-relay sip-notify username br2ph2 password cisco codec g711ulaw R3#confi R3#configure t R3#configure terminal Enter configuration commands, one per line. End with CNTL/Z. R3(config)#voic R3(config)#voice re R3(config)#voice register poo R3(config)#voice register pool 1 R3(config-register-pool)#restart *No contact info available for pool 1.* ___ 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] SCCP to SIP conversion on CME
Awesome..can't quite see straight today. Glad to hear that you got it working. I do believe I saw (in the IPExpert blog after the CCIE Voice techtorial?) that Ben Ng stated that the phones would be setup with the correct firmware, so conversion shouldn't be an issue. Always good to know the process, though. mike On Tue, Sep 14, 2010 at 2:11 PM, linuxboss.9 linuxbos...@gmail.com wrote: Amy..Brian..Mike et all, Thanks to all for your responses. no create profile create profile commands under voice register global resolved the issue. I provided wrong tftp-server parameters before. Instead of cleaning it I tried to troubleshoot it in the existing state which took around 5 hours. Nevertheless a good exercise. Now I am SCCP-SIP conversion specialist :-) Have a wonderful day ! -Shrini On Tue, Sep 14, 2010 at 11:41 AM, Brian Valentine bkvalent...@gmail.comwrote: Gig0/0.11 is 10.21.200.1? You might want to make that the source address for your tftp server. Looks like the LOAD statement is already there, but you need to serve the files via tftp. you also need an ntp server command under voice register global. Make sure the firmware files are served up using the tftp-server global config commands. I will typically look in the root of the flash drive for the SEPMAC.cnf or .cnf.xml files... delete any you find. Make sure you leave the defaults there.. just delete the ones with specific MAC addresses. Then under voice register global, offer the no create prof and then issue the create prof commands. See if you have any more SEPMAC.cnf or .cnf.xml files. If not, something is wrong in your config. You can debug tftp events while the phone reboots to watch and see what it is downloading for your tftp server. If you aren't getting anything when the phone boots, you might not have it pointed at the right IP address in your dhcp scope. Any time you change anything at the DN, Pool, or global levels, you should go to the voice register global and issue the same commands no create prof and then create prof before you restart your pools. SCCP phones don't require the create cnf-files every time, but sip phones do require the create profile to be issued with every change. Hope some of that helps. As an aside, you should also replace the dtmf-relay sip-notify command with dtmf-relay rtp-nte in the voice register pool. I don't think this is your problem with the phones registering as SCCP, but it will help save your hours more troubleshooting later. Brian On Tue, Sep 14, 2010 at 2:23 PM, groganhockey groganhoc...@gmail.com wrote: You need a LOAD statement under voice register global. mike On Tue, Sep 14, 2010 at 4:43 AM, linuxboss.9 linuxbos...@gmail.com wrote: I used below configuration to register 7961GE as SIP to CME but it is showing as SCCP registered. I have all the SIP firmware in root directory of flash. It should start downloading the SIP firmware but there is no action..there are no debug messages because the phone is already SCCP registered. Did switch port shut/no shut ..no change. Can anyone guide me where I am wrong. voice service voip allow-connections sip to sip fax protocol cisco sip bind control source-interface GigabitEthernet0/0.11 bind media source-interface GigabitEthernet0/0.11 registrar server expires max 1200 min 300 voice register global mode cme source-address 10.21.200.1 port 5060 max-dn 10 max-pool 5 load 7961GE SIP41.8-5-4S authenticate register tftp-path flash: create profile sync 0005355132715547 voice register dn 1 number name Br2Ph2 label Br2 voice register pool 1 id mac 0AAA.F999.D562 type 7961GE number 1 dn 1 dtmf-relay sip-notify username br2ph2 password cisco codec g711ulaw R3#confi R3#configure t R3#configure terminal Enter configuration commands, one per line. End with CNTL/Z. R3(config)#voic R3(config)#voice re R3(config)#voice register poo R3(config)#voice register pool 1 R3(config-register-pool)#restart No contact info available for pool 1. ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Proctorlabs Down?
or is it just me? mike ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] extension Mobility
Yes, you can. They will show up and act as a shared line. Is the system prompting you for a PIN, or what specifically? mike On Wed, Sep 8, 2010 at 5:32 PM, Leslie Meade lme...@signal.ca wrote: Can you have the same number both as an extension mobility number as well as a normal number on a phone ? I get the user logged into mobility but when they try to log out I get please enter the admin password, instead of a logout button . Cheers Leslie ___ 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] Cant get to PSTN router / BR1 / BR2
Nope, just wrapped up a session with no disconnects. mike On Mon, Sep 6, 2010 at 12:30 PM, chase mergenthal cm3_...@hotmail.comwrote: I'm not sure what happened; but my connection to the PSTN router / BR1 / BR2 dropped in my lab session and i cant reconnect; anyone else having problems? -Chase ___ 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] Call-Manger-Fallback
I think this is the Connection Monitor Duration paramter under UCM Enterprise Parameters. mike On Thu, Sep 2, 2010 at 1:02 PM, Damon Chaput dcha...@1800contacts.comwrote: Hi, I am working on Vol 2 Lab 3 section 3.1. When I take the WAN down, fallback works as expected and occurs immediately. When I bring the WAN back up MGCP kicks back in very quickly and the gateway registers right away. For the phones, it takes over a minute and a half after MGCP backhaul is complete. Just curious if there are any timers to adjust so the phones will recognize that the WAN is back up and register back to CUCM quicker. I tried adjusting the “ Keepalive” seconds to 10 under call-manager-fallback but it didn’t seem to make a difference. Sep 2 17:39:46.958: ISDN Se0/0/0:15 Q931: L3_ShutDown: Shutting down ISDN Layer 3 Sep 2 17:39:46.958: %ISDN-6-LAYER2DOWN: Layer 2 for Interface Se0/0/0:15, TEI 0 changed to down Sep 2 17:39:46.962: ISDN Se0/0/0:15 Q931: Ux_DLRelInd: DL_REL_IND received from L2 Sep 2 17:39:48.950: %LINK-5-CHANGED: Interface Serial0/0/0:15, changed state to administratively down Sep 2 17:39:53.370: %LINK-3-UPDOWN: Interface Serial0/0/0:15, changed state to up Sep 2 17:39:53.370: %LINK-3-UPDOWN: Interface Serial0/0/0:0, changed state to down Sep 2 17:39:53.370: %LINK-3-UPDOWN: Interface Serial0/0/0:1, changed state to down Sep 2 17:39:53.370: %LINK-3-UPDOWN: Interface Serial0/0/0:2, changed state to down Sep 2 17:39:55.871: %ISDN-6-LAYER2UP: Layer 2 for Interface Se0/0/0:15, TEI 0 changed to up Sep 2 17:41:35.240: %IPPHONE-6-UNREGISTER_NORMAL: ephone-1:SEP0014A9816FDB IP:10.10.202.50 Socket:2 DeviceType:Phone has unregistered normally. Sep 2 17:41:35.272: %IPPHONE-6-UNREGISTER_NORMAL: ephone-2:SEP0014A98BBB7F IP:10.10.202.57 Socket:3 DeviceType:Phone has unregistered normally. Sep 2 17:41:35.464: %IPPHONE-6-UNREGISTER_NORMAL: ephone-3:SEP0014A9739190 IP:10.10.202.58 Socket:4 DeviceType:Phone has unregistered normally. Sep 2 17:41:45.492: %IPPHONE-6-UNREGISTER_NORMAL: ephone-4:SEP002290BAA05B IP:10.10.202.56 Socket:1 DeviceType:Phone has unregistered normally. Thanks, Damon Chaput ___ 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] Docs available during the lab
According to the most recent Ask the Expert with Ben Ng, there are four documents on the desktop: We have four SRND documents ready to be opened, also you have the online Cisco document page. 1. UC 7 SRND 2. CUCME 7 SRND 3. UCCX 7 SRND 4. Enterprise QoS SRND 3.3 On Tue, Aug 31, 2010 at 1:33 PM, Carhart, David dcarh...@lvbrands.comwrote: Does anyone no where I can get a list of the docs that you are provided for the voice lab? Thanks David Carhart dcarh...@lvbrands.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] Docs available during the lab
Amy, I wasn't sure if Ben meant the CME Admin Guide (which is more helpful than the SRND, I think) in the Ask the Expert forum. Does anybody know for sure? mike On Tue, Aug 31, 2010 at 2:21 PM, Amy Ryan ar...@ipexpert.com wrote: David, Based on the voice techtorial offered at Cisco Live this year, below is what was identified. -Unity Connection Administration Guide -QOS SRND -CUCME Administration Guide -CUCM SRND -UCCX SRND And you will have access to the cisco product/technology support page. HTH, Amy --- Amy Ryan CCIE #24677 (Voice) Technical Instructor - IPexpert, Inc. Mailto: ar...@ipexpert.com Telephone: +1.810.326.1444 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat eFax: +1.810.454.0130 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ From: Carhart, David dcarh...@lvbrands.com Date: Tue, 31 Aug 2010 14:33:40 -0400 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Docs available during the lab Does anyone no where I can get a list of the docs that you are provided for the voice lab? Thanks David Carhart dcarh...@lvbrands.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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Docs available during the lab
And the URL for the Product / Technology support page is: http://www.cisco.com/cisco/web/psa/default.html?mode=prod On Tue, Aug 31, 2010 at 2:54 PM, groganhockey groganhoc...@gmail.comwrote: Amy, I wasn't sure if Ben meant the CME Admin Guide (which is more helpful than the SRND, I think) in the Ask the Expert forum. Does anybody know for sure? mike On Tue, Aug 31, 2010 at 2:21 PM, Amy Ryan ar...@ipexpert.com wrote: David, Based on the voice techtorial offered at Cisco Live this year, below is what was identified. -Unity Connection Administration Guide -QOS SRND -CUCME Administration Guide -CUCM SRND -UCCX SRND And you will have access to the cisco product/technology support page. HTH, Amy --- Amy Ryan CCIE #24677 (Voice) Technical Instructor - IPexpert, Inc. Mailto: ar...@ipexpert.com Telephone: +1.810.326.1444 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat eFax: +1.810.454.0130 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ From: Carhart, David dcarh...@lvbrands.com Date: Tue, 31 Aug 2010 14:33:40 -0400 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Docs available during the lab Does anyone no where I can get a list of the docs that you are provided for the voice lab? Thanks David Carhart dcarh...@lvbrands.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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Vol2 Lab6 CME to UC SIP Integration
All, Working through a SIP integration between UCME and UC using SIP, and I've got a couple of issues versus how the solutions guide lays out the solution. Item 1.4 states Make sure that calls within the HQ or BR1 site use g711ulaw and calls between sites use g729. (1 pt) But in item 6.2, the solution shows the SIP dial-peer to the UC server with codec g711ulaw. Calls from BR2 to voicemail at HQ are now using g711ulaw, so haven't we just cost ourselves a point? (Better 1 point than the 4 points for the SIP UC integration I guess). I have been trying to work around this issue by various means, but haven't come up with a solution yet. I have hardcoded the UC dial-peer to g729 Because BR2 has SIP phones, I have registered a transcoder to UCME at BR2, and the call to voicemail from the SIP phones invokes the transcoder correctly. The specific issue that I'm facing is if I dial by extension within the Main Greeting, UC transfers the call back to BR2, but the call is rejected. I can get around this specific limitation by configuring voice-class codec on the UC dial-peer, but then the SIP phone uses g711u straight to UC. Here's the debug ccsip messages from the call when UC tries to send the call back to 3001. BR2# BR2#sh sccp conn sess_idconn_id stype mode codec sport rport ripaddr 1 1xcode sendrecv g72917442 2000 10.10.202.1 1 2xcode sendrecv g711u 17630 2000 10.10.202.1 Total number of active session(s) 1, and connection(s) 2 BR2# BR2# BR2# BR2# BR2# BR2# BR2# Aug 31 03:14:11.319: //-1//SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:3...@10.10.110.3:5060 SIP/2.0 From: sip:3...@10.10.210.12 sip%3a3...@10.10.210.12 ;tag=b479d639c3b54f96995687a535357a46 To: BR2 Phone3 sip:3...@10.10.110.3 sip%3a3...@10.10.110.3 ;tag=91A708-25BC Via: SIP/2.0/UDP 10.10.210.12:5060 ;branch=z9hG4bKb9702a2ec4174a48916a263f9434aa8b Max-Forwards: 70 User-Agent: Cisco-UnityConnection/7.0 Contact: sip:10.10.210.12:5060;transport=tcp Call-ID: a01a55e6-b3e411df-81e98f0a-3d3c3...@10.10.110.3 CSeq: 2 INVITE Allow-Events: kpml Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,SUBSCRIBE Content-Length: 256 Content-Type: application/sdp v=0 o=10.10.210.12 3324197134 3324197134 IN IP4 10.10.210.12 s=No Subject c=IN IP4 0.0.0.0 t=0 0 m=audio 16702 RTP/AVP 0 18 101 a=rtpmap:0 pcmu/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Aug 31 03:14:11.331: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Date: Tue, 31 Aug 2010 03:14:11 GMT From: sip:3...@10.10.210.12 sip%3a3...@10.10.210.12 ;tag=b479d639c3b54f96995687a535357a46 Allow-Events: telephone-event Content-Length: 0 To: BR2 Phone3 sip:3...@10.10.110.3 sip%3a3...@10.10.110.3 ;tag=91A708-25BC Call-ID: a01a55e6-b3e411df-81e98f0a-3d3c3...@10.10.110.3 Via: SIP/2.0/UDP 10.10.210.12:5060 ;branch=z9hG4bKb9702a2ec4174a48916a263f9434aa8b CSeq: 2 INVITE Server: Cisco-SIPGateway/IOS-12.x Aug 31 03:14:11.331: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 488 Not Acceptable Media Date: Tue, 31 Aug 2010 03:14:11 GMT From: sip:3...@10.10.210.12 sip%3a3...@10.10.210.12 ;tag=b479d639c3b54f96995687a535357a46 Allow-Events: telephone-event Content-Length: 0 To: BR2 Phone3 sip:3...@10.10.110.3 sip%3a3...@10.10.110.3 ;tag=91A708-25BC Call-ID: a01a55e6-b3e411df-81e98f0a-3d3c3...@10.10.110.3 Via: SIP/2.0/UDP 10.10.210.12:5060 ;branch=z9hG4bKb9702a2ec4174a48916a263f9434aa8b CSeq: 2 INVITE Server: Cisco-SIPGateway/IOS-12.x Aug 31 03:14:11.343: //-1//SIP/Msg/ccsipDisplayMsg: Received: ACK sip:3...@10.10.110.3:5060 SIP/2.0 From: sip:3...@10.10.210.12 sip%3a3...@10.10.210.12 ;tag=b479d639c3b54f96995687a535357a46 To: BR2 Phone3 sip:3...@10.10.110.3 sip%3a3...@10.10.110.3 ;tag=91A708-25BC Via: SIP/2.0/UDP 10.10.210.12:5060 ;branch=z9hG4bKb9702a2ec4174a48916a263f9434aa8b Max-Forwards: 70 User-Agent: Cisco-UnityConnection/7.0 Call-ID: a01a55e6-b3e411df-81e98f0a-3d3c3...@10.10.110.3 CSeq: 2 ACK Content-Length: 0 BR2# Is there a way to make this work? mike ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol2 Lab6 CME to UC SIP Integration
Actually, I found a workaround to get this working correctly. voice service voip sip g729 annexb-all This allows the call to setup and complete successfully between SCCP and SIP endpoints at BR2 (as long as there is a transcoder for the SIP phones). mike On Mon, Aug 30, 2010 at 10:44 PM, groganhockey groganhoc...@gmail.comwrote: All, Working through a SIP integration between UCME and UC using SIP, and I've got a couple of issues versus how the solutions guide lays out the solution. Item 1.4 states Make sure that calls within the HQ or BR1 site use g711ulaw and calls between sites use g729. (1 pt) But in item 6.2, the solution shows the SIP dial-peer to the UC server with codec g711ulaw. Calls from BR2 to voicemail at HQ are now using g711ulaw, so haven't we just cost ourselves a point? (Better 1 point than the 4 points for the SIP UC integration I guess). I have been trying to work around this issue by various means, but haven't come up with a solution yet. I have hardcoded the UC dial-peer to g729 Because BR2 has SIP phones, I have registered a transcoder to UCME at BR2, and the call to voicemail from the SIP phones invokes the transcoder correctly. The specific issue that I'm facing is if I dial by extension within the Main Greeting, UC transfers the call back to BR2, but the call is rejected. I can get around this specific limitation by configuring voice-class codec on the UC dial-peer, but then the SIP phone uses g711u straight to UC. Here's the debug ccsip messages from the call when UC tries to send the call back to 3001. BR2# BR2#sh sccp conn sess_idconn_id stype mode codec sport rport ripaddr 1 1xcode sendrecv g72917442 2000 10.10.202.1 1 2xcode sendrecv g711u 17630 2000 10.10.202.1 Total number of active session(s) 1, and connection(s) 2 BR2# BR2# BR2# BR2# BR2# BR2# BR2# Aug 31 03:14:11.319: //-1//SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:3...@10.10.110.3:5060 SIP/2.0 From: sip:3...@10.10.210.12 sip%3a3...@10.10.210.12 ;tag=b479d639c3b54f96995687a535357a46 To: BR2 Phone3 sip:3...@10.10.110.3 sip%3a3...@10.10.110.3 ;tag=91A708-25BC Via: SIP/2.0/UDP 10.10.210.12:5060 ;branch=z9hG4bKb9702a2ec4174a48916a263f9434aa8b Max-Forwards: 70 User-Agent: Cisco-UnityConnection/7.0 Contact: sip:10.10.210.12:5060;transport=tcp Call-ID: a01a55e6-b3e411df-81e98f0a-3d3c3...@10.10.110.3 CSeq: 2 INVITE Allow-Events: kpml Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,SUBSCRIBE Content-Length: 256 Content-Type: application/sdp v=0 o=10.10.210.12 3324197134 3324197134 IN IP4 10.10.210.12 s=No Subject c=IN IP4 0.0.0.0 t=0 0 m=audio 16702 RTP/AVP 0 18 101 a=rtpmap:0 pcmu/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Aug 31 03:14:11.331: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Date: Tue, 31 Aug 2010 03:14:11 GMT From: sip:3...@10.10.210.12 sip%3a3...@10.10.210.12 ;tag=b479d639c3b54f96995687a535357a46 Allow-Events: telephone-event Content-Length: 0 To: BR2 Phone3 sip:3...@10.10.110.3 sip%3a3...@10.10.110.3 ;tag=91A708-25BC Call-ID: a01a55e6-b3e411df-81e98f0a-3d3c3...@10.10.110.3 Via: SIP/2.0/UDP 10.10.210.12:5060 ;branch=z9hG4bKb9702a2ec4174a48916a263f9434aa8b CSeq: 2 INVITE Server: Cisco-SIPGateway/IOS-12.x Aug 31 03:14:11.331: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 488 Not Acceptable Media Date: Tue, 31 Aug 2010 03:14:11 GMT From: sip:3...@10.10.210.12 sip%3a3...@10.10.210.12 ;tag=b479d639c3b54f96995687a535357a46 Allow-Events: telephone-event Content-Length: 0 To: BR2 Phone3 sip:3...@10.10.110.3 sip%3a3...@10.10.110.3 ;tag=91A708-25BC Call-ID: a01a55e6-b3e411df-81e98f0a-3d3c3...@10.10.110.3 Via: SIP/2.0/UDP 10.10.210.12:5060 ;branch=z9hG4bKb9702a2ec4174a48916a263f9434aa8b CSeq: 2 INVITE Server: Cisco-SIPGateway/IOS-12.x Aug 31 03:14:11.343: //-1//SIP/Msg/ccsipDisplayMsg: Received: ACK sip:3...@10.10.110.3:5060 SIP/2.0 From: sip:3...@10.10.210.12 sip%3a3...@10.10.210.12 ;tag=b479d639c3b54f96995687a535357a46 To: BR2 Phone3 sip:3...@10.10.110.3 sip%3a3...@10.10.110.3 ;tag=91A708-25BC Via: SIP/2.0/UDP 10.10.210.12:5060 ;branch=z9hG4bKb9702a2ec4174a48916a263f9434aa8b Max-Forwards: 70 User-Agent: Cisco-UnityConnection/7.0 Call-ID: a01a55e6-b3e411df-81e98f0a-3d3c3...@10.10.110.3 CSeq: 2 ACK Content-Length: 0 BR2# Is there a way to make this work? mike ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] ProctorLabs IPCC license file location?
Unfortunately, none of my servers was being reloaded when I requested a reload. :( On Sat, Aug 28, 2010 at 8:48 AM, Ohamien Uhakheme oham...@gmail.com wrote: Email support and they will mail you the file, or you can load Vol 2 Lab 1, and it should be on the root of C: Ohamien On Fri, Aug 27, 2010 at 9:01 PM, groganhockey groganhoc...@gmail.comwrote: Am I missing it? I don't see it on the IPCC server? ___ 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] Proctorlabs not resolving
Tyson, I had my first session on PL on Friday night, and this was one of the issues that I ran into. I could not get the servers to reload no matter what I tried (it seemed). I had another session scheduled for this morning, but cancelled it as a result of the frustrating session I had on Friday. Lucky for me, I have a pretty good home lab, so I can do quite a bit of testing and scenarios on my own equipment, at my leisure. My lab is a mix of current and previous generation(s) of equipment, however, so I can't simply copy and paste the IOS configs into my own gear. There are two reasons that I chose to purchase PL rack time: 1. Convenience of loading the various IPExpert scenarios. 2. Access to a full lab of the actual equipment. What are your recommendations in the short term in the event that the servers will not reload? Thanks, mike On Sun, Aug 29, 2010 at 9:34 PM, Tyson Scott tsc...@ipexpert.com wrote: Bill, Unfortunately the servers are the number one issue for voice customers that we can't provide a good solution to in the event these problems happen again. To provide what you are suggesting below we would have to give administrative access to the VMware servers. I will give you the honest answer as to why would never do it. At least 1 or 2 times a month someone deletes the flash on the routers/switches in the pod. This is annoying but manageable as the files are small and we can simply upload them via the USB interface or xmodem them on the switches. recovering a server after someone makes an accidental mistake is not so easily achieved. And it is easy enough to wipe out a datastore with intended or unintended actions. The datastore per server is 1.5 TB. Not as easy/quick of a problem to resolve. Regards, Tyson Scott - CCIE #13513 RS, Security, and SP Managing Partner / Sr. Instructor - IPexpert, Inc. Mailto: tsc...@ipexpert.com Telephone: +1.810.326.1444, ext. 208 Live Assistance, Please visit: www.ipexpert.com/chat eFax: +1.810.454.0130 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities and our public website at www.ipexpert.com -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Bill Lake Sent: Sunday, August 29, 2010 9:15 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Proctorlabs not resolving Hello, I would have to agree that not being able to load the servers especially is an issue. To work the voice labs you must have the servers available and configured properly for the lab. While we could use the downloaded configurations for the routers/switch, we can not do so for the servers. Also, just having the servers on does not allow us to remotely load the server configurations as those backups/configs are not supplied in the Proctor Labs voice section. Perhaps as a backup Proctor Labs can provide instructions to power on the servers remotely without the web page, how to get and load the configurations for the servers and at that point it could be a solutions that we could use to work our voice labs. Bill ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] ProctorLabs IPCC license file location?
Am I missing it? I don't see it on the IPCC server? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Gatekeeper-Controlled ICT MTP
All, I'm trying to understand a behavior I'm seeing in setting up a GK-controlled ICT between UCM clusters. When I place a call across the GK-ICT, the receiving cluster attempts to allocate an MTP for the call. On the trunk: MTP Required is not checked Inbound/Outbound Fast Start are not checked From reviewing the traces, I see the MediaManager and MediaTerminationPointControl services working to allocate an MTP, so it does not appear to be a transcoding issue. It appears that the software MTP allocation fails and the system allocates a HW transcoder. This is confirmed if I set the regions to all G711. The call still invokes a transcoder, but the streams are both reported as g711u. If I take the transcoders offline, the call completes without the transcoder, with the media stream directly between the two phones, albeit there are supplementary services issues. Is this a function of the GK-ICT in general? Or is the UCM trying to add an MTP for additional capabilities? Any thoughts are welcome while I continue to test. mike ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE #26721 - I PASSED!
Congratulations, Matthew. You're a great inspiration and you have made great contributions to the list and via your blog! mike On Tue, Aug 17, 2010 at 5:05 PM, Matthew Berry ciscovoiceg...@gmail.comwrote: I just got my score report. I passed guys. More follow-up to come later. Right now I'm now on cloud nine. :) CCIE #26271 Thanks, Matthew Berry ciscovoiceg...@gmail.com http://ciscovoiceguru.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] Choosing the right ISR?
I'm just glad I can finally contribute *something* to these discussions! :) FYI, cisco has moved the doc in the past, so make sure you remember the title in case it moves again. mike On Thu, Sep 24, 2009 at 9:38 AM, shikamaru shikam...@kagadis.com wrote: MUCH respect, Mike. This is the perfect document for this kind of question. Thank you. On Wed, Sep 23, 2009 at 7:29 PM, mike deal groganhoc...@gmail.com wrote: I've used this document in the past for sizing purposes: http://www.cisco.com/en/US/prod/collateral/routers/ps259/product_data_sheet0900aecd8057f2e0.pdf mike On Wed, Sep 23, 2009 at 7:59 PM, Nara Shikamaru shikam...@kagadis.comwrote: I had no idea there was a PRI limit. I was thinking, potentially, I may need to terminate 8 PRIs on a 2811 but in truth I'm planning on having 3 2811 for redundancy and spread the span against all three. Plenty of ports between them. I guess my question was also whether the 2811 can handle this kind of scenario, but then if it couldn't I don't think Cisco would allow for 4 PRIs to be terminated to it. I'll ask my AM tomorrow. Thanks, Michael. On Wed, Sep 23, 2009 at 5:26 PM, Michael Ciarfello mciarfe...@iplogic.com wrote: Each ISR router is supposed to only be able to handle X number of PRIs (not physical, more CPU / resource load wise.) I would work with your Cisco AM to have them help you detemine what the limits and loading are. I can't find what documents discussed it. I know I came across a third-party testing report (Mircom maybe.) that had like max 4 PRIs on a 2811. My number might be off, but there was a limit. That's why I would suggest working with your Cisco AM--they should be able to help with those numbers. If you are a partner, the PDI helpdesk should be able to help. If not, then that's what the AM will help you with. Not sure if TAC would assist with these design questions, but you can always try. -- *From:* ccie_voice-boun...@onlinestudylist.com [ ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nara Shikamaru [ shikam...@kagadis.com] *Sent:* Wednesday, September 23, 2009 12:01 PM *To:* OSL Group *Subject:* [OSL | CCIE_Voice] Choosing the right ISR? Okay, my question is not really out of the modules, just a question about a real world scenario. I'm preparing to increase the size of our VoIP network and am aware of the principle differences between the ISRs. Our remote sites will have subscribers, so SRST is not really an issue, and the ISRs are only being used to terminate PRIs and will not be used to route data VLAN traffic. This being the case, are there caveats to using 2811 routers with 8 VWIC ports? I don't really know what to expect by way of offnet traffic, but have had success with the 2811 line and am wondering if I can repurpose for the new network and not have too much to worry about. Also, I am planning on configuring some hardware conferencing but I have no idea yet how popular it will be, no transcoding is planned as our sites are currently all on G711. -- -Shikamaru -- -Shikamaru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- -Shikamaru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
